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 "rdt.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
47 #include "url.h"
48 #include "rtpenc.h"
49 #include "mpegts.h"
50 
51 //#define DEBUG
52 
53 /* Timeout values for socket poll, in ms,
54  * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
61 
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 
66 #define RTSP_FLAG_OPTS(name, longname) \
67  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68  { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
69  { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
70 
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72  { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
73  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76 
77 #define RTSP_REORDERING_OPTS() \
78  { "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 }
79 
81  { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
82  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83  { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88  RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
89  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
90  { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
91  { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
92  { "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 },
94  { NULL },
95 };
96 
97 static const AVOption sdp_options[] = {
98  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
99  { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 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  codec->codec_id = handler->codec_id;
184  rtsp_st->dynamic_handler = handler;
185  if (handler->alloc) {
186  rtsp_st->dynamic_protocol_context = handler->alloc();
187  if (!rtsp_st->dynamic_protocol_context)
188  rtsp_st->dynamic_handler = NULL;
189  }
190 }
191 
192 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
193 static int sdp_parse_rtpmap(AVFormatContext *s,
194  AVStream *st, RTSPStream *rtsp_st,
195  int payload_type, const char *p)
196 {
197  AVCodecContext *codec = st->codec;
198  char buf[256];
199  int i;
200  AVCodec *c;
201  const char *c_name;
202 
203  /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
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  /* search into AVRtpPayloadTypes[] */
213  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
214  }
215 
216  if (codec->codec_id == AV_CODEC_ID_NONE) {
217  RTPDynamicProtocolHandler *handler =
219  init_rtp_handler(handler, rtsp_st, codec);
220  /* If no dynamic handler was found, check with the list of standard
221  * allocated types, if such a stream for some reason happens to
222  * use a private payload type. This isn't handled in rtpdec.c, since
223  * the format name from the rtpmap line never is passed into rtpdec. */
224  if (!rtsp_st->dynamic_handler)
225  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
226  }
227 
228  c = avcodec_find_decoder(codec->codec_id);
229  if (c && c->name)
230  c_name = c->name;
231  else
232  c_name = "(null)";
233 
234  get_word_sep(buf, sizeof(buf), "/", &p);
235  i = atoi(buf);
236  switch (codec->codec_type) {
237  case AVMEDIA_TYPE_AUDIO:
238  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
241  if (i > 0) {
242  codec->sample_rate = i;
243  avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
244  get_word_sep(buf, sizeof(buf), "/", &p);
245  i = atoi(buf);
246  if (i > 0)
247  codec->channels = i;
248  // TODO: there is a bug here; if it is a mono stream, and
249  // less than 22000Hz, faad upconverts to stereo and twice
250  // the frequency. No problem, but the sample rate is being
251  // set here by the sdp line. Patch on its way. (rdm)
252  }
253  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
254  codec->sample_rate);
255  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
256  codec->channels);
257  break;
258  case AVMEDIA_TYPE_VIDEO:
259  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
260  if (i > 0)
261  avpriv_set_pts_info(st, 32, 1, i);
262  break;
263  default:
264  break;
265  }
266  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
267  rtsp_st->dynamic_handler->init(s, st->index,
268  rtsp_st->dynamic_protocol_context);
269  return 0;
270 }
271 
272 /* parse the attribute line from the fmtp a line of an sdp response. This
273  * is broken out as a function because it is used in rtp_h264.c, which is
274  * forthcoming. */
275 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
276  char *value, int value_size)
277 {
278  *p += strspn(*p, SPACE_CHARS);
279  if (**p) {
280  get_word_sep(attr, attr_size, "=", p);
281  if (**p == '=')
282  (*p)++;
283  get_word_sep(value, value_size, ";", p);
284  if (**p == ';')
285  (*p)++;
286  return 1;
287  }
288  return 0;
289 }
290 
291 typedef struct SDPParseState {
292  /* SDP only */
293  struct sockaddr_storage default_ip;
294  int default_ttl;
295  int skip_media;
296 } SDPParseState;
297 
298 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
299  int letter, const char *buf)
300 {
301  RTSPState *rt = s->priv_data;
302  char buf1[64], st_type[64];
303  const char *p;
304  enum AVMediaType codec_type;
305  int payload_type, i;
306  AVStream *st;
307  RTSPStream *rtsp_st;
308  struct sockaddr_storage sdp_ip;
309  int ttl;
310 
311  av_dlog(s, "sdp: %c='%s'\n", letter, buf);
312 
313  p = buf;
314  if (s1->skip_media && letter != 'm')
315  return;
316  switch (letter) {
317  case 'c':
318  get_word(buf1, sizeof(buf1), &p);
319  if (strcmp(buf1, "IN") != 0)
320  return;
321  get_word(buf1, sizeof(buf1), &p);
322  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
323  return;
324  get_word_sep(buf1, sizeof(buf1), "/", &p);
325  if (get_sockaddr(buf1, &sdp_ip))
326  return;
327  ttl = 16;
328  if (*p == '/') {
329  p++;
330  get_word_sep(buf1, sizeof(buf1), "/", &p);
331  ttl = atoi(buf1);
332  }
333  if (s->nb_streams == 0) {
334  s1->default_ip = sdp_ip;
335  s1->default_ttl = ttl;
336  } else {
337  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
338  rtsp_st->sdp_ip = sdp_ip;
339  rtsp_st->sdp_ttl = ttl;
340  }
341  break;
342  case 's':
343  av_dict_set(&s->metadata, "title", p, 0);
344  break;
345  case 'i':
346  if (s->nb_streams == 0) {
347  av_dict_set(&s->metadata, "comment", p, 0);
348  break;
349  }
350  break;
351  case 'm':
352  /* new stream */
353  s1->skip_media = 0;
354  codec_type = AVMEDIA_TYPE_UNKNOWN;
355  get_word(st_type, sizeof(st_type), &p);
356  if (!strcmp(st_type, "audio")) {
357  codec_type = AVMEDIA_TYPE_AUDIO;
358  } else if (!strcmp(st_type, "video")) {
359  codec_type = AVMEDIA_TYPE_VIDEO;
360  } else if (!strcmp(st_type, "application")) {
361  codec_type = AVMEDIA_TYPE_DATA;
362  }
363  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
364  s1->skip_media = 1;
365  return;
366  }
367  rtsp_st = av_mallocz(sizeof(RTSPStream));
368  if (!rtsp_st)
369  return;
370  rtsp_st->stream_index = -1;
371  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
372 
373  rtsp_st->sdp_ip = s1->default_ip;
374  rtsp_st->sdp_ttl = s1->default_ttl;
375 
376  get_word(buf1, sizeof(buf1), &p); /* port */
377  rtsp_st->sdp_port = atoi(buf1);
378 
379  get_word(buf1, sizeof(buf1), &p); /* protocol */
380  if (!strcmp(buf1, "udp"))
382 
383  /* XXX: handle list of formats */
384  get_word(buf1, sizeof(buf1), &p); /* format list */
385  rtsp_st->sdp_payload_type = atoi(buf1);
386 
387  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
388  /* no corresponding stream */
389  if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
390  rt->ts = ff_mpegts_parse_open(s);
391  } else if (rt->server_type == RTSP_SERVER_WMS &&
392  codec_type == AVMEDIA_TYPE_DATA) {
393  /* RTX stream, a stream that carries all the other actual
394  * audio/video streams. Don't expose this to the callers. */
395  } else {
396  st = avformat_new_stream(s, NULL);
397  if (!st)
398  return;
399  st->id = rt->nb_rtsp_streams - 1;
400  rtsp_st->stream_index = st->index;
401  st->codec->codec_type = codec_type;
402  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
403  RTPDynamicProtocolHandler *handler;
404  /* if standard payload type, we can find the codec right now */
406  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
407  st->codec->sample_rate > 0)
408  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
409  /* Even static payload types may need a custom depacketizer */
410  handler = ff_rtp_handler_find_by_id(
411  rtsp_st->sdp_payload_type, st->codec->codec_type);
412  init_rtp_handler(handler, rtsp_st, st->codec);
413  if (handler && handler->init)
414  handler->init(s, st->index,
415  rtsp_st->dynamic_protocol_context);
416  }
417  }
418  /* put a default control url */
419  av_strlcpy(rtsp_st->control_url, rt->control_uri,
420  sizeof(rtsp_st->control_url));
421  break;
422  case 'a':
423  if (av_strstart(p, "control:", &p)) {
424  if (s->nb_streams == 0) {
425  if (!strncmp(p, "rtsp://", 7))
426  av_strlcpy(rt->control_uri, p,
427  sizeof(rt->control_uri));
428  } else {
429  char proto[32];
430  /* get the control url */
431  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
432 
433  /* XXX: may need to add full url resolution */
434  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
435  NULL, NULL, 0, p);
436  if (proto[0] == '\0') {
437  /* relative control URL */
438  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
439  av_strlcat(rtsp_st->control_url, "/",
440  sizeof(rtsp_st->control_url));
441  av_strlcat(rtsp_st->control_url, p,
442  sizeof(rtsp_st->control_url));
443  } else
444  av_strlcpy(rtsp_st->control_url, p,
445  sizeof(rtsp_st->control_url));
446  }
447  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
448  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
449  get_word(buf1, sizeof(buf1), &p);
450  payload_type = atoi(buf1);
451  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
452  if (rtsp_st->stream_index >= 0) {
453  st = s->streams[rtsp_st->stream_index];
454  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
455  }
456  } else if (av_strstart(p, "fmtp:", &p) ||
457  av_strstart(p, "framesize:", &p)) {
458  /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
459  // let dynamic protocol handlers have a stab at the line.
460  get_word(buf1, sizeof(buf1), &p);
461  payload_type = atoi(buf1);
462  for (i = 0; i < rt->nb_rtsp_streams; i++) {
463  rtsp_st = rt->rtsp_streams[i];
464  if (rtsp_st->sdp_payload_type == payload_type &&
465  rtsp_st->dynamic_handler &&
467  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
468  rtsp_st->dynamic_protocol_context, buf);
469  }
470  } else if (av_strstart(p, "range:", &p)) {
471  int64_t start, end;
472 
473  // this is so that seeking on a streamed file can work.
474  rtsp_parse_range_npt(p, &start, &end);
475  s->start_time = start;
476  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
477  s->duration = (end == AV_NOPTS_VALUE) ?
478  AV_NOPTS_VALUE : end - start;
479  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
480  if (atoi(p) == 1)
482  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
483  s->nb_streams > 0) {
484  st = s->streams[s->nb_streams - 1];
485  st->codec->sample_rate = atoi(p);
486  } else {
487  if (rt->server_type == RTSP_SERVER_WMS)
489  if (s->nb_streams > 0) {
490  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
491 
492  if (rt->server_type == RTSP_SERVER_REAL)
493  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
494 
495  if (rtsp_st->dynamic_handler &&
497  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
498  rtsp_st->stream_index,
499  rtsp_st->dynamic_protocol_context, buf);
500  }
501  }
502  break;
503  }
504 }
505 
506 int ff_sdp_parse(AVFormatContext *s, const char *content)
507 {
508  RTSPState *rt = s->priv_data;
509  const char *p;
510  int letter;
511  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
512  * contain long SDP lines containing complete ASF Headers (several
513  * kB) or arrays of MDPR (RM stream descriptor) headers plus
514  * "rulebooks" describing their properties. Therefore, the SDP line
515  * buffer is large.
516  *
517  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
518  * in rtpdec_xiph.c. */
519  char buf[16384], *q;
520  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
521 
522  p = content;
523  for (;;) {
524  p += strspn(p, SPACE_CHARS);
525  letter = *p;
526  if (letter == '\0')
527  break;
528  p++;
529  if (*p != '=')
530  goto next_line;
531  p++;
532  /* get the content */
533  q = buf;
534  while (*p != '\n' && *p != '\r' && *p != '\0') {
535  if ((q - buf) < sizeof(buf) - 1)
536  *q++ = *p;
537  p++;
538  }
539  *q = '\0';
540  sdp_parse_line(s, s1, letter, buf);
541  next_line:
542  while (*p != '\n' && *p != '\0')
543  p++;
544  if (*p == '\n')
545  p++;
546  }
547  rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
548  if (!rt->p) return AVERROR(ENOMEM);
549  return 0;
550 }
551 #endif /* CONFIG_RTPDEC */
552 
554 {
555  RTSPState *rt = s->priv_data;
556  int i;
557 
558  for (i = 0; i < rt->nb_rtsp_streams; i++) {
559  RTSPStream *rtsp_st = rt->rtsp_streams[i];
560  if (!rtsp_st)
561  continue;
562  if (rtsp_st->transport_priv) {
563  if (s->oformat) {
564  AVFormatContext *rtpctx = rtsp_st->transport_priv;
565  av_write_trailer(rtpctx);
567  uint8_t *ptr;
568  avio_close_dyn_buf(rtpctx->pb, &ptr);
569  av_free(ptr);
570  } else {
571  avio_close(rtpctx->pb);
572  }
573  avformat_free_context(rtpctx);
574  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
576  else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
578  }
579  rtsp_st->transport_priv = NULL;
580  if (rtsp_st->rtp_handle)
581  ffurl_close(rtsp_st->rtp_handle);
582  rtsp_st->rtp_handle = NULL;
583  }
584 }
585 
586 /* close and free RTSP streams */
588 {
589  RTSPState *rt = s->priv_data;
590  int i;
591  RTSPStream *rtsp_st;
592 
594  for (i = 0; i < rt->nb_rtsp_streams; i++) {
595  rtsp_st = rt->rtsp_streams[i];
596  if (rtsp_st) {
597  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
598  rtsp_st->dynamic_handler->free(
599  rtsp_st->dynamic_protocol_context);
600  av_free(rtsp_st);
601  }
602  }
603  av_free(rt->rtsp_streams);
604  if (rt->asf_ctx) {
606  }
607  if (rt->ts && CONFIG_RTPDEC)
609  av_free(rt->p);
610  av_free(rt->recvbuf);
611 }
612 
614 {
615  RTSPState *rt = s->priv_data;
616  AVStream *st = NULL;
617  int reordering_queue_size = rt->reordering_queue_size;
618  if (reordering_queue_size < 0) {
620  reordering_queue_size = 0;
621  else
622  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
623  }
624 
625  /* open the RTP context */
626  if (rtsp_st->stream_index >= 0)
627  st = s->streams[rtsp_st->stream_index];
628  if (!st)
630 
631  if (s->oformat && CONFIG_RTSP_MUXER) {
632  int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
633  rtsp_st->rtp_handle,
635  rtsp_st->stream_index);
636  /* Ownership of rtp_handle is passed to the rtp mux context */
637  rtsp_st->rtp_handle = NULL;
638  if (ret < 0)
639  return ret;
640  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
641  return 0; // Don't need to open any parser here
642  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
643  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
644  rtsp_st->dynamic_protocol_context,
645  rtsp_st->dynamic_handler);
646  else if (CONFIG_RTPDEC)
647  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
648  rtsp_st->sdp_payload_type,
649  reordering_queue_size);
650 
651  if (!rtsp_st->transport_priv) {
652  return AVERROR(ENOMEM);
653  } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
654  if (rtsp_st->dynamic_handler) {
656  rtsp_st->dynamic_protocol_context,
657  rtsp_st->dynamic_handler);
658  }
659  }
660 
661  return 0;
662 }
663 
664 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
665 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
666 {
667  const char *q;
668  char *p;
669  int v;
670 
671  q = *pp;
672  q += strspn(q, SPACE_CHARS);
673  v = strtol(q, &p, 10);
674  if (*p == '-') {
675  p++;
676  *min_ptr = v;
677  v = strtol(p, &p, 10);
678  *max_ptr = v;
679  } else {
680  *min_ptr = v;
681  *max_ptr = v;
682  }
683  *pp = p;
684 }
685 
686 /* XXX: only one transport specification is parsed */
687 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
688 {
689  char transport_protocol[16];
690  char profile[16];
691  char lower_transport[16];
692  char parameter[16];
693  RTSPTransportField *th;
694  char buf[256];
695 
696  reply->nb_transports = 0;
697 
698  for (;;) {
699  p += strspn(p, SPACE_CHARS);
700  if (*p == '\0')
701  break;
702 
703  th = &reply->transports[reply->nb_transports];
704 
705  get_word_sep(transport_protocol, sizeof(transport_protocol),
706  "/", &p);
707  if (!av_strcasecmp (transport_protocol, "rtp")) {
708  get_word_sep(profile, sizeof(profile), "/;,", &p);
709  lower_transport[0] = '\0';
710  /* rtp/avp/<protocol> */
711  if (*p == '/') {
712  get_word_sep(lower_transport, sizeof(lower_transport),
713  ";,", &p);
714  }
716  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
717  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
718  /* x-pn-tng/<protocol> */
719  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
720  profile[0] = '\0';
722  } else if (!av_strcasecmp(transport_protocol, "raw")) {
723  get_word_sep(profile, sizeof(profile), "/;,", &p);
724  lower_transport[0] = '\0';
725  /* raw/raw/<protocol> */
726  if (*p == '/') {
727  get_word_sep(lower_transport, sizeof(lower_transport),
728  ";,", &p);
729  }
731  }
732  if (!av_strcasecmp(lower_transport, "TCP"))
734  else
736 
737  if (*p == ';')
738  p++;
739  /* get each parameter */
740  while (*p != '\0' && *p != ',') {
741  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
742  if (!strcmp(parameter, "port")) {
743  if (*p == '=') {
744  p++;
745  rtsp_parse_range(&th->port_min, &th->port_max, &p);
746  }
747  } else if (!strcmp(parameter, "client_port")) {
748  if (*p == '=') {
749  p++;
750  rtsp_parse_range(&th->client_port_min,
751  &th->client_port_max, &p);
752  }
753  } else if (!strcmp(parameter, "server_port")) {
754  if (*p == '=') {
755  p++;
756  rtsp_parse_range(&th->server_port_min,
757  &th->server_port_max, &p);
758  }
759  } else if (!strcmp(parameter, "interleaved")) {
760  if (*p == '=') {
761  p++;
762  rtsp_parse_range(&th->interleaved_min,
763  &th->interleaved_max, &p);
764  }
765  } else if (!strcmp(parameter, "multicast")) {
768  } else if (!strcmp(parameter, "ttl")) {
769  if (*p == '=') {
770  char *end;
771  p++;
772  th->ttl = strtol(p, &end, 10);
773  p = end;
774  }
775  } else if (!strcmp(parameter, "destination")) {
776  if (*p == '=') {
777  p++;
778  get_word_sep(buf, sizeof(buf), ";,", &p);
779  get_sockaddr(buf, &th->destination);
780  }
781  } else if (!strcmp(parameter, "source")) {
782  if (*p == '=') {
783  p++;
784  get_word_sep(buf, sizeof(buf), ";,", &p);
785  av_strlcpy(th->source, buf, sizeof(th->source));
786  }
787  } else if (!strcmp(parameter, "mode")) {
788  if (*p == '=') {
789  p++;
790  get_word_sep(buf, sizeof(buf), ";, ", &p);
791  if (!strcmp(buf, "record") ||
792  !strcmp(buf, "receive"))
793  th->mode_record = 1;
794  }
795  }
796 
797  while (*p != ';' && *p != '\0' && *p != ',')
798  p++;
799  if (*p == ';')
800  p++;
801  }
802  if (*p == ',')
803  p++;
804 
805  reply->nb_transports++;
806  }
807 }
808 
809 static void handle_rtp_info(RTSPState *rt, const char *url,
810  uint32_t seq, uint32_t rtptime)
811 {
812  int i;
813  if (!rtptime || !url[0])
814  return;
815  if (rt->transport != RTSP_TRANSPORT_RTP)
816  return;
817  for (i = 0; i < rt->nb_rtsp_streams; i++) {
818  RTSPStream *rtsp_st = rt->rtsp_streams[i];
819  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
820  if (!rtpctx)
821  continue;
822  if (!strcmp(rtsp_st->control_url, url)) {
823  rtpctx->base_timestamp = rtptime;
824  break;
825  }
826  }
827 }
828 
829 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
830 {
831  int read = 0;
832  char key[20], value[1024], url[1024] = "";
833  uint32_t seq = 0, rtptime = 0;
834 
835  for (;;) {
836  p += strspn(p, SPACE_CHARS);
837  if (!*p)
838  break;
839  get_word_sep(key, sizeof(key), "=", &p);
840  if (*p != '=')
841  break;
842  p++;
843  get_word_sep(value, sizeof(value), ";, ", &p);
844  read++;
845  if (!strcmp(key, "url"))
846  av_strlcpy(url, value, sizeof(url));
847  else if (!strcmp(key, "seq"))
848  seq = strtoul(value, NULL, 10);
849  else if (!strcmp(key, "rtptime"))
850  rtptime = strtoul(value, NULL, 10);
851  if (*p == ',') {
852  handle_rtp_info(rt, url, seq, rtptime);
853  url[0] = '\0';
854  seq = rtptime = 0;
855  read = 0;
856  }
857  if (*p)
858  p++;
859  }
860  if (read > 0)
861  handle_rtp_info(rt, url, seq, rtptime);
862 }
863 
864 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
865  RTSPState *rt, const char *method)
866 {
867  const char *p;
868 
869  /* NOTE: we do case independent match for broken servers */
870  p = buf;
871  if (av_stristart(p, "Session:", &p)) {
872  int t;
873  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
874  if (av_stristart(p, ";timeout=", &p) &&
875  (t = strtol(p, NULL, 10)) > 0) {
876  reply->timeout = t;
877  }
878  } else if (av_stristart(p, "Content-Length:", &p)) {
879  reply->content_length = strtol(p, NULL, 10);
880  } else if (av_stristart(p, "Transport:", &p)) {
881  rtsp_parse_transport(reply, p);
882  } else if (av_stristart(p, "CSeq:", &p)) {
883  reply->seq = strtol(p, NULL, 10);
884  } else if (av_stristart(p, "Range:", &p)) {
885  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
886  } else if (av_stristart(p, "RealChallenge1:", &p)) {
887  p += strspn(p, SPACE_CHARS);
888  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
889  } else if (av_stristart(p, "Server:", &p)) {
890  p += strspn(p, SPACE_CHARS);
891  av_strlcpy(reply->server, p, sizeof(reply->server));
892  } else if (av_stristart(p, "Notice:", &p) ||
893  av_stristart(p, "X-Notice:", &p)) {
894  reply->notice = strtol(p, NULL, 10);
895  } else if (av_stristart(p, "Location:", &p)) {
896  p += strspn(p, SPACE_CHARS);
897  av_strlcpy(reply->location, p , sizeof(reply->location));
898  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
899  p += strspn(p, SPACE_CHARS);
900  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
901  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
902  p += strspn(p, SPACE_CHARS);
903  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
904  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
905  p += strspn(p, SPACE_CHARS);
906  if (method && !strcmp(method, "DESCRIBE"))
907  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
908  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
909  p += strspn(p, SPACE_CHARS);
910  if (method && !strcmp(method, "PLAY"))
911  rtsp_parse_rtp_info(rt, p);
912  } else if (av_stristart(p, "Public:", &p) && rt) {
913  if (strstr(p, "GET_PARAMETER") &&
914  method && !strcmp(method, "OPTIONS"))
915  rt->get_parameter_supported = 1;
916  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
917  p += strspn(p, SPACE_CHARS);
918  rt->accept_dynamic_rate = atoi(p);
919  } else if (av_stristart(p, "Content-Type:", &p)) {
920  p += strspn(p, SPACE_CHARS);
921  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
922  }
923 }
924 
925 /* skip a RTP/TCP interleaved packet */
927 {
928  RTSPState *rt = s->priv_data;
929  int ret, len, len1;
930  uint8_t buf[1024];
931 
932  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
933  if (ret != 3)
934  return;
935  len = AV_RB16(buf + 1);
936 
937  av_dlog(s, "skipping RTP packet len=%d\n", len);
938 
939  /* skip payload */
940  while (len > 0) {
941  len1 = len;
942  if (len1 > sizeof(buf))
943  len1 = sizeof(buf);
944  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
945  if (ret != len1)
946  return;
947  len -= len1;
948  }
949 }
950 
952  unsigned char **content_ptr,
953  int return_on_interleaved_data, const char *method)
954 {
955  RTSPState *rt = s->priv_data;
956  char buf[4096], buf1[1024], *q;
957  unsigned char ch;
958  const char *p;
959  int ret, content_length, line_count = 0, request = 0;
960  unsigned char *content = NULL;
961 
962 start:
963  line_count = 0;
964  request = 0;
965  content = NULL;
966  memset(reply, 0, sizeof(*reply));
967 
968  /* parse reply (XXX: use buffers) */
969  rt->last_reply[0] = '\0';
970  for (;;) {
971  q = buf;
972  for (;;) {
973  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
974  av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
975  if (ret != 1)
976  return AVERROR_EOF;
977  if (ch == '\n')
978  break;
979  if (ch == '$') {
980  /* XXX: only parse it if first char on line ? */
981  if (return_on_interleaved_data) {
982  return 1;
983  } else
985  } else if (ch != '\r') {
986  if ((q - buf) < sizeof(buf) - 1)
987  *q++ = ch;
988  }
989  }
990  *q = '\0';
991 
992  av_dlog(s, "line='%s'\n", buf);
993 
994  /* test if last line */
995  if (buf[0] == '\0')
996  break;
997  p = buf;
998  if (line_count == 0) {
999  /* get reply code */
1000  get_word(buf1, sizeof(buf1), &p);
1001  if (!strncmp(buf1, "RTSP/", 5)) {
1002  get_word(buf1, sizeof(buf1), &p);
1003  reply->status_code = atoi(buf1);
1004  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1005  } else {
1006  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1007  get_word(buf1, sizeof(buf1), &p); // object
1008  request = 1;
1009  }
1010  } else {
1011  ff_rtsp_parse_line(reply, p, rt, method);
1012  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1013  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1014  }
1015  line_count++;
1016  }
1017 
1018  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1019  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1020 
1021  content_length = reply->content_length;
1022  if (content_length > 0) {
1023  /* leave some room for a trailing '\0' (useful for simple parsing) */
1024  content = av_malloc(content_length + 1);
1025  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1026  content[content_length] = '\0';
1027  }
1028  if (content_ptr)
1029  *content_ptr = content;
1030  else
1031  av_free(content);
1032 
1033  if (request) {
1034  char buf[1024];
1035  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1036  const char* ptr = buf;
1037 
1038  if (!strcmp(reply->reason, "OPTIONS")) {
1039  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1040  if (reply->seq)
1041  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1042  if (reply->session_id[0])
1043  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1044  reply->session_id);
1045  } else {
1046  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1047  }
1048  av_strlcat(buf, "\r\n", sizeof(buf));
1049 
1050  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1051  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1052  ptr = base64buf;
1053  }
1054  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1055 
1056  rt->last_cmd_time = av_gettime();
1057  /* Even if the request from the server had data, it is not the data
1058  * that the caller wants or expects. The memory could also be leaked
1059  * if the actual following reply has content data. */
1060  if (content_ptr)
1061  av_freep(content_ptr);
1062  /* If method is set, this is called from ff_rtsp_send_cmd,
1063  * where a reply to exactly this request is awaited. For
1064  * callers from within packet receiving, we just want to
1065  * return to the caller and go back to receiving packets. */
1066  if (method)
1067  goto start;
1068  return 0;
1069  }
1070 
1071  if (rt->seq != reply->seq) {
1072  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1073  rt->seq, reply->seq);
1074  }
1075 
1076  /* EOS */
1077  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1078  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1079  reply->notice == 2306 /* Continuous Feed Terminated */) {
1080  rt->state = RTSP_STATE_IDLE;
1081  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1082  return AVERROR(EIO); /* data or server error */
1083  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1084  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1085  return AVERROR(EPERM);
1086 
1087  return 0;
1088 }
1089 
1103 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1104  const char *method, const char *url,
1105  const char *headers,
1106  const unsigned char *send_content,
1107  int send_content_length)
1108 {
1109  RTSPState *rt = s->priv_data;
1110  char buf[4096], *out_buf;
1111  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1112 
1113  /* Add in RTSP headers */
1114  out_buf = buf;
1115  rt->seq++;
1116  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1117  if (headers)
1118  av_strlcat(buf, headers, sizeof(buf));
1119  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1120  if (rt->session_id[0] != '\0' && (!headers ||
1121  !strstr(headers, "\nIf-Match:"))) {
1122  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1123  }
1124  if (rt->auth[0]) {
1125  char *str = ff_http_auth_create_response(&rt->auth_state,
1126  rt->auth, url, method);
1127  if (str)
1128  av_strlcat(buf, str, sizeof(buf));
1129  av_free(str);
1130  }
1131  if (send_content_length > 0 && send_content)
1132  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1133  av_strlcat(buf, "\r\n", sizeof(buf));
1134 
1135  /* base64 encode rtsp if tunneling */
1136  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1137  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1138  out_buf = base64buf;
1139  }
1140 
1141  av_dlog(s, "Sending:\n%s--\n", buf);
1142 
1143  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1144  if (send_content_length > 0 && send_content) {
1145  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1146  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1147  "with content data not supported\n");
1148  return AVERROR_PATCHWELCOME;
1149  }
1150  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1151  }
1152  rt->last_cmd_time = av_gettime();
1153 
1154  return 0;
1155 }
1156 
1157 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1158  const char *url, const char *headers)
1159 {
1160  return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1161 }
1162 
1163 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1164  const char *headers, RTSPMessageHeader *reply,
1165  unsigned char **content_ptr)
1166 {
1167  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1168  content_ptr, NULL, 0);
1169 }
1170 
1172  const char *method, const char *url,
1173  const char *header,
1174  RTSPMessageHeader *reply,
1175  unsigned char **content_ptr,
1176  const unsigned char *send_content,
1177  int send_content_length)
1178 {
1179  RTSPState *rt = s->priv_data;
1180  HTTPAuthType cur_auth_type;
1181  int ret, attempts = 0;
1182 
1183 retry:
1184  cur_auth_type = rt->auth_state.auth_type;
1185  if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1186  send_content,
1187  send_content_length)))
1188  return ret;
1189 
1190  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1191  return ret;
1192  attempts++;
1193 
1194  if (reply->status_code == 401 &&
1195  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1196  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1197  goto retry;
1198 
1199  if (reply->status_code > 400){
1200  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1201  method,
1202  reply->status_code,
1203  reply->reason);
1204  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1205  }
1206 
1207  return 0;
1208 }
1209 
1210 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1211  int lower_transport, const char *real_challenge)
1212 {
1213  RTSPState *rt = s->priv_data;
1214  int rtx = 0, j, i, err, interleave = 0, port_off;
1215  RTSPStream *rtsp_st;
1216  RTSPMessageHeader reply1, *reply = &reply1;
1217  char cmd[2048];
1218  const char *trans_pref;
1219 
1220  if (rt->transport == RTSP_TRANSPORT_RDT)
1221  trans_pref = "x-pn-tng";
1222  else if (rt->transport == RTSP_TRANSPORT_RAW)
1223  trans_pref = "RAW/RAW";
1224  else
1225  trans_pref = "RTP/AVP";
1226 
1227  /* default timeout: 1 minute */
1228  rt->timeout = 60;
1229 
1230  /* for each stream, make the setup request */
1231  /* XXX: we assume the same server is used for the control of each
1232  * RTSP stream */
1233 
1234  /* Choose a random starting offset within the first half of the
1235  * port range, to allow for a number of ports to try even if the offset
1236  * happens to be at the end of the random range. */
1237  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1238  /* even random offset */
1239  port_off -= port_off & 0x01;
1240 
1241  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1242  char transport[2048];
1243 
1244  /*
1245  * WMS serves all UDP data over a single connection, the RTX, which
1246  * isn't necessarily the first in the SDP but has to be the first
1247  * to be set up, else the second/third SETUP will fail with a 461.
1248  */
1249  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1250  rt->server_type == RTSP_SERVER_WMS) {
1251  if (i == 0) {
1252  /* rtx first */
1253  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1254  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1255  if (len >= 4 &&
1256  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1257  "/rtx"))
1258  break;
1259  }
1260  if (rtx == rt->nb_rtsp_streams)
1261  return -1; /* no RTX found */
1262  rtsp_st = rt->rtsp_streams[rtx];
1263  } else
1264  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1265  } else
1266  rtsp_st = rt->rtsp_streams[i];
1267 
1268  /* RTP/UDP */
1269  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1270  char buf[256];
1271 
1272  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1273  port = reply->transports[0].client_port_min;
1274  goto have_port;
1275  }
1276 
1277  /* first try in specified port range */
1278  while (j <= rt->rtp_port_max) {
1279  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1280  "?localport=%d", j);
1281  /* we will use two ports per rtp stream (rtp and rtcp) */
1282  j += 2;
1283  if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1284  &s->interrupt_callback, NULL))
1285  goto rtp_opened;
1286  }
1287 
1288  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1289  err = AVERROR(EIO);
1290  goto fail;
1291 
1292  rtp_opened:
1293  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1294  have_port:
1295  snprintf(transport, sizeof(transport) - 1,
1296  "%s/UDP;", trans_pref);
1297  if (rt->server_type != RTSP_SERVER_REAL)
1298  av_strlcat(transport, "unicast;", sizeof(transport));
1299  av_strlcatf(transport, sizeof(transport),
1300  "client_port=%d", port);
1301  if (rt->transport == RTSP_TRANSPORT_RTP &&
1302  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1303  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1304  }
1305 
1306  /* RTP/TCP */
1307  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1308  /* For WMS streams, the application streams are only used for
1309  * UDP. When trying to set it up for TCP streams, the server
1310  * will return an error. Therefore, we skip those streams. */
1311  if (rt->server_type == RTSP_SERVER_WMS &&
1312  (rtsp_st->stream_index < 0 ||
1313  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1315  continue;
1316  snprintf(transport, sizeof(transport) - 1,
1317  "%s/TCP;", trans_pref);
1318  if (rt->transport != RTSP_TRANSPORT_RDT)
1319  av_strlcat(transport, "unicast;", sizeof(transport));
1320  av_strlcatf(transport, sizeof(transport),
1321  "interleaved=%d-%d",
1322  interleave, interleave + 1);
1323  interleave += 2;
1324  }
1325 
1326  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1327  snprintf(transport, sizeof(transport) - 1,
1328  "%s/UDP;multicast", trans_pref);
1329  }
1330  if (s->oformat) {
1331  av_strlcat(transport, ";mode=record", sizeof(transport));
1332  } else if (rt->server_type == RTSP_SERVER_REAL ||
1334  av_strlcat(transport, ";mode=play", sizeof(transport));
1335  snprintf(cmd, sizeof(cmd),
1336  "Transport: %s\r\n",
1337  transport);
1338  if (rt->accept_dynamic_rate)
1339  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1340  if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1341  char real_res[41], real_csum[9];
1342  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1343  real_challenge);
1344  av_strlcatf(cmd, sizeof(cmd),
1345  "If-Match: %s\r\n"
1346  "RealChallenge2: %s, sd=%s\r\n",
1347  rt->session_id, real_res, real_csum);
1348  }
1349  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1350  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1351  err = 1;
1352  goto fail;
1353  } else if (reply->status_code != RTSP_STATUS_OK ||
1354  reply->nb_transports != 1) {
1355  err = AVERROR_INVALIDDATA;
1356  goto fail;
1357  }
1358 
1359  /* XXX: same protocol for all streams is required */
1360  if (i > 0) {
1361  if (reply->transports[0].lower_transport != rt->lower_transport ||
1362  reply->transports[0].transport != rt->transport) {
1363  err = AVERROR_INVALIDDATA;
1364  goto fail;
1365  }
1366  } else {
1367  rt->lower_transport = reply->transports[0].lower_transport;
1368  rt->transport = reply->transports[0].transport;
1369  }
1370 
1371  /* Fail if the server responded with another lower transport mode
1372  * than what we requested. */
1373  if (reply->transports[0].lower_transport != lower_transport) {
1374  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1375  err = AVERROR_INVALIDDATA;
1376  goto fail;
1377  }
1378 
1379  switch(reply->transports[0].lower_transport) {
1381  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1382  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1383  break;
1384 
1385  case RTSP_LOWER_TRANSPORT_UDP: {
1386  char url[1024], options[30] = "";
1387 
1388  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1389  av_strlcpy(options, "?connect=1", sizeof(options));
1390  /* Use source address if specified */
1391  if (reply->transports[0].source[0]) {
1392  ff_url_join(url, sizeof(url), "rtp", NULL,
1393  reply->transports[0].source,
1394  reply->transports[0].server_port_min, "%s", options);
1395  } else {
1396  ff_url_join(url, sizeof(url), "rtp", NULL, host,
1397  reply->transports[0].server_port_min, "%s", options);
1398  }
1399  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1400  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1401  err = AVERROR_INVALIDDATA;
1402  goto fail;
1403  }
1404  /* Try to initialize the connection state in a
1405  * potential NAT router by sending dummy packets.
1406  * RTP/RTCP dummy packets are used for RDT, too.
1407  */
1408  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1409  CONFIG_RTPDEC)
1411  break;
1412  }
1414  char url[1024], namebuf[50], optbuf[20] = "";
1415  struct sockaddr_storage addr;
1416  int port, ttl;
1417 
1418  if (reply->transports[0].destination.ss_family) {
1419  addr = reply->transports[0].destination;
1420  port = reply->transports[0].port_min;
1421  ttl = reply->transports[0].ttl;
1422  } else {
1423  addr = rtsp_st->sdp_ip;
1424  port = rtsp_st->sdp_port;
1425  ttl = rtsp_st->sdp_ttl;
1426  }
1427  if (ttl > 0)
1428  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1429  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1430  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1431  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1432  port, "%s", optbuf);
1433  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1434  &s->interrupt_callback, NULL) < 0) {
1435  err = AVERROR_INVALIDDATA;
1436  goto fail;
1437  }
1438  break;
1439  }
1440  }
1441 
1442  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1443  goto fail;
1444  }
1445 
1446  if (rt->nb_rtsp_streams && reply->timeout > 0)
1447  rt->timeout = reply->timeout;
1448 
1449  if (rt->server_type == RTSP_SERVER_REAL)
1450  rt->need_subscription = 1;
1451 
1452  return 0;
1453 
1454 fail:
1455  ff_rtsp_undo_setup(s);
1456  return err;
1457 }
1458 
1460 {
1461  RTSPState *rt = s->priv_data;
1462  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1463  ffurl_close(rt->rtsp_hd);
1464  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1465 }
1466 
1468 {
1469  RTSPState *rt = s->priv_data;
1470  char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1471  int port, err, tcp_fd;
1472  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1473  int lower_transport_mask = 0;
1474  char real_challenge[64] = "";
1475  struct sockaddr_storage peer;
1476  socklen_t peer_len = sizeof(peer);
1477 
1478  if (rt->rtp_port_max < rt->rtp_port_min) {
1479  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1480  "than min port %d\n", rt->rtp_port_max,
1481  rt->rtp_port_min);
1482  return AVERROR(EINVAL);
1483  }
1484 
1485  if (!ff_network_init())
1486  return AVERROR(EIO);
1487 
1488  if (s->max_delay < 0) /* Not set by the caller */
1490 
1495  }
1496  /* Only pass through valid flags from here */
1498 
1499 redirect:
1500  lower_transport_mask = rt->lower_transport_mask;
1501  /* extract hostname and port */
1502  av_url_split(NULL, 0, auth, sizeof(auth),
1503  host, sizeof(host), &port, path, sizeof(path), s->filename);
1504  if (*auth) {
1505  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1506  }
1507  if (port < 0)
1508  port = RTSP_DEFAULT_PORT;
1509 
1510  if (!lower_transport_mask)
1511  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1512 
1513  if (s->oformat) {
1514  /* Only UDP or TCP - UDP multicast isn't supported. */
1515  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1516  (1 << RTSP_LOWER_TRANSPORT_TCP);
1517  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1518  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1519  "only UDP and TCP are supported for output.\n");
1520  err = AVERROR(EINVAL);
1521  goto fail;
1522  }
1523  }
1524 
1525  /* Construct the URI used in request; this is similar to s->filename,
1526  * but with authentication credentials removed and RTSP specific options
1527  * stripped out. */
1528  ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1529  host, port, "%s", path);
1530 
1531  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1532  /* set up initial handshake for tunneling */
1533  char httpname[1024];
1534  char sessioncookie[17];
1535  char headers[1024];
1536 
1537  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1538  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1540 
1541  /* GET requests */
1542  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1543  &s->interrupt_callback) < 0) {
1544  err = AVERROR(EIO);
1545  goto fail;
1546  }
1547 
1548  /* generate GET headers */
1549  snprintf(headers, sizeof(headers),
1550  "x-sessioncookie: %s\r\n"
1551  "Accept: application/x-rtsp-tunnelled\r\n"
1552  "Pragma: no-cache\r\n"
1553  "Cache-Control: no-cache\r\n",
1554  sessioncookie);
1555  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1556 
1557  /* complete the connection */
1558  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1559  err = AVERROR(EIO);
1560  goto fail;
1561  }
1562 
1563  /* POST requests */
1564  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1565  &s->interrupt_callback) < 0 ) {
1566  err = AVERROR(EIO);
1567  goto fail;
1568  }
1569 
1570  /* generate POST headers */
1571  snprintf(headers, sizeof(headers),
1572  "x-sessioncookie: %s\r\n"
1573  "Content-Type: application/x-rtsp-tunnelled\r\n"
1574  "Pragma: no-cache\r\n"
1575  "Cache-Control: no-cache\r\n"
1576  "Content-Length: 32767\r\n"
1577  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1578  sessioncookie);
1579  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1580  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1581 
1582  /* Initialize the authentication state for the POST session. The HTTP
1583  * protocol implementation doesn't properly handle multi-pass
1584  * authentication for POST requests, since it would require one of
1585  * the following:
1586  * - implementing Expect: 100-continue, which many HTTP servers
1587  * don't support anyway, even less the RTSP servers that do HTTP
1588  * tunneling
1589  * - sending the whole POST data until getting a 401 reply specifying
1590  * what authentication method to use, then resending all that data
1591  * - waiting for potential 401 replies directly after sending the
1592  * POST header (waiting for some unspecified time)
1593  * Therefore, we copy the full auth state, which works for both basic
1594  * and digest. (For digest, we would have to synchronize the nonce
1595  * count variable between the two sessions, if we'd do more requests
1596  * with the original session, though.)
1597  */
1599 
1600  /* complete the connection */
1601  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1602  err = AVERROR(EIO);
1603  goto fail;
1604  }
1605  } else {
1606  /* open the tcp connection */
1607  ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1608  if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1609  &s->interrupt_callback, NULL) < 0) {
1610  err = AVERROR(EIO);
1611  goto fail;
1612  }
1613  rt->rtsp_hd_out = rt->rtsp_hd;
1614  }
1615  rt->seq = 0;
1616 
1617  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1618  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1619  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1620  NULL, 0, NI_NUMERICHOST);
1621  }
1622 
1623  /* request options supported by the server; this also detects server
1624  * type */
1625  for (rt->server_type = RTSP_SERVER_RTP;;) {
1626  cmd[0] = 0;
1627  if (rt->server_type == RTSP_SERVER_REAL)
1628  av_strlcat(cmd,
1629  /*
1630  * The following entries are required for proper
1631  * streaming from a Realmedia server. They are
1632  * interdependent in some way although we currently
1633  * don't quite understand how. Values were copied
1634  * from mplayer SVN r23589.
1635  * ClientChallenge is a 16-byte ID in hex
1636  * CompanyID is a 16-byte ID in base64
1637  */
1638  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1639  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1640  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1641  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1642  sizeof(cmd));
1643  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1644  if (reply->status_code != RTSP_STATUS_OK) {
1645  err = AVERROR_INVALIDDATA;
1646  goto fail;
1647  }
1648 
1649  /* detect server type if not standard-compliant RTP */
1650  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1652  continue;
1653  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1655  } else if (rt->server_type == RTSP_SERVER_REAL)
1656  strcpy(real_challenge, reply->real_challenge);
1657  break;
1658  }
1659 
1660  if (s->iformat && CONFIG_RTSP_DEMUXER)
1661  err = ff_rtsp_setup_input_streams(s, reply);
1662  else if (CONFIG_RTSP_MUXER)
1663  err = ff_rtsp_setup_output_streams(s, host);
1664  if (err)
1665  goto fail;
1666 
1667  do {
1668  int lower_transport = ff_log2_tab[lower_transport_mask &
1669  ~(lower_transport_mask - 1)];
1670 
1671  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1672  rt->server_type == RTSP_SERVER_REAL ?
1673  real_challenge : NULL);
1674  if (err < 0)
1675  goto fail;
1676  lower_transport_mask &= ~(1 << lower_transport);
1677  if (lower_transport_mask == 0 && err == 1) {
1678  err = AVERROR(EPROTONOSUPPORT);
1679  goto fail;
1680  }
1681  } while (err);
1682 
1683  rt->lower_transport_mask = lower_transport_mask;
1684  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1685  rt->state = RTSP_STATE_IDLE;
1686  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1687  return 0;
1688  fail:
1691  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1692  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1693  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1694  reply->status_code,
1695  s->filename);
1696  goto redirect;
1697  }
1698  ff_network_close();
1699  return err;
1700 }
1701 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1702 
1703 #if CONFIG_RTPDEC
1704 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1705  uint8_t *buf, int buf_size, int64_t wait_end)
1706 {
1707  RTSPState *rt = s->priv_data;
1708  RTSPStream *rtsp_st;
1709  int n, i, ret, tcp_fd, timeout_cnt = 0;
1710  int max_p = 0;
1711  struct pollfd *p = rt->p;
1712  int *fds = NULL, fdsnum, fdsidx;
1713 
1714  for (;;) {
1716  return AVERROR_EXIT;
1717  if (wait_end && wait_end - av_gettime() < 0)
1718  return AVERROR(EAGAIN);
1719  max_p = 0;
1720  if (rt->rtsp_hd) {
1721  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1722  p[max_p].fd = tcp_fd;
1723  p[max_p++].events = POLLIN;
1724  } else {
1725  tcp_fd = -1;
1726  }
1727  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1728  rtsp_st = rt->rtsp_streams[i];
1729  if (rtsp_st->rtp_handle) {
1730  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1731  &fds, &fdsnum)) {
1732  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1733  return ret;
1734  }
1735  if (fdsnum != 2) {
1736  av_log(s, AV_LOG_ERROR,
1737  "Number of fds %d not supported\n", fdsnum);
1738  return AVERROR_INVALIDDATA;
1739  }
1740  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1741  p[max_p].fd = fds[fdsidx];
1742  p[max_p++].events = POLLIN;
1743  }
1744  av_free(fds);
1745  }
1746  }
1747  n = poll(p, max_p, POLL_TIMEOUT_MS);
1748  if (n > 0) {
1749  int j = 1 - (tcp_fd == -1);
1750  timeout_cnt = 0;
1751  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1752  rtsp_st = rt->rtsp_streams[i];
1753  if (rtsp_st->rtp_handle) {
1754  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1755  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1756  if (ret > 0) {
1757  *prtsp_st = rtsp_st;
1758  return ret;
1759  }
1760  }
1761  j+=2;
1762  }
1763  }
1764 #if CONFIG_RTSP_DEMUXER
1765  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1766  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1767  if (rt->state == RTSP_STATE_STREAMING) {
1769  return AVERROR_EOF;
1770  else
1772  "Unable to answer to TEARDOWN\n");
1773  } else
1774  return 0;
1775  } else {
1776  RTSPMessageHeader reply;
1777  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1778  if (ret < 0)
1779  return ret;
1780  /* XXX: parse message */
1781  if (rt->state != RTSP_STATE_STREAMING)
1782  return 0;
1783  }
1784  }
1785 #endif
1786  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1787  return AVERROR(ETIMEDOUT);
1788  } else if (n < 0 && errno != EINTR)
1789  return AVERROR(errno);
1790  }
1791 }
1792 
1793 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1794  const uint8_t *buf, int len)
1795 {
1796  RTSPState *rt = s->priv_data;
1797  int i;
1798  if (len < 0)
1799  return len;
1800  if (rt->nb_rtsp_streams == 1) {
1801  *rtsp_st = rt->rtsp_streams[0];
1802  return len;
1803  }
1804  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1805  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1806  int no_ssrc = 0;
1807  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1808  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1809  if (!rtpctx)
1810  continue;
1811  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1812  *rtsp_st = rt->rtsp_streams[i];
1813  return len;
1814  }
1815  if (!rtpctx->ssrc)
1816  no_ssrc = 1;
1817  }
1818  if (no_ssrc) {
1820  "Unable to pick stream for packet - SSRC not known for "
1821  "all streams\n");
1822  return AVERROR(EAGAIN);
1823  }
1824  } else {
1825  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1826  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1827  *rtsp_st = rt->rtsp_streams[i];
1828  return len;
1829  }
1830  }
1831  }
1832  }
1833  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1834  return AVERROR(EAGAIN);
1835 }
1836 
1838 {
1839  RTSPState *rt = s->priv_data;
1840  int ret, len;
1841  RTSPStream *rtsp_st, *first_queue_st = NULL;
1842  int64_t wait_end = 0;
1843 
1844  if (rt->nb_byes == rt->nb_rtsp_streams)
1845  return AVERROR_EOF;
1846 
1847  /* get next frames from the same RTP packet */
1848  if (rt->cur_transport_priv) {
1849  if (rt->transport == RTSP_TRANSPORT_RDT) {
1850  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1851  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1852  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1853  } else if (rt->ts && CONFIG_RTPDEC) {
1854  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1855  if (ret >= 0) {
1856  rt->recvbuf_pos += ret;
1857  ret = rt->recvbuf_pos < rt->recvbuf_len;
1858  }
1859  } else
1860  ret = -1;
1861  if (ret == 0) {
1862  rt->cur_transport_priv = NULL;
1863  return 0;
1864  } else if (ret == 1) {
1865  return 0;
1866  } else
1867  rt->cur_transport_priv = NULL;
1868  }
1869 
1870  if (rt->transport == RTSP_TRANSPORT_RTP) {
1871  int i;
1872  int64_t first_queue_time = 0;
1873  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1874  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1875  int64_t queue_time;
1876  if (!rtpctx)
1877  continue;
1878  queue_time = ff_rtp_queued_packet_time(rtpctx);
1879  if (queue_time && (queue_time - first_queue_time < 0 ||
1880  !first_queue_time)) {
1881  first_queue_time = queue_time;
1882  first_queue_st = rt->rtsp_streams[i];
1883  }
1884  }
1885  if (first_queue_time)
1886  wait_end = first_queue_time + s->max_delay;
1887  }
1888 
1889  /* read next RTP packet */
1890  redo:
1891  if (!rt->recvbuf) {
1893  if (!rt->recvbuf)
1894  return AVERROR(ENOMEM);
1895  }
1896 
1897  switch(rt->lower_transport) {
1898  default:
1899 #if CONFIG_RTSP_DEMUXER
1901  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1902  break;
1903 #endif
1906  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1907  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1908  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1909  break;
1911  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1912  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1913  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1915  break;
1916  }
1917  if (len == AVERROR(EAGAIN) && first_queue_st &&
1918  rt->transport == RTSP_TRANSPORT_RTP) {
1919  rtsp_st = first_queue_st;
1920  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1921  goto end;
1922  }
1923  if (len < 0)
1924  return len;
1925  if (len == 0)
1926  return AVERROR_EOF;
1927  if (rt->transport == RTSP_TRANSPORT_RDT) {
1928  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1929  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1930  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1931  if (ret < 0) {
1932  /* Either bad packet, or a RTCP packet. Check if the
1933  * first_rtcp_ntp_time field was initialized. */
1934  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1935  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1936  /* first_rtcp_ntp_time has been initialized for this stream,
1937  * copy the same value to all other uninitialized streams,
1938  * in order to map their timestamp origin to the same ntp time
1939  * as this one. */
1940  int i;
1941  AVStream *st = NULL;
1942  if (rtsp_st->stream_index >= 0)
1943  st = s->streams[rtsp_st->stream_index];
1944  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1945  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1946  AVStream *st2 = NULL;
1947  if (rt->rtsp_streams[i]->stream_index >= 0)
1948  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1949  if (rtpctx2 && st && st2 &&
1950  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1951  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1952  rtpctx2->rtcp_ts_offset = av_rescale_q(
1953  rtpctx->rtcp_ts_offset, st->time_base,
1954  st2->time_base);
1955  }
1956  }
1957  }
1958  if (ret == -RTCP_BYE) {
1959  rt->nb_byes++;
1960 
1961  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1962  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1963 
1964  if (rt->nb_byes == rt->nb_rtsp_streams)
1965  return AVERROR_EOF;
1966  }
1967  }
1968  } else if (rt->ts && CONFIG_RTPDEC) {
1969  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1970  if (ret >= 0) {
1971  if (ret < len) {
1972  rt->recvbuf_len = len;
1973  rt->recvbuf_pos = ret;
1974  rt->cur_transport_priv = rt->ts;
1975  return 1;
1976  } else {
1977  ret = 0;
1978  }
1979  }
1980  } else {
1981  return AVERROR_INVALIDDATA;
1982  }
1983 end:
1984  if (ret < 0)
1985  goto redo;
1986  if (ret == 1)
1987  /* more packets may follow, so we save the RTP context */
1988  rt->cur_transport_priv = rtsp_st->transport_priv;
1989 
1990  return ret;
1991 }
1992 #endif /* CONFIG_RTPDEC */
1993 
1994 #if CONFIG_SDP_DEMUXER
1995 static int sdp_probe(AVProbeData *p1)
1996 {
1997  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1998 
1999  /* we look for a line beginning "c=IN IP" */
2000  while (p < p_end && *p != '\0') {
2001  if (p + sizeof("c=IN IP") - 1 < p_end &&
2002  av_strstart(p, "c=IN IP", NULL))
2003  return AVPROBE_SCORE_MAX / 2;
2004 
2005  while (p < p_end - 1 && *p != '\n') p++;
2006  if (++p >= p_end)
2007  break;
2008  if (*p == '\r')
2009  p++;
2010  }
2011  return 0;
2012 }
2013 
2014 static int sdp_read_header(AVFormatContext *s)
2015 {
2016  RTSPState *rt = s->priv_data;
2017  RTSPStream *rtsp_st;
2018  int size, i, err;
2019  char *content;
2020  char url[1024];
2021 
2022  if (!ff_network_init())
2023  return AVERROR(EIO);
2024 
2025  if (s->max_delay < 0) /* Not set by the caller */
2027  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2029 
2030  /* read the whole sdp file */
2031  /* XXX: better loading */
2032  content = av_malloc(SDP_MAX_SIZE);
2033  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2034  if (size <= 0) {
2035  av_free(content);
2036  return AVERROR_INVALIDDATA;
2037  }
2038  content[size] ='\0';
2039 
2040  err = ff_sdp_parse(s, content);
2041  av_free(content);
2042  if (err) goto fail;
2043 
2044  /* open each RTP stream */
2045  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2046  char namebuf[50];
2047  rtsp_st = rt->rtsp_streams[i];
2048 
2049  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2050  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2051  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2052  ff_url_join(url, sizeof(url), "rtp", NULL,
2053  namebuf, rtsp_st->sdp_port,
2054  "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2055  rtsp_st->sdp_ttl,
2056  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2057  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2058  &s->interrupt_callback, NULL) < 0) {
2059  err = AVERROR_INVALIDDATA;
2060  goto fail;
2061  }
2062  }
2063  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2064  goto fail;
2065  }
2066  return 0;
2067 fail:
2069  ff_network_close();
2070  return err;
2071 }
2072 
2073 static int sdp_read_close(AVFormatContext *s)
2074 {
2076  ff_network_close();
2077  return 0;
2078 }
2079 
2080 static const AVClass sdp_demuxer_class = {
2081  .class_name = "SDP demuxer",
2082  .item_name = av_default_item_name,
2083  .option = sdp_options,
2084  .version = LIBAVUTIL_VERSION_INT,
2085 };
2086 
2087 AVInputFormat ff_sdp_demuxer = {
2088  .name = "sdp",
2089  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2090  .priv_data_size = sizeof(RTSPState),
2091  .read_probe = sdp_probe,
2092  .read_header = sdp_read_header,
2094  .read_close = sdp_read_close,
2095  .priv_class = &sdp_demuxer_class,
2096 };
2097 #endif /* CONFIG_SDP_DEMUXER */
2098 
2099 #if CONFIG_RTP_DEMUXER
2100 static int rtp_probe(AVProbeData *p)
2101 {
2102  if (av_strstart(p->filename, "rtp:", NULL))
2103  return AVPROBE_SCORE_MAX;
2104  return 0;
2105 }
2106 
2107 static int rtp_read_header(AVFormatContext *s)
2108 {
2109  uint8_t recvbuf[1500];
2110  char host[500], sdp[500];
2111  int ret, port;
2112  URLContext* in = NULL;
2113  int payload_type;
2114  AVCodecContext codec = { 0 };
2115  struct sockaddr_storage addr;
2116  AVIOContext pb;
2117  socklen_t addrlen = sizeof(addr);
2118  RTSPState *rt = s->priv_data;
2119 
2120  if (!ff_network_init())
2121  return AVERROR(EIO);
2122 
2123  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2124  &s->interrupt_callback, NULL);
2125  if (ret)
2126  goto fail;
2127 
2128  while (1) {
2129  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2130  if (ret == AVERROR(EAGAIN))
2131  continue;
2132  if (ret < 0)
2133  goto fail;
2134  if (ret < 12) {
2135  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2136  continue;
2137  }
2138 
2139  if ((recvbuf[0] & 0xc0) != 0x80) {
2140  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2141  "received\n");
2142  continue;
2143  }
2144 
2145  if (RTP_PT_IS_RTCP(recvbuf[1]))
2146  continue;
2147 
2148  payload_type = recvbuf[1] & 0x7f;
2149  break;
2150  }
2151  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2152  ffurl_close(in);
2153  in = NULL;
2154 
2155  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2156  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2157  "without an SDP file describing it\n",
2158  payload_type);
2159  goto fail;
2160  }
2161  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2162  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2163  "properly you need an SDP file "
2164  "describing it\n");
2165  }
2166 
2167  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2168  NULL, 0, s->filename);
2169 
2170  snprintf(sdp, sizeof(sdp),
2171  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2172  addr.ss_family == AF_INET ? 4 : 6, host,
2173  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2174  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2175  port, payload_type);
2176  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2177 
2178  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2179  s->pb = &pb;
2180 
2181  /* sdp_read_header initializes this again */
2182  ff_network_close();
2183 
2184  rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2185 
2186  ret = sdp_read_header(s);
2187  s->pb = NULL;
2188  return ret;
2189 
2190 fail:
2191  if (in)
2192  ffurl_close(in);
2193  ff_network_close();
2194  return ret;
2195 }
2196 
2197 static const AVClass rtp_demuxer_class = {
2198  .class_name = "RTP demuxer",
2199  .item_name = av_default_item_name,
2200  .option = rtp_options,
2201  .version = LIBAVUTIL_VERSION_INT,
2202 };
2203 
2204 AVInputFormat ff_rtp_demuxer = {
2205  .name = "rtp",
2206  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2207  .priv_data_size = sizeof(RTSPState),
2208  .read_probe = rtp_probe,
2209  .read_header = rtp_read_header,
2211  .read_close = sdp_read_close,
2212  .flags = AVFMT_NOFILE,
2213  .priv_class = &rtp_demuxer_class,
2214 };
2215 #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:3173
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 * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:61
Bytestream IO Context.
Definition: avio.h:68
Realmedia Data Transport.
Definition: rtsp.h:58
int size
void ff_rtsp_undo_setup(AVFormatContext *s)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:553
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:331
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1005
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
Definition: aviobuf.c:945
AVOption.
Definition: opt.h:233
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:114
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:186
const char * filename
Definition: avformat.h:339
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:3283
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:265
#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:613
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:155
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:623
char location[4096]
the "Location:" field.
Definition: rtsp.h:151
int mode_record
transport set to record data
Definition: rtsp.h:111
enum AVMediaType codec_type
Definition: rtp.c:39
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:150
void ff_network_close(void)
Definition: network.c:148
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_RTP_PORT_MAX
Definition: rtsp.h:78
#define freeaddrinfo
Definition: network.h:182
int ctx_flags
Format-specific flags, see AVFMTCTX_xx.
Definition: avformat.h:864
#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:135
AVCodec.
Definition: avcodec.h:2960
#define AI_NUMERICHOST
Definition: network.h:151
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
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...) av_printf_format(7
Assemble a URL string from components.
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:374
#define RECVBUF_SIZE
Definition: rtsp.c:59
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:151
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:828
#define RTP_PT_PRIVATE
Definition: rtp.h:76
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:147
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:206
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:396
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:38
int recvbuf_len
Definition: rtsp.h:322
Public dictionary API.
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:43
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:358
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:147
Opaque data information usually continuous.
Definition: avutil.h:181
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:123
int ff_network_init(void)
Definition: network.c:121
AVOptions.
miscellaneous OS support macros and functions.
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:92
PayloadContext *(* alloc)(void)
Allocate any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:128
int id
Format-specific stream ID.
Definition: avformat.h:629
#define POLL_TIMEOUT_MS
Definition: rtsp.c:55
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:60
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:130
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
Definition: avformat.h:876
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:413
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:42
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:130
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:88
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:451
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
static float t
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:446
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:291
struct AVOutputFormat * oformat
Definition: avformat.h:842
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:176
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:76
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:122
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
#define CONFIG_RTPDEC
Definition: config.h:348
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:176
AVDictionary * metadata
Definition: avformat.h:972
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
Definition: mem.c:139
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:347
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:249
#define AV_RB16
Definition: intreadwrite.h:53
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:88
int avio_close(AVIOContext *s)
Close the resource accessed by the AVIOContext s and free it.
Definition: aviobuf.c:782
const AVOption ff_rtsp_options[]
Definition: rtsp.c:80
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:181
URLContext * rtsp_hd
Definition: rtsp.h:219
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:146
AVStream * avformat_new_stream(AVFormatContext *s, AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:2736
const char * name
Name of the codec implementation.
Definition: avcodec.h:2967
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:330
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:493
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:177
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:67
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:109
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:74
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:418
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:641
MpegTSContext * ff_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2110
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:341
int seq
RTSP command sequence number.
Definition: rtsp.h:240
#define CONFIG_RTSP_DEMUXER
Definition: config.h:796
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:340
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:338
unsigned int nb_streams
A list of all streams in the file.
Definition: avformat.h:875
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:405
#define NI_NUMERICHOST
Definition: network.h:159
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:56
char filename[1024]
input or output filename
Definition: avformat.h:878
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:175
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes.
Definition: base64.h:59
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:140
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:54
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: <xyz>[;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
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:137
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:178
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:125
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:340
#define ENC
Definition: rtsp.c:64
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:428
int ff_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:2126
Raw data (over UDP)
Definition: rtsp.h:59
LIBAVUTIL_VERSION_INT
Definition: eval.c:52
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:431
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:424
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:166
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:536
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:37
Stream structure.
Definition: avformat.h:622
int nb_byes
Definition: rtsp.h:335
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:261
NULL
Definition: eval.c:52
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:429
enum AVMediaType codec_type
Definition: avcodec.h:1347
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
Definition: avio.c:369
enum AVCodecID codec_id
Definition: avcodec.h:1350
int rtp_port_max
Definition: rtsp.h:386
Definition: rtp.h:99
int sample_rate
samples per second
Definition: avcodec.h:2104
AVIOContext * pb
I/O context.
Definition: avformat.h:861
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:381
av_default_item_name
Definition: dnxhdenc.c:43
int server_port_max
Definition: rtsp.h:104
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
main external API structure.
Definition: avcodec.h:1339
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:1515
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:66
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
#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:117
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
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:63
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:58
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:510
#define SPACE_CHARS
Definition: internal.h:177
void * priv_data
Definition: url.h:44
struct RTSPState RTSPState
Private data for the RTSP demuxer.
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:440
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:177
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:85
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:71
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:641
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:713
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:587
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:229
#define s1
Definition: regdef.h:38
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:2662
This structure contains the data a format has to probe a file.
Definition: avformat.h:338
#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:243
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:77
int interleaved_max
Definition: rtsp.h:92
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:109
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:286
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:137
#define CONFIG_RTSP_MUXER
Definition: config.h:1130
int64_t start_time
Decoding: position of the first frame of the component, in AV_TIME_BASE fractional seconds...
Definition: avformat.h:885
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:230
#define DEC
Definition: rtsp.c:63
const OptionDef options[]
Definition: avserver.c:4665
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...
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:181
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:97
void ff_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:2151
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:149
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:97
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:203
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:437
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:258
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:77
struct AVInputFormat * iformat
Can only be iformat or oformat, not both at the same time.
Definition: avformat.h:841
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:2713
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:91
uint32_t base_timestamp
Definition: rtpdec.h:152
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:183
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:2105
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:424
void * priv_data
Format private data.
Definition: avformat.h:848
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:567
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:764
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:430
#define MAX_TIMEOUTS
Definition: rtsp.c:57
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:565
int ai_flags
Definition: network.h:102
int64_t duration
Decoding: duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:893
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:455
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get random data.
Definition: random_seed.c:85
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:669
int interleaved_max
Definition: rtsp.h:422
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:58
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:754
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:113
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:37
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:422
This structure stores compressed data.
Definition: avcodec.h:898
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:209
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:251
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:158
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:414
#define OFFSET(x)
Definition: rtsp.c:62
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:96
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:415
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