FFmpeg  2.8.17
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rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
32 #include "avformat.h"
33 #include "avio_internal.h"
34 
35 #if HAVE_POLL_H
36 #include <poll.h>
37 #endif
38 #include "internal.h"
39 #include "network.h"
40 #include "os_support.h"
41 #include "http.h"
42 #include "rtsp.h"
43 
44 #include "rtpdec.h"
45 #include "rtpproto.h"
46 #include "rdt.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
49 #include "url.h"
50 #include "rtpenc.h"
51 #include "mpegts.h"
52 
53 /* Timeout values for socket poll, in ms,
54  * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
61 
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 
66 #define RTSP_FLAG_OPTS(name, longname) \
67  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68  { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
69 
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71  { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75  { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
76 
77 #define COMMON_OPTS() \
78  { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
79  { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
80 
81 
83  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
84  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92  { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
93  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96  { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97  { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
98  COMMON_OPTS(),
99  { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
100  { NULL },
101 };
102 
103 static const AVOption sdp_options[] = {
104  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106  { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
107  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
108  COMMON_OPTS(),
109  { NULL },
110 };
111 
112 static const AVOption rtp_options[] = {
113  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
114  COMMON_OPTS(),
115  { NULL },
116 };
117 
118 
120 {
121  AVDictionary *opts = NULL;
122  char buf[256];
123 
124  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125  av_dict_set(&opts, "buffer_size", buf, 0);
126 
127  return opts;
128 }
129 
130 static void get_word_until_chars(char *buf, int buf_size,
131  const char *sep, const char **pp)
132 {
133  const char *p;
134  char *q;
135 
136  p = *pp;
137  p += strspn(p, SPACE_CHARS);
138  q = buf;
139  while (!strchr(sep, *p) && *p != '\0') {
140  if ((q - buf) < buf_size - 1)
141  *q++ = *p;
142  p++;
143  }
144  if (buf_size > 0)
145  *q = '\0';
146  *pp = p;
147 }
148 
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
150  const char **pp)
151 {
152  if (**pp == '/') (*pp)++;
153  get_word_until_chars(buf, buf_size, sep, pp);
154 }
155 
156 static void get_word(char *buf, int buf_size, const char **pp)
157 {
158  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
159 }
160 
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
162  * and end time.
163  * Used for seeking in the rtp stream.
164  */
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
166 {
167  char buf[256];
168 
169  p += strspn(p, SPACE_CHARS);
170  if (!av_stristart(p, "npt=", &p))
171  return;
172 
173  *start = AV_NOPTS_VALUE;
174  *end = AV_NOPTS_VALUE;
175 
176  get_word_sep(buf, sizeof(buf), "-", &p);
177  if (av_parse_time(start, buf, 1) < 0)
178  return;
179  if (*p == '-') {
180  p++;
181  get_word_sep(buf, sizeof(buf), "-", &p);
182  if (av_parse_time(end, buf, 1) < 0)
183  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
184  }
185 }
186 
187 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
188 {
189  struct addrinfo hints = { 0 }, *ai = NULL;
190  hints.ai_flags = AI_NUMERICHOST;
191  if (getaddrinfo(buf, NULL, &hints, &ai))
192  return -1;
193  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
194  freeaddrinfo(ai);
195  return 0;
196 }
197 
198 #if CONFIG_RTPDEC
199 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
200  RTSPStream *rtsp_st, AVStream *st)
201 {
202  AVCodecContext *codec = st ? st->codec : NULL;
203  if (!handler)
204  return;
205  if (codec)
206  codec->codec_id = handler->codec_id;
207  rtsp_st->dynamic_handler = handler;
208  if (st)
209  st->need_parsing = handler->need_parsing;
210  if (handler->priv_data_size) {
212  if (!rtsp_st->dynamic_protocol_context)
213  rtsp_st->dynamic_handler = NULL;
214  }
215 }
216 
217 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
218  AVStream *st)
219 {
220  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
221  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
222  rtsp_st->dynamic_protocol_context);
223  if (ret < 0) {
224  if (rtsp_st->dynamic_protocol_context) {
225  if (rtsp_st->dynamic_handler->close)
226  rtsp_st->dynamic_handler->close(
227  rtsp_st->dynamic_protocol_context);
229  }
230  rtsp_st->dynamic_protocol_context = NULL;
231  rtsp_st->dynamic_handler = NULL;
232  }
233  }
234 }
235 
236 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
237 static int sdp_parse_rtpmap(AVFormatContext *s,
238  AVStream *st, RTSPStream *rtsp_st,
239  int payload_type, const char *p)
240 {
241  AVCodecContext *codec = st->codec;
242  char buf[256];
243  int i;
244  AVCodec *c;
245  const char *c_name;
246 
247  /* See if we can handle this kind of payload.
248  * The space should normally not be there but some Real streams or
249  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
250  * have a trailing space. */
251  get_word_sep(buf, sizeof(buf), "/ ", &p);
252  if (payload_type < RTP_PT_PRIVATE) {
253  /* We are in a standard case
254  * (from http://www.iana.org/assignments/rtp-parameters). */
255  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
256  }
257 
258  if (codec->codec_id == AV_CODEC_ID_NONE) {
259  RTPDynamicProtocolHandler *handler =
261  init_rtp_handler(handler, rtsp_st, st);
262  /* If no dynamic handler was found, check with the list of standard
263  * allocated types, if such a stream for some reason happens to
264  * use a private payload type. This isn't handled in rtpdec.c, since
265  * the format name from the rtpmap line never is passed into rtpdec. */
266  if (!rtsp_st->dynamic_handler)
267  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
268  }
269 
270  c = avcodec_find_decoder(codec->codec_id);
271  if (c && c->name)
272  c_name = c->name;
273  else
274  c_name = "(null)";
275 
276  get_word_sep(buf, sizeof(buf), "/", &p);
277  i = atoi(buf);
278  switch (codec->codec_type) {
279  case AVMEDIA_TYPE_AUDIO:
280  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
283  if (i > 0) {
284  codec->sample_rate = i;
285  avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
286  get_word_sep(buf, sizeof(buf), "/", &p);
287  i = atoi(buf);
288  if (i > 0)
289  codec->channels = i;
290  }
291  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
292  codec->sample_rate);
293  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
294  codec->channels);
295  break;
296  case AVMEDIA_TYPE_VIDEO:
297  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
298  if (i > 0)
299  avpriv_set_pts_info(st, 32, 1, i);
300  break;
301  default:
302  break;
303  }
304  finalize_rtp_handler_init(s, rtsp_st, st);
305  return 0;
306 }
307 
308 /* parse the attribute line from the fmtp a line of an sdp response. This
309  * is broken out as a function because it is used in rtp_h264.c, which is
310  * forthcoming. */
311 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
312  char *value, int value_size)
313 {
314  *p += strspn(*p, SPACE_CHARS);
315  if (**p) {
316  get_word_sep(attr, attr_size, "=", p);
317  if (**p == '=')
318  (*p)++;
319  get_word_sep(value, value_size, ";", p);
320  if (**p == ';')
321  (*p)++;
322  return 1;
323  }
324  return 0;
325 }
326 
327 typedef struct SDPParseState {
328  /* SDP only */
329  struct sockaddr_storage default_ip;
330  int default_ttl;
331  int skip_media; ///< set if an unknown m= line occurs
332  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
333  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
334  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
335  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
336  int seen_rtpmap;
337  int seen_fmtp;
338  char delayed_fmtp[2048];
339 } SDPParseState;
340 
341 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
342  struct RTSPSource ***dest, int *dest_count)
343 {
344  RTSPSource *rtsp_src, *rtsp_src2;
345  int i;
346  for (i = 0; i < count; i++) {
347  rtsp_src = addrs[i];
348  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
349  if (!rtsp_src2)
350  continue;
351  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
352  dynarray_add(dest, dest_count, rtsp_src2);
353  }
354 }
355 
356 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
357  int payload_type, const char *line)
358 {
359  int i;
360 
361  for (i = 0; i < rt->nb_rtsp_streams; i++) {
362  RTSPStream *rtsp_st = rt->rtsp_streams[i];
363  if (rtsp_st->sdp_payload_type == payload_type &&
364  rtsp_st->dynamic_handler &&
365  rtsp_st->dynamic_handler->parse_sdp_a_line) {
366  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
367  rtsp_st->dynamic_protocol_context, line);
368  }
369  }
370 }
371 
372 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
373  int letter, const char *buf)
374 {
375  RTSPState *rt = s->priv_data;
376  char buf1[64], st_type[64];
377  const char *p;
378  enum AVMediaType codec_type;
379  int payload_type;
380  AVStream *st;
381  RTSPStream *rtsp_st;
382  RTSPSource *rtsp_src;
383  struct sockaddr_storage sdp_ip;
384  int ttl;
385 
386  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
387 
388  p = buf;
389  if (s1->skip_media && letter != 'm')
390  return;
391  switch (letter) {
392  case 'c':
393  get_word(buf1, sizeof(buf1), &p);
394  if (strcmp(buf1, "IN") != 0)
395  return;
396  get_word(buf1, sizeof(buf1), &p);
397  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
398  return;
399  get_word_sep(buf1, sizeof(buf1), "/", &p);
400  if (get_sockaddr(buf1, &sdp_ip))
401  return;
402  ttl = 16;
403  if (*p == '/') {
404  p++;
405  get_word_sep(buf1, sizeof(buf1), "/", &p);
406  ttl = atoi(buf1);
407  }
408  if (s->nb_streams == 0) {
409  s1->default_ip = sdp_ip;
410  s1->default_ttl = ttl;
411  } else {
412  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
413  rtsp_st->sdp_ip = sdp_ip;
414  rtsp_st->sdp_ttl = ttl;
415  }
416  break;
417  case 's':
418  av_dict_set(&s->metadata, "title", p, 0);
419  break;
420  case 'i':
421  if (s->nb_streams == 0) {
422  av_dict_set(&s->metadata, "comment", p, 0);
423  break;
424  }
425  break;
426  case 'm':
427  /* new stream */
428  s1->skip_media = 0;
429  s1->seen_fmtp = 0;
430  s1->seen_rtpmap = 0;
431  codec_type = AVMEDIA_TYPE_UNKNOWN;
432  get_word(st_type, sizeof(st_type), &p);
433  if (!strcmp(st_type, "audio")) {
434  codec_type = AVMEDIA_TYPE_AUDIO;
435  } else if (!strcmp(st_type, "video")) {
436  codec_type = AVMEDIA_TYPE_VIDEO;
437  } else if (!strcmp(st_type, "application")) {
438  codec_type = AVMEDIA_TYPE_DATA;
439  } else if (!strcmp(st_type, "text")) {
440  codec_type = AVMEDIA_TYPE_SUBTITLE;
441  }
442  if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
443  !(rt->media_type_mask & (1 << codec_type)) ||
444  rt->nb_rtsp_streams >= s->max_streams
445  ) {
446  s1->skip_media = 1;
447  return;
448  }
449  rtsp_st = av_mallocz(sizeof(RTSPStream));
450  if (!rtsp_st)
451  return;
452  rtsp_st->stream_index = -1;
453  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
454 
455  rtsp_st->sdp_ip = s1->default_ip;
456  rtsp_st->sdp_ttl = s1->default_ttl;
457 
458  copy_default_source_addrs(s1->default_include_source_addrs,
459  s1->nb_default_include_source_addrs,
460  &rtsp_st->include_source_addrs,
461  &rtsp_st->nb_include_source_addrs);
462  copy_default_source_addrs(s1->default_exclude_source_addrs,
463  s1->nb_default_exclude_source_addrs,
464  &rtsp_st->exclude_source_addrs,
465  &rtsp_st->nb_exclude_source_addrs);
466 
467  get_word(buf1, sizeof(buf1), &p); /* port */
468  rtsp_st->sdp_port = atoi(buf1);
469 
470  get_word(buf1, sizeof(buf1), &p); /* protocol */
471  if (!strcmp(buf1, "udp"))
473  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
474  rtsp_st->feedback = 1;
475 
476  /* XXX: handle list of formats */
477  get_word(buf1, sizeof(buf1), &p); /* format list */
478  rtsp_st->sdp_payload_type = atoi(buf1);
479 
480  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
481  /* no corresponding stream */
482  if (rt->transport == RTSP_TRANSPORT_RAW) {
483  if (CONFIG_RTPDEC && !rt->ts)
484  rt->ts = avpriv_mpegts_parse_open(s);
485  } else {
487  handler = ff_rtp_handler_find_by_id(
489  init_rtp_handler(handler, rtsp_st, NULL);
490  finalize_rtp_handler_init(s, rtsp_st, NULL);
491  }
492  } else if (rt->server_type == RTSP_SERVER_WMS &&
493  codec_type == AVMEDIA_TYPE_DATA) {
494  /* RTX stream, a stream that carries all the other actual
495  * audio/video streams. Don't expose this to the callers. */
496  } else {
497  st = avformat_new_stream(s, NULL);
498  if (!st)
499  return;
500  st->id = rt->nb_rtsp_streams - 1;
501  rtsp_st->stream_index = st->index;
502  st->codec->codec_type = codec_type;
503  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
505  /* if standard payload type, we can find the codec right now */
507  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
508  st->codec->sample_rate > 0)
509  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
510  /* Even static payload types may need a custom depacketizer */
511  handler = ff_rtp_handler_find_by_id(
512  rtsp_st->sdp_payload_type, st->codec->codec_type);
513  init_rtp_handler(handler, rtsp_st, st);
514  finalize_rtp_handler_init(s, rtsp_st, st);
515  }
516  if (rt->default_lang[0])
517  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
518  }
519  /* put a default control url */
520  av_strlcpy(rtsp_st->control_url, rt->control_uri,
521  sizeof(rtsp_st->control_url));
522  break;
523  case 'a':
524  if (av_strstart(p, "control:", &p)) {
525  if (s->nb_streams == 0) {
526  if (!strncmp(p, "rtsp://", 7))
527  av_strlcpy(rt->control_uri, p,
528  sizeof(rt->control_uri));
529  } else {
530  char proto[32];
531  /* get the control url */
532  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
533 
534  /* XXX: may need to add full url resolution */
535  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
536  NULL, NULL, 0, p);
537  if (proto[0] == '\0') {
538  /* relative control URL */
539  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
540  av_strlcat(rtsp_st->control_url, "/",
541  sizeof(rtsp_st->control_url));
542  av_strlcat(rtsp_st->control_url, p,
543  sizeof(rtsp_st->control_url));
544  } else
545  av_strlcpy(rtsp_st->control_url, p,
546  sizeof(rtsp_st->control_url));
547  }
548  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
549  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
550  get_word(buf1, sizeof(buf1), &p);
551  payload_type = atoi(buf1);
552  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
553  if (rtsp_st->stream_index >= 0) {
554  st = s->streams[rtsp_st->stream_index];
555  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
556  }
557  s1->seen_rtpmap = 1;
558  if (s1->seen_fmtp) {
559  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
560  }
561  } else if (av_strstart(p, "fmtp:", &p) ||
562  av_strstart(p, "framesize:", &p)) {
563  // let dynamic protocol handlers have a stab at the line.
564  get_word(buf1, sizeof(buf1), &p);
565  payload_type = atoi(buf1);
566  if (s1->seen_rtpmap) {
567  parse_fmtp(s, rt, payload_type, buf);
568  } else {
569  s1->seen_fmtp = 1;
570  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
571  }
572  } else if (av_strstart(p, "range:", &p)) {
573  int64_t start, end;
574 
575  // this is so that seeking on a streamed file can work.
576  rtsp_parse_range_npt(p, &start, &end);
577  s->start_time = start;
578  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
579  s->duration = (end == AV_NOPTS_VALUE) ?
580  AV_NOPTS_VALUE : end - start;
581  } else if (av_strstart(p, "lang:", &p)) {
582  if (s->nb_streams > 0) {
583  get_word(buf1, sizeof(buf1), &p);
584  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
585  if (rtsp_st->stream_index >= 0) {
586  st = s->streams[rtsp_st->stream_index];
587  av_dict_set(&st->metadata, "language", buf1, 0);
588  }
589  } else
590  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
591  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
592  if (atoi(p) == 1)
594  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
595  s->nb_streams > 0) {
596  st = s->streams[s->nb_streams - 1];
597  st->codec->sample_rate = atoi(p);
598  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
599  // RFC 4568
600  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
601  get_word(buf1, sizeof(buf1), &p); // ignore tag
602  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
603  p += strspn(p, SPACE_CHARS);
604  if (av_strstart(p, "inline:", &p))
605  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
606  } else if (av_strstart(p, "source-filter:", &p)) {
607  int exclude = 0;
608  get_word(buf1, sizeof(buf1), &p);
609  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
610  return;
611  exclude = !strcmp(buf1, "excl");
612 
613  get_word(buf1, sizeof(buf1), &p);
614  if (strcmp(buf1, "IN") != 0)
615  return;
616  get_word(buf1, sizeof(buf1), &p);
617  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
618  return;
619  // not checking that the destination address actually matches or is wildcard
620  get_word(buf1, sizeof(buf1), &p);
621 
622  while (*p != '\0') {
623  rtsp_src = av_mallocz(sizeof(*rtsp_src));
624  if (!rtsp_src)
625  return;
626  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
627  if (exclude) {
628  if (s->nb_streams == 0) {
629  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
630  } else {
631  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
632  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
633  }
634  } else {
635  if (s->nb_streams == 0) {
636  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
637  } else {
638  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
639  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
640  }
641  }
642  }
643  } else {
644  if (rt->server_type == RTSP_SERVER_WMS)
646  if (s->nb_streams > 0) {
647  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
648 
649  if (rt->server_type == RTSP_SERVER_REAL)
650  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
651 
652  if (rtsp_st->dynamic_handler &&
654  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
655  rtsp_st->stream_index,
656  rtsp_st->dynamic_protocol_context, buf);
657  }
658  }
659  break;
660  }
661 }
662 
663 int ff_sdp_parse(AVFormatContext *s, const char *content)
664 {
665  RTSPState *rt = s->priv_data;
666  const char *p;
667  int letter, i;
668  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
669  * contain long SDP lines containing complete ASF Headers (several
670  * kB) or arrays of MDPR (RM stream descriptor) headers plus
671  * "rulebooks" describing their properties. Therefore, the SDP line
672  * buffer is large.
673  *
674  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
675  * in rtpdec_xiph.c. */
676  char buf[16384], *q;
677  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
678 
679  p = content;
680  for (;;) {
681  p += strspn(p, SPACE_CHARS);
682  letter = *p;
683  if (letter == '\0')
684  break;
685  p++;
686  if (*p != '=')
687  goto next_line;
688  p++;
689  /* get the content */
690  q = buf;
691  while (*p != '\n' && *p != '\r' && *p != '\0') {
692  if ((q - buf) < sizeof(buf) - 1)
693  *q++ = *p;
694  p++;
695  }
696  *q = '\0';
697  sdp_parse_line(s, s1, letter, buf);
698  next_line:
699  while (*p != '\n' && *p != '\0')
700  p++;
701  if (*p == '\n')
702  p++;
703  }
704 
705  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
706  av_freep(&s1->default_include_source_addrs[i]);
707  av_freep(&s1->default_include_source_addrs);
708  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
709  av_freep(&s1->default_exclude_source_addrs[i]);
710  av_freep(&s1->default_exclude_source_addrs);
711 
712  rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
713  if (!rt->p) return AVERROR(ENOMEM);
714  return 0;
715 }
716 #endif /* CONFIG_RTPDEC */
717 
718 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
719 {
720  RTSPState *rt = s->priv_data;
721  int i;
722 
723  for (i = 0; i < rt->nb_rtsp_streams; i++) {
724  RTSPStream *rtsp_st = rt->rtsp_streams[i];
725  if (!rtsp_st)
726  continue;
727  if (rtsp_st->transport_priv) {
728  if (s->oformat) {
729  AVFormatContext *rtpctx = rtsp_st->transport_priv;
730  av_write_trailer(rtpctx);
732  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
733  ff_rtsp_tcp_write_packet(s, rtsp_st);
734  ffio_free_dyn_buf(&rtpctx->pb);
735  } else {
736  avio_closep(&rtpctx->pb);
737  }
738  avformat_free_context(rtpctx);
739  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
741  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
743  }
744  rtsp_st->transport_priv = NULL;
745  if (rtsp_st->rtp_handle)
746  ffurl_close(rtsp_st->rtp_handle);
747  rtsp_st->rtp_handle = NULL;
748  }
749 }
750 
751 /* close and free RTSP streams */
753 {
754  RTSPState *rt = s->priv_data;
755  int i, j;
756  RTSPStream *rtsp_st;
757 
758  ff_rtsp_undo_setup(s, 0);
759  for (i = 0; i < rt->nb_rtsp_streams; i++) {
760  rtsp_st = rt->rtsp_streams[i];
761  if (rtsp_st) {
762  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
763  if (rtsp_st->dynamic_handler->close)
764  rtsp_st->dynamic_handler->close(
765  rtsp_st->dynamic_protocol_context);
767  }
768  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
769  av_freep(&rtsp_st->include_source_addrs[j]);
770  av_freep(&rtsp_st->include_source_addrs);
771  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
772  av_freep(&rtsp_st->exclude_source_addrs[j]);
773  av_freep(&rtsp_st->exclude_source_addrs);
774 
775  av_freep(&rtsp_st);
776  }
777  }
778  av_freep(&rt->rtsp_streams);
779  if (rt->asf_ctx) {
781  }
782  if (CONFIG_RTPDEC && rt->ts)
784  av_freep(&rt->p);
785  av_freep(&rt->recvbuf);
786 }
787 
789 {
790  RTSPState *rt = s->priv_data;
791  AVStream *st = NULL;
792  int reordering_queue_size = rt->reordering_queue_size;
793  if (reordering_queue_size < 0) {
795  reordering_queue_size = 0;
796  else
797  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
798  }
799 
800  /* open the RTP context */
801  if (rtsp_st->stream_index >= 0)
802  st = s->streams[rtsp_st->stream_index];
803  if (!st)
805 
806  if (CONFIG_RTSP_MUXER && s->oformat && st) {
807  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
808  s, st, rtsp_st->rtp_handle,
810  rtsp_st->stream_index);
811  /* Ownership of rtp_handle is passed to the rtp mux context */
812  rtsp_st->rtp_handle = NULL;
813  if (ret < 0)
814  return ret;
815  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
816  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
817  return 0; // Don't need to open any parser here
818  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
819  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
820  rtsp_st->dynamic_protocol_context,
821  rtsp_st->dynamic_handler);
822  else if (CONFIG_RTPDEC)
823  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
824  rtsp_st->sdp_payload_type,
825  reordering_queue_size);
826 
827  if (!rtsp_st->transport_priv) {
828  return AVERROR(ENOMEM);
829  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
830  if (rtsp_st->dynamic_handler) {
832  rtsp_st->dynamic_protocol_context,
833  rtsp_st->dynamic_handler);
834  }
835  if (rtsp_st->crypto_suite[0])
837  rtsp_st->crypto_suite,
838  rtsp_st->crypto_params);
839  }
840 
841  return 0;
842 }
843 
844 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
845 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
846 {
847  const char *q;
848  char *p;
849  int v;
850 
851  q = *pp;
852  q += strspn(q, SPACE_CHARS);
853  v = strtol(q, &p, 10);
854  if (*p == '-') {
855  p++;
856  *min_ptr = v;
857  v = strtol(p, &p, 10);
858  *max_ptr = v;
859  } else {
860  *min_ptr = v;
861  *max_ptr = v;
862  }
863  *pp = p;
864 }
865 
866 /* XXX: only one transport specification is parsed */
867 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
868 {
869  char transport_protocol[16];
870  char profile[16];
871  char lower_transport[16];
872  char parameter[16];
874  char buf[256];
875 
876  reply->nb_transports = 0;
877 
878  for (;;) {
879  p += strspn(p, SPACE_CHARS);
880  if (*p == '\0')
881  break;
882 
883  th = &reply->transports[reply->nb_transports];
884 
885  get_word_sep(transport_protocol, sizeof(transport_protocol),
886  "/", &p);
887  if (!av_strcasecmp (transport_protocol, "rtp")) {
888  get_word_sep(profile, sizeof(profile), "/;,", &p);
889  lower_transport[0] = '\0';
890  /* rtp/avp/<protocol> */
891  if (*p == '/') {
892  get_word_sep(lower_transport, sizeof(lower_transport),
893  ";,", &p);
894  }
896  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
897  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
898  /* x-pn-tng/<protocol> */
899  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
900  profile[0] = '\0';
902  } else if (!av_strcasecmp(transport_protocol, "raw")) {
903  get_word_sep(profile, sizeof(profile), "/;,", &p);
904  lower_transport[0] = '\0';
905  /* raw/raw/<protocol> */
906  if (*p == '/') {
907  get_word_sep(lower_transport, sizeof(lower_transport),
908  ";,", &p);
909  }
911  }
912  if (!av_strcasecmp(lower_transport, "TCP"))
914  else
916 
917  if (*p == ';')
918  p++;
919  /* get each parameter */
920  while (*p != '\0' && *p != ',') {
921  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
922  if (!strcmp(parameter, "port")) {
923  if (*p == '=') {
924  p++;
925  rtsp_parse_range(&th->port_min, &th->port_max, &p);
926  }
927  } else if (!strcmp(parameter, "client_port")) {
928  if (*p == '=') {
929  p++;
930  rtsp_parse_range(&th->client_port_min,
931  &th->client_port_max, &p);
932  }
933  } else if (!strcmp(parameter, "server_port")) {
934  if (*p == '=') {
935  p++;
936  rtsp_parse_range(&th->server_port_min,
937  &th->server_port_max, &p);
938  }
939  } else if (!strcmp(parameter, "interleaved")) {
940  if (*p == '=') {
941  p++;
942  rtsp_parse_range(&th->interleaved_min,
943  &th->interleaved_max, &p);
944  }
945  } else if (!strcmp(parameter, "multicast")) {
948  } else if (!strcmp(parameter, "ttl")) {
949  if (*p == '=') {
950  char *end;
951  p++;
952  th->ttl = strtol(p, &end, 10);
953  p = end;
954  }
955  } else if (!strcmp(parameter, "destination")) {
956  if (*p == '=') {
957  p++;
958  get_word_sep(buf, sizeof(buf), ";,", &p);
959  get_sockaddr(buf, &th->destination);
960  }
961  } else if (!strcmp(parameter, "source")) {
962  if (*p == '=') {
963  p++;
964  get_word_sep(buf, sizeof(buf), ";,", &p);
965  av_strlcpy(th->source, buf, sizeof(th->source));
966  }
967  } else if (!strcmp(parameter, "mode")) {
968  if (*p == '=') {
969  p++;
970  get_word_sep(buf, sizeof(buf), ";, ", &p);
971  if (!strcmp(buf, "record") ||
972  !strcmp(buf, "receive"))
973  th->mode_record = 1;
974  }
975  }
976 
977  while (*p != ';' && *p != '\0' && *p != ',')
978  p++;
979  if (*p == ';')
980  p++;
981  }
982  if (*p == ',')
983  p++;
984 
985  reply->nb_transports++;
986  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
987  break;
988  }
989 }
990 
991 static void handle_rtp_info(RTSPState *rt, const char *url,
992  uint32_t seq, uint32_t rtptime)
993 {
994  int i;
995  if (!rtptime || !url[0])
996  return;
997  if (rt->transport != RTSP_TRANSPORT_RTP)
998  return;
999  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1000  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1001  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1002  if (!rtpctx)
1003  continue;
1004  if (!strcmp(rtsp_st->control_url, url)) {
1005  rtpctx->base_timestamp = rtptime;
1006  break;
1007  }
1008  }
1009 }
1010 
1011 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1012 {
1013  int read = 0;
1014  char key[20], value[1024], url[1024] = "";
1015  uint32_t seq = 0, rtptime = 0;
1016 
1017  for (;;) {
1018  p += strspn(p, SPACE_CHARS);
1019  if (!*p)
1020  break;
1021  get_word_sep(key, sizeof(key), "=", &p);
1022  if (*p != '=')
1023  break;
1024  p++;
1025  get_word_sep(value, sizeof(value), ";, ", &p);
1026  read++;
1027  if (!strcmp(key, "url"))
1028  av_strlcpy(url, value, sizeof(url));
1029  else if (!strcmp(key, "seq"))
1030  seq = strtoul(value, NULL, 10);
1031  else if (!strcmp(key, "rtptime"))
1032  rtptime = strtoul(value, NULL, 10);
1033  if (*p == ',') {
1034  handle_rtp_info(rt, url, seq, rtptime);
1035  url[0] = '\0';
1036  seq = rtptime = 0;
1037  read = 0;
1038  }
1039  if (*p)
1040  p++;
1041  }
1042  if (read > 0)
1043  handle_rtp_info(rt, url, seq, rtptime);
1044 }
1045 
1046 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1047  RTSPState *rt, const char *method)
1048 {
1049  const char *p;
1050 
1051  /* NOTE: we do case independent match for broken servers */
1052  p = buf;
1053  if (av_stristart(p, "Session:", &p)) {
1054  int t;
1055  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1056  if (av_stristart(p, ";timeout=", &p) &&
1057  (t = strtol(p, NULL, 10)) > 0) {
1058  reply->timeout = t;
1059  }
1060  } else if (av_stristart(p, "Content-Length:", &p)) {
1061  reply->content_length = strtol(p, NULL, 10);
1062  } else if (av_stristart(p, "Transport:", &p)) {
1063  rtsp_parse_transport(reply, p);
1064  } else if (av_stristart(p, "CSeq:", &p)) {
1065  reply->seq = strtol(p, NULL, 10);
1066  } else if (av_stristart(p, "Range:", &p)) {
1067  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1068  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1069  p += strspn(p, SPACE_CHARS);
1070  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1071  } else if (av_stristart(p, "Server:", &p)) {
1072  p += strspn(p, SPACE_CHARS);
1073  av_strlcpy(reply->server, p, sizeof(reply->server));
1074  } else if (av_stristart(p, "Notice:", &p) ||
1075  av_stristart(p, "X-Notice:", &p)) {
1076  reply->notice = strtol(p, NULL, 10);
1077  } else if (av_stristart(p, "Location:", &p)) {
1078  p += strspn(p, SPACE_CHARS);
1079  av_strlcpy(reply->location, p , sizeof(reply->location));
1080  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1081  p += strspn(p, SPACE_CHARS);
1082  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1083  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1084  p += strspn(p, SPACE_CHARS);
1085  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1086  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1087  p += strspn(p, SPACE_CHARS);
1088  if (method && !strcmp(method, "DESCRIBE"))
1089  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1090  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1091  p += strspn(p, SPACE_CHARS);
1092  if (method && !strcmp(method, "PLAY"))
1093  rtsp_parse_rtp_info(rt, p);
1094  } else if (av_stristart(p, "Public:", &p) && rt) {
1095  if (strstr(p, "GET_PARAMETER") &&
1096  method && !strcmp(method, "OPTIONS"))
1097  rt->get_parameter_supported = 1;
1098  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1099  p += strspn(p, SPACE_CHARS);
1100  rt->accept_dynamic_rate = atoi(p);
1101  } else if (av_stristart(p, "Content-Type:", &p)) {
1102  p += strspn(p, SPACE_CHARS);
1103  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1104  }
1105 }
1106 
1107 /* skip a RTP/TCP interleaved packet */
1109 {
1110  RTSPState *rt = s->priv_data;
1111  int ret, len, len1;
1112  uint8_t buf[1024];
1113 
1114  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1115  if (ret != 3)
1116  return;
1117  len = AV_RB16(buf + 1);
1118 
1119  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1120 
1121  /* skip payload */
1122  while (len > 0) {
1123  len1 = len;
1124  if (len1 > sizeof(buf))
1125  len1 = sizeof(buf);
1126  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1127  if (ret != len1)
1128  return;
1129  len -= len1;
1130  }
1131 }
1132 
1134  unsigned char **content_ptr,
1135  int return_on_interleaved_data, const char *method)
1136 {
1137  RTSPState *rt = s->priv_data;
1138  char buf[4096], buf1[1024], *q;
1139  unsigned char ch;
1140  const char *p;
1141  int ret, content_length, line_count = 0, request = 0;
1142  unsigned char *content = NULL;
1143 
1144 start:
1145  line_count = 0;
1146  request = 0;
1147  content = NULL;
1148  memset(reply, 0, sizeof(*reply));
1149 
1150  /* parse reply (XXX: use buffers) */
1151  rt->last_reply[0] = '\0';
1152  for (;;) {
1153  q = buf;
1154  for (;;) {
1155  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1156  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1157  if (ret != 1)
1158  return AVERROR_EOF;
1159  if (ch == '\n')
1160  break;
1161  if (ch == '$' && q == buf) {
1162  if (return_on_interleaved_data) {
1163  return 1;
1164  } else
1166  } else if (ch != '\r') {
1167  if ((q - buf) < sizeof(buf) - 1)
1168  *q++ = ch;
1169  }
1170  }
1171  *q = '\0';
1172 
1173  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1174 
1175  /* test if last line */
1176  if (buf[0] == '\0')
1177  break;
1178  p = buf;
1179  if (line_count == 0) {
1180  /* get reply code */
1181  get_word(buf1, sizeof(buf1), &p);
1182  if (!strncmp(buf1, "RTSP/", 5)) {
1183  get_word(buf1, sizeof(buf1), &p);
1184  reply->status_code = atoi(buf1);
1185  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1186  } else {
1187  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1188  get_word(buf1, sizeof(buf1), &p); // object
1189  request = 1;
1190  }
1191  } else {
1192  ff_rtsp_parse_line(reply, p, rt, method);
1193  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1194  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1195  }
1196  line_count++;
1197  }
1198 
1199  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1200  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1201 
1202  content_length = reply->content_length;
1203  if (content_length > 0) {
1204  /* leave some room for a trailing '\0' (useful for simple parsing) */
1205  content = av_malloc(content_length + 1);
1206  if (!content)
1207  return AVERROR(ENOMEM);
1208  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1209  content[content_length] = '\0';
1210  }
1211  if (content_ptr)
1212  *content_ptr = content;
1213  else
1214  av_freep(&content);
1215 
1216  if (request) {
1217  char buf[1024];
1218  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1219  const char* ptr = buf;
1220 
1221  if (!strcmp(reply->reason, "OPTIONS")) {
1222  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1223  if (reply->seq)
1224  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1225  if (reply->session_id[0])
1226  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1227  reply->session_id);
1228  } else {
1229  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1230  }
1231  av_strlcat(buf, "\r\n", sizeof(buf));
1232 
1233  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1234  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1235  ptr = base64buf;
1236  }
1237  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1238 
1240  /* Even if the request from the server had data, it is not the data
1241  * that the caller wants or expects. The memory could also be leaked
1242  * if the actual following reply has content data. */
1243  if (content_ptr)
1244  av_freep(content_ptr);
1245  /* If method is set, this is called from ff_rtsp_send_cmd,
1246  * where a reply to exactly this request is awaited. For
1247  * callers from within packet receiving, we just want to
1248  * return to the caller and go back to receiving packets. */
1249  if (method)
1250  goto start;
1251  return 0;
1252  }
1253 
1254  if (rt->seq != reply->seq) {
1255  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1256  rt->seq, reply->seq);
1257  }
1258 
1259  /* EOS */
1260  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1261  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1262  reply->notice == 2306 /* Continuous Feed Terminated */) {
1263  rt->state = RTSP_STATE_IDLE;
1264  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1265  return AVERROR(EIO); /* data or server error */
1266  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1267  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1268  return AVERROR(EPERM);
1269 
1270  return 0;
1271 }
1272 
1273 /**
1274  * Send a command to the RTSP server without waiting for the reply.
1275  *
1276  * @param s RTSP (de)muxer context
1277  * @param method the method for the request
1278  * @param url the target url for the request
1279  * @param headers extra header lines to include in the request
1280  * @param send_content if non-null, the data to send as request body content
1281  * @param send_content_length the length of the send_content data, or 0 if
1282  * send_content is null
1283  *
1284  * @return zero if success, nonzero otherwise
1285  */
1286 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1287  const char *method, const char *url,
1288  const char *headers,
1289  const unsigned char *send_content,
1290  int send_content_length)
1291 {
1292  RTSPState *rt = s->priv_data;
1293  char buf[4096], *out_buf;
1294  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1295 
1296  /* Add in RTSP headers */
1297  out_buf = buf;
1298  rt->seq++;
1299  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1300  if (headers)
1301  av_strlcat(buf, headers, sizeof(buf));
1302  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1303  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1304  if (rt->session_id[0] != '\0' && (!headers ||
1305  !strstr(headers, "\nIf-Match:"))) {
1306  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1307  }
1308  if (rt->auth[0]) {
1309  char *str = ff_http_auth_create_response(&rt->auth_state,
1310  rt->auth, url, method);
1311  if (str)
1312  av_strlcat(buf, str, sizeof(buf));
1313  av_free(str);
1314  }
1315  if (send_content_length > 0 && send_content)
1316  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1317  av_strlcat(buf, "\r\n", sizeof(buf));
1318 
1319  /* base64 encode rtsp if tunneling */
1320  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1321  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1322  out_buf = base64buf;
1323  }
1324 
1325  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1326 
1327  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1328  if (send_content_length > 0 && send_content) {
1329  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1330  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1331  "with content data not supported\n");
1332  return AVERROR_PATCHWELCOME;
1333  }
1334  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1335  }
1337 
1338  return 0;
1339 }
1340 
1341 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1342  const char *url, const char *headers)
1343 {
1344  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1345 }
1346 
1347 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1348  const char *headers, RTSPMessageHeader *reply,
1349  unsigned char **content_ptr)
1350 {
1351  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1352  content_ptr, NULL, 0);
1353 }
1354 
1356  const char *method, const char *url,
1357  const char *header,
1358  RTSPMessageHeader *reply,
1359  unsigned char **content_ptr,
1360  const unsigned char *send_content,
1361  int send_content_length)
1362 {
1363  RTSPState *rt = s->priv_data;
1364  HTTPAuthType cur_auth_type;
1365  int ret, attempts = 0;
1366 
1367 retry:
1368  cur_auth_type = rt->auth_state.auth_type;
1369  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1370  send_content,
1371  send_content_length)))
1372  return ret;
1373 
1374  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1375  return ret;
1376  attempts++;
1377 
1378  if (reply->status_code == 401 &&
1379  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1380  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1381  goto retry;
1382 
1383  if (reply->status_code > 400){
1384  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1385  method,
1386  reply->status_code,
1387  reply->reason);
1388  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1389  }
1390 
1391  return 0;
1392 }
1393 
1394 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1395  int lower_transport, const char *real_challenge)
1396 {
1397  RTSPState *rt = s->priv_data;
1398  int rtx = 0, j, i, err, interleave = 0, port_off;
1399  RTSPStream *rtsp_st;
1400  RTSPMessageHeader reply1, *reply = &reply1;
1401  char cmd[2048];
1402  const char *trans_pref;
1403 
1404  if (rt->transport == RTSP_TRANSPORT_RDT)
1405  trans_pref = "x-pn-tng";
1406  else if (rt->transport == RTSP_TRANSPORT_RAW)
1407  trans_pref = "RAW/RAW";
1408  else
1409  trans_pref = "RTP/AVP";
1410 
1411  /* default timeout: 1 minute */
1412  rt->timeout = 60;
1413 
1414  /* Choose a random starting offset within the first half of the
1415  * port range, to allow for a number of ports to try even if the offset
1416  * happens to be at the end of the random range. */
1417  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1418  /* even random offset */
1419  port_off -= port_off & 0x01;
1420 
1421  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1422  char transport[2048];
1423 
1424  /*
1425  * WMS serves all UDP data over a single connection, the RTX, which
1426  * isn't necessarily the first in the SDP but has to be the first
1427  * to be set up, else the second/third SETUP will fail with a 461.
1428  */
1429  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1430  rt->server_type == RTSP_SERVER_WMS) {
1431  if (i == 0) {
1432  /* rtx first */
1433  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1434  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1435  if (len >= 4 &&
1436  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1437  "/rtx"))
1438  break;
1439  }
1440  if (rtx == rt->nb_rtsp_streams)
1441  return -1; /* no RTX found */
1442  rtsp_st = rt->rtsp_streams[rtx];
1443  } else
1444  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1445  } else
1446  rtsp_st = rt->rtsp_streams[i];
1447 
1448  /* RTP/UDP */
1449  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1450  char buf[256];
1451 
1452  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1453  port = reply->transports[0].client_port_min;
1454  goto have_port;
1455  }
1456 
1457  /* first try in specified port range */
1458  while (j <= rt->rtp_port_max) {
1459  AVDictionary *opts = map_to_opts(rt);
1460 
1461  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1462  "?localport=%d", j);
1463  /* we will use two ports per rtp stream (rtp and rtcp) */
1464  j += 2;
1465  err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1466  &s->interrupt_callback, &opts);
1467 
1468  av_dict_free(&opts);
1469 
1470  if (!err)
1471  goto rtp_opened;
1472  }
1473  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1474  err = AVERROR(EIO);
1475  goto fail;
1476 
1477  rtp_opened:
1478  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1479  have_port:
1480  snprintf(transport, sizeof(transport) - 1,
1481  "%s/UDP;", trans_pref);
1482  if (rt->server_type != RTSP_SERVER_REAL)
1483  av_strlcat(transport, "unicast;", sizeof(transport));
1484  av_strlcatf(transport, sizeof(transport),
1485  "client_port=%d", port);
1486  if (rt->transport == RTSP_TRANSPORT_RTP &&
1487  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1488  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1489  }
1490 
1491  /* RTP/TCP */
1492  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1493  /* For WMS streams, the application streams are only used for
1494  * UDP. When trying to set it up for TCP streams, the server
1495  * will return an error. Therefore, we skip those streams. */
1496  if (rt->server_type == RTSP_SERVER_WMS &&
1497  (rtsp_st->stream_index < 0 ||
1498  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1500  continue;
1501  snprintf(transport, sizeof(transport) - 1,
1502  "%s/TCP;", trans_pref);
1503  if (rt->transport != RTSP_TRANSPORT_RDT)
1504  av_strlcat(transport, "unicast;", sizeof(transport));
1505  av_strlcatf(transport, sizeof(transport),
1506  "interleaved=%d-%d",
1507  interleave, interleave + 1);
1508  interleave += 2;
1509  }
1510 
1511  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1512  snprintf(transport, sizeof(transport) - 1,
1513  "%s/UDP;multicast", trans_pref);
1514  }
1515  if (s->oformat) {
1516  av_strlcat(transport, ";mode=record", sizeof(transport));
1517  } else if (rt->server_type == RTSP_SERVER_REAL ||
1519  av_strlcat(transport, ";mode=play", sizeof(transport));
1520  snprintf(cmd, sizeof(cmd),
1521  "Transport: %s\r\n",
1522  transport);
1523  if (rt->accept_dynamic_rate)
1524  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1525  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1526  char real_res[41], real_csum[9];
1527  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1528  real_challenge);
1529  av_strlcatf(cmd, sizeof(cmd),
1530  "If-Match: %s\r\n"
1531  "RealChallenge2: %s, sd=%s\r\n",
1532  rt->session_id, real_res, real_csum);
1533  }
1534  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1535  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1536  err = 1;
1537  goto fail;
1538  } else if (reply->status_code != RTSP_STATUS_OK ||
1539  reply->nb_transports != 1) {
1541  goto fail;
1542  }
1543 
1544  /* XXX: same protocol for all streams is required */
1545  if (i > 0) {
1546  if (reply->transports[0].lower_transport != rt->lower_transport ||
1547  reply->transports[0].transport != rt->transport) {
1548  err = AVERROR_INVALIDDATA;
1549  goto fail;
1550  }
1551  } else {
1552  rt->lower_transport = reply->transports[0].lower_transport;
1553  rt->transport = reply->transports[0].transport;
1554  }
1555 
1556  /* Fail if the server responded with another lower transport mode
1557  * than what we requested. */
1558  if (reply->transports[0].lower_transport != lower_transport) {
1559  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1560  err = AVERROR_INVALIDDATA;
1561  goto fail;
1562  }
1563 
1564  switch(reply->transports[0].lower_transport) {
1566  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1567  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1568  break;
1569 
1570  case RTSP_LOWER_TRANSPORT_UDP: {
1571  char url[1024], options[30] = "";
1572  const char *peer = host;
1573 
1574  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1575  av_strlcpy(options, "?connect=1", sizeof(options));
1576  /* Use source address if specified */
1577  if (reply->transports[0].source[0])
1578  peer = reply->transports[0].source;
1579  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1580  reply->transports[0].server_port_min, "%s", options);
1581  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1582  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1583  err = AVERROR_INVALIDDATA;
1584  goto fail;
1585  }
1586  break;
1587  }
1589  char url[1024], namebuf[50], optbuf[20] = "";
1590  struct sockaddr_storage addr;
1591  int port, ttl;
1592 
1593  if (reply->transports[0].destination.ss_family) {
1594  addr = reply->transports[0].destination;
1595  port = reply->transports[0].port_min;
1596  ttl = reply->transports[0].ttl;
1597  } else {
1598  addr = rtsp_st->sdp_ip;
1599  port = rtsp_st->sdp_port;
1600  ttl = rtsp_st->sdp_ttl;
1601  }
1602  if (ttl > 0)
1603  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1604  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1605  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1606  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1607  port, "%s", optbuf);
1608  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1609  &s->interrupt_callback, NULL) < 0) {
1610  err = AVERROR_INVALIDDATA;
1611  goto fail;
1612  }
1613  break;
1614  }
1615  }
1616 
1617  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1618  goto fail;
1619  }
1620 
1621  if (rt->nb_rtsp_streams && reply->timeout > 0)
1622  rt->timeout = reply->timeout;
1623 
1624  if (rt->server_type == RTSP_SERVER_REAL)
1625  rt->need_subscription = 1;
1626 
1627  return 0;
1628 
1629 fail:
1630  ff_rtsp_undo_setup(s, 0);
1631  return err;
1632 }
1633 
1635 {
1636  RTSPState *rt = s->priv_data;
1637  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1638  ffurl_close(rt->rtsp_hd);
1639  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1640 }
1641 
1643 {
1644  RTSPState *rt = s->priv_data;
1645  char proto[128], host[1024], path[1024];
1646  char tcpname[1024], cmd[2048], auth[128];
1647  const char *lower_rtsp_proto = "tcp";
1648  int port, err, tcp_fd;
1649  RTSPMessageHeader reply1, *reply = &reply1;
1650  int lower_transport_mask = 0;
1651  int default_port = RTSP_DEFAULT_PORT;
1652  char real_challenge[64] = "";
1653  struct sockaddr_storage peer;
1654  socklen_t peer_len = sizeof(peer);
1655 
1656  if (rt->rtp_port_max < rt->rtp_port_min) {
1657  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1658  "than min port %d\n", rt->rtp_port_max,
1659  rt->rtp_port_min);
1660  return AVERROR(EINVAL);
1661  }
1662 
1663  if (!ff_network_init())
1664  return AVERROR(EIO);
1665 
1666  if (s->max_delay < 0) /* Not set by the caller */
1668 
1673  }
1674  /* Only pass through valid flags from here */
1676 
1677 redirect:
1678  memset(&reply1, 0, sizeof(reply1));
1679  /* extract hostname and port */
1680  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1681  host, sizeof(host), &port, path, sizeof(path), s->filename);
1682 
1683  if (!strcmp(proto, "rtsps")) {
1684  lower_rtsp_proto = "tls";
1685  default_port = RTSPS_DEFAULT_PORT;
1687  }
1688 
1689  if (*auth) {
1690  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1691  }
1692  if (port < 0)
1693  port = default_port;
1694 
1695  lower_transport_mask = rt->lower_transport_mask;
1696 
1697  if (!lower_transport_mask)
1698  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1699 
1700  if (s->oformat) {
1701  /* Only UDP or TCP - UDP multicast isn't supported. */
1702  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1703  (1 << RTSP_LOWER_TRANSPORT_TCP);
1704  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1705  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1706  "only UDP and TCP are supported for output.\n");
1707  err = AVERROR(EINVAL);
1708  goto fail;
1709  }
1710  }
1711 
1712  /* Construct the URI used in request; this is similar to s->filename,
1713  * but with authentication credentials removed and RTSP specific options
1714  * stripped out. */
1715  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1716  host, port, "%s", path);
1717 
1718  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1719  /* set up initial handshake for tunneling */
1720  char httpname[1024];
1721  char sessioncookie[17];
1722  char headers[1024];
1723 
1724  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1725  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1727 
1728  /* GET requests */
1729  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1730  &s->interrupt_callback) < 0) {
1731  err = AVERROR(EIO);
1732  goto fail;
1733  }
1734 
1735  /* generate GET headers */
1736  snprintf(headers, sizeof(headers),
1737  "x-sessioncookie: %s\r\n"
1738  "Accept: application/x-rtsp-tunnelled\r\n"
1739  "Pragma: no-cache\r\n"
1740  "Cache-Control: no-cache\r\n",
1741  sessioncookie);
1742  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1743 
1744  /* complete the connection */
1745  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1746  err = AVERROR(EIO);
1747  goto fail;
1748  }
1749 
1750  /* POST requests */
1751  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1752  &s->interrupt_callback) < 0 ) {
1753  err = AVERROR(EIO);
1754  goto fail;
1755  }
1756 
1757  /* generate POST headers */
1758  snprintf(headers, sizeof(headers),
1759  "x-sessioncookie: %s\r\n"
1760  "Content-Type: application/x-rtsp-tunnelled\r\n"
1761  "Pragma: no-cache\r\n"
1762  "Cache-Control: no-cache\r\n"
1763  "Content-Length: 32767\r\n"
1764  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1765  sessioncookie);
1766  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1767  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1768 
1769  /* Initialize the authentication state for the POST session. The HTTP
1770  * protocol implementation doesn't properly handle multi-pass
1771  * authentication for POST requests, since it would require one of
1772  * the following:
1773  * - implementing Expect: 100-continue, which many HTTP servers
1774  * don't support anyway, even less the RTSP servers that do HTTP
1775  * tunneling
1776  * - sending the whole POST data until getting a 401 reply specifying
1777  * what authentication method to use, then resending all that data
1778  * - waiting for potential 401 replies directly after sending the
1779  * POST header (waiting for some unspecified time)
1780  * Therefore, we copy the full auth state, which works for both basic
1781  * and digest. (For digest, we would have to synchronize the nonce
1782  * count variable between the two sessions, if we'd do more requests
1783  * with the original session, though.)
1784  */
1786 
1787  /* complete the connection */
1788  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1789  err = AVERROR(EIO);
1790  goto fail;
1791  }
1792  } else {
1793  int ret;
1794  /* open the tcp connection */
1795  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1796  host, port,
1797  "?timeout=%d", rt->stimeout);
1798  if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1799  &s->interrupt_callback, NULL)) < 0) {
1800  err = ret;
1801  goto fail;
1802  }
1803  rt->rtsp_hd_out = rt->rtsp_hd;
1804  }
1805  rt->seq = 0;
1806 
1807  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1808  if (tcp_fd < 0) {
1809  err = tcp_fd;
1810  goto fail;
1811  }
1812  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1813  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1814  NULL, 0, NI_NUMERICHOST);
1815  }
1816 
1817  /* request options supported by the server; this also detects server
1818  * type */
1819  for (rt->server_type = RTSP_SERVER_RTP;;) {
1820  cmd[0] = 0;
1821  if (rt->server_type == RTSP_SERVER_REAL)
1822  av_strlcat(cmd,
1823  /*
1824  * The following entries are required for proper
1825  * streaming from a Realmedia server. They are
1826  * interdependent in some way although we currently
1827  * don't quite understand how. Values were copied
1828  * from mplayer SVN r23589.
1829  * ClientChallenge is a 16-byte ID in hex
1830  * CompanyID is a 16-byte ID in base64
1831  */
1832  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1833  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1834  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1835  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1836  sizeof(cmd));
1837  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1838  if (reply->status_code != RTSP_STATUS_OK) {
1840  goto fail;
1841  }
1842 
1843  /* detect server type if not standard-compliant RTP */
1844  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1846  continue;
1847  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1849  } else if (rt->server_type == RTSP_SERVER_REAL)
1850  strcpy(real_challenge, reply->real_challenge);
1851  break;
1852  }
1853 
1854  if (CONFIG_RTSP_DEMUXER && s->iformat)
1855  err = ff_rtsp_setup_input_streams(s, reply);
1856  else if (CONFIG_RTSP_MUXER)
1857  err = ff_rtsp_setup_output_streams(s, host);
1858  else
1859  av_assert0(0);
1860  if (err)
1861  goto fail;
1862 
1863  do {
1864  int lower_transport = ff_log2_tab[lower_transport_mask &
1865  ~(lower_transport_mask - 1)];
1866 
1867  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1868  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1869  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1870 
1871  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1872  rt->server_type == RTSP_SERVER_REAL ?
1873  real_challenge : NULL);
1874  if (err < 0)
1875  goto fail;
1876  lower_transport_mask &= ~(1 << lower_transport);
1877  if (lower_transport_mask == 0 && err == 1) {
1878  err = AVERROR(EPROTONOSUPPORT);
1879  goto fail;
1880  }
1881  } while (err);
1882 
1883  rt->lower_transport_mask = lower_transport_mask;
1884  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1885  rt->state = RTSP_STATE_IDLE;
1886  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1887  return 0;
1888  fail:
1891  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1892  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1893  rt->session_id[0] = '\0';
1894  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1895  reply->status_code,
1896  s->filename);
1897  goto redirect;
1898  }
1899  ff_network_close();
1900  return err;
1901 }
1902 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1903 
1904 #if CONFIG_RTPDEC
1905 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1906  uint8_t *buf, int buf_size, int64_t wait_end)
1907 {
1908  RTSPState *rt = s->priv_data;
1909  RTSPStream *rtsp_st;
1910  int n, i, ret, tcp_fd, timeout_cnt = 0;
1911  int max_p = 0;
1912  struct pollfd *p = rt->p;
1913  int *fds = NULL, fdsnum, fdsidx;
1914 
1915  for (;;) {
1917  return AVERROR_EXIT;
1918  if (wait_end && wait_end - av_gettime_relative() < 0)
1919  return AVERROR(EAGAIN);
1920  max_p = 0;
1921  if (rt->rtsp_hd) {
1922  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1923  p[max_p].fd = tcp_fd;
1924  p[max_p++].events = POLLIN;
1925  } else {
1926  tcp_fd = -1;
1927  }
1928  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1929  rtsp_st = rt->rtsp_streams[i];
1930  if (rtsp_st->rtp_handle) {
1931  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1932  &fds, &fdsnum)) {
1933  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1934  return ret;
1935  }
1936  if (fdsnum != 2) {
1937  av_log(s, AV_LOG_ERROR,
1938  "Number of fds %d not supported\n", fdsnum);
1939  return AVERROR_INVALIDDATA;
1940  }
1941  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1942  p[max_p].fd = fds[fdsidx];
1943  p[max_p++].events = POLLIN;
1944  }
1945  av_freep(&fds);
1946  }
1947  }
1948  n = poll(p, max_p, POLL_TIMEOUT_MS);
1949  if (n > 0) {
1950  int j = 1 - (tcp_fd == -1);
1951  timeout_cnt = 0;
1952  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1953  rtsp_st = rt->rtsp_streams[i];
1954  if (rtsp_st->rtp_handle) {
1955  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1956  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1957  if (ret > 0) {
1958  *prtsp_st = rtsp_st;
1959  return ret;
1960  }
1961  }
1962  j+=2;
1963  }
1964  }
1965 #if CONFIG_RTSP_DEMUXER
1966  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1967  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1968  if (rt->state == RTSP_STATE_STREAMING) {
1970  return AVERROR_EOF;
1971  else
1973  "Unable to answer to TEARDOWN\n");
1974  } else
1975  return 0;
1976  } else {
1977  RTSPMessageHeader reply;
1978  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1979  if (ret < 0)
1980  return ret;
1981  /* XXX: parse message */
1982  if (rt->state != RTSP_STATE_STREAMING)
1983  return 0;
1984  }
1985  }
1986 #endif
1987  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1988  return AVERROR(ETIMEDOUT);
1989  } else if (n < 0 && errno != EINTR)
1990  return AVERROR(errno);
1991  }
1992 }
1993 
1994 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1995  const uint8_t *buf, int len)
1996 {
1997  RTSPState *rt = s->priv_data;
1998  int i;
1999  if (len < 0)
2000  return len;
2001  if (rt->nb_rtsp_streams == 1) {
2002  *rtsp_st = rt->rtsp_streams[0];
2003  return len;
2004  }
2005  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2006  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2007  int no_ssrc = 0;
2008  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2009  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2010  if (!rtpctx)
2011  continue;
2012  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2013  *rtsp_st = rt->rtsp_streams[i];
2014  return len;
2015  }
2016  if (!rtpctx->ssrc)
2017  no_ssrc = 1;
2018  }
2019  if (no_ssrc) {
2021  "Unable to pick stream for packet - SSRC not known for "
2022  "all streams\n");
2023  return AVERROR(EAGAIN);
2024  }
2025  } else {
2026  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2027  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2028  *rtsp_st = rt->rtsp_streams[i];
2029  return len;
2030  }
2031  }
2032  }
2033  }
2034  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2035  return AVERROR(EAGAIN);
2036 }
2037 
2039 {
2040  RTSPState *rt = s->priv_data;
2041  int ret, len;
2042  RTSPStream *rtsp_st, *first_queue_st = NULL;
2043  int64_t wait_end = 0;
2044 
2045  if (rt->nb_byes == rt->nb_rtsp_streams)
2046  return AVERROR_EOF;
2047 
2048  /* get next frames from the same RTP packet */
2049  if (rt->cur_transport_priv) {
2050  if (rt->transport == RTSP_TRANSPORT_RDT) {
2051  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2052  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2053  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2054  } else if (CONFIG_RTPDEC && rt->ts) {
2055  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2056  if (ret >= 0) {
2057  rt->recvbuf_pos += ret;
2058  ret = rt->recvbuf_pos < rt->recvbuf_len;
2059  }
2060  } else
2061  ret = -1;
2062  if (ret == 0) {
2063  rt->cur_transport_priv = NULL;
2064  return 0;
2065  } else if (ret == 1) {
2066  return 0;
2067  } else
2068  rt->cur_transport_priv = NULL;
2069  }
2070 
2071 redo:
2072  if (rt->transport == RTSP_TRANSPORT_RTP) {
2073  int i;
2074  int64_t first_queue_time = 0;
2075  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2076  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2077  int64_t queue_time;
2078  if (!rtpctx)
2079  continue;
2080  queue_time = ff_rtp_queued_packet_time(rtpctx);
2081  if (queue_time && (queue_time - first_queue_time < 0 ||
2082  !first_queue_time)) {
2083  first_queue_time = queue_time;
2084  first_queue_st = rt->rtsp_streams[i];
2085  }
2086  }
2087  if (first_queue_time) {
2088  wait_end = first_queue_time + s->max_delay;
2089  } else {
2090  wait_end = 0;
2091  first_queue_st = NULL;
2092  }
2093  }
2094 
2095  /* read next RTP packet */
2096  if (!rt->recvbuf) {
2098  if (!rt->recvbuf)
2099  return AVERROR(ENOMEM);
2100  }
2101 
2102  switch(rt->lower_transport) {
2103  default:
2104 #if CONFIG_RTSP_DEMUXER
2106  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2107  break;
2108 #endif
2111  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2112  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2113  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2114  break;
2116  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2117  wait_end && wait_end < av_gettime_relative())
2118  len = AVERROR(EAGAIN);
2119  else
2120  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2121  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2122  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2124  break;
2125  }
2126  if (len == AVERROR(EAGAIN) && first_queue_st &&
2127  rt->transport == RTSP_TRANSPORT_RTP) {
2128  rtsp_st = first_queue_st;
2129  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2130  goto end;
2131  }
2132  if (len < 0)
2133  return len;
2134  if (len == 0)
2135  return AVERROR_EOF;
2136  if (rt->transport == RTSP_TRANSPORT_RDT) {
2137  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2138  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2139  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2140  if (rtsp_st->feedback) {
2141  AVIOContext *pb = NULL;
2143  pb = s->pb;
2144  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2145  }
2146  if (ret < 0) {
2147  /* Either bad packet, or a RTCP packet. Check if the
2148  * first_rtcp_ntp_time field was initialized. */
2149  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2150  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2151  /* first_rtcp_ntp_time has been initialized for this stream,
2152  * copy the same value to all other uninitialized streams,
2153  * in order to map their timestamp origin to the same ntp time
2154  * as this one. */
2155  int i;
2156  AVStream *st = NULL;
2157  if (rtsp_st->stream_index >= 0)
2158  st = s->streams[rtsp_st->stream_index];
2159  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2160  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2161  AVStream *st2 = NULL;
2162  if (rt->rtsp_streams[i]->stream_index >= 0)
2163  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2164  if (rtpctx2 && st && st2 &&
2165  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2166  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2167  rtpctx2->rtcp_ts_offset = av_rescale_q(
2168  rtpctx->rtcp_ts_offset, st->time_base,
2169  st2->time_base);
2170  }
2171  }
2172  // Make real NTP start time available in AVFormatContext
2173  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2174  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2175  if (rtpctx->st) {
2176  s->start_time_realtime -=
2177  av_rescale (rtpctx->rtcp_ts_offset,
2178  (uint64_t) rtpctx->st->time_base.num * 1000000,
2179  rtpctx->st->time_base.den);
2180  }
2181  }
2182  }
2183  if (ret == -RTCP_BYE) {
2184  rt->nb_byes++;
2185 
2186  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2187  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2188 
2189  if (rt->nb_byes == rt->nb_rtsp_streams)
2190  return AVERROR_EOF;
2191  }
2192  }
2193  } else if (CONFIG_RTPDEC && rt->ts) {
2194  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2195  if (ret >= 0) {
2196  if (ret < len) {
2197  rt->recvbuf_len = len;
2198  rt->recvbuf_pos = ret;
2199  rt->cur_transport_priv = rt->ts;
2200  return 1;
2201  } else {
2202  ret = 0;
2203  }
2204  }
2205  } else {
2206  return AVERROR_INVALIDDATA;
2207  }
2208 end:
2209  if (ret < 0)
2210  goto redo;
2211  if (ret == 1)
2212  /* more packets may follow, so we save the RTP context */
2213  rt->cur_transport_priv = rtsp_st->transport_priv;
2214 
2215  return ret;
2216 }
2217 #endif /* CONFIG_RTPDEC */
2218 
2219 #if CONFIG_SDP_DEMUXER
2220 static int sdp_probe(AVProbeData *p1)
2221 {
2222  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2223 
2224  /* we look for a line beginning "c=IN IP" */
2225  while (p < p_end && *p != '\0') {
2226  if (sizeof("c=IN IP") - 1 < p_end - p &&
2227  av_strstart(p, "c=IN IP", NULL))
2228  return AVPROBE_SCORE_EXTENSION;
2229 
2230  while (p < p_end - 1 && *p != '\n') p++;
2231  if (++p >= p_end)
2232  break;
2233  if (*p == '\r')
2234  p++;
2235  }
2236  return 0;
2237 }
2238 
2239 static void append_source_addrs(char *buf, int size, const char *name,
2240  int count, struct RTSPSource **addrs)
2241 {
2242  int i;
2243  if (!count)
2244  return;
2245  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2246  for (i = 1; i < count; i++)
2247  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2248 }
2249 
2250 static int sdp_read_header(AVFormatContext *s)
2251 {
2252  RTSPState *rt = s->priv_data;
2253  RTSPStream *rtsp_st;
2254  int size, i, err;
2255  char *content;
2256  char url[1024];
2257 
2258  if (!ff_network_init())
2259  return AVERROR(EIO);
2260 
2261  if (s->max_delay < 0) /* Not set by the caller */
2263  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2265 
2266  /* read the whole sdp file */
2267  /* XXX: better loading */
2268  content = av_malloc(SDP_MAX_SIZE);
2269  if (!content)
2270  return AVERROR(ENOMEM);
2271  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2272  if (size <= 0) {
2273  av_free(content);
2274  return AVERROR_INVALIDDATA;
2275  }
2276  content[size] ='\0';
2277 
2278  err = ff_sdp_parse(s, content);
2279  av_freep(&content);
2280  if (err) goto fail;
2281 
2282  /* open each RTP stream */
2283  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2284  char namebuf[50];
2285  rtsp_st = rt->rtsp_streams[i];
2286 
2287  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2288  AVDictionary *opts = map_to_opts(rt);
2289 
2290  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2291  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2292  ff_url_join(url, sizeof(url), "rtp", NULL,
2293  namebuf, rtsp_st->sdp_port,
2294  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2295  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2296  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2297  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2298 
2299  append_source_addrs(url, sizeof(url), "sources",
2300  rtsp_st->nb_include_source_addrs,
2301  rtsp_st->include_source_addrs);
2302  append_source_addrs(url, sizeof(url), "block",
2303  rtsp_st->nb_exclude_source_addrs,
2304  rtsp_st->exclude_source_addrs);
2305  err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2306  &s->interrupt_callback, &opts);
2307 
2308  av_dict_free(&opts);
2309 
2310  if (err < 0) {
2311  err = AVERROR_INVALIDDATA;
2312  goto fail;
2313  }
2314  }
2315  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2316  goto fail;
2317  }
2318  return 0;
2319 fail:
2321  ff_network_close();
2322  return err;
2323 }
2324 
2325 static int sdp_read_close(AVFormatContext *s)
2326 {
2328  ff_network_close();
2329  return 0;
2330 }
2331 
2332 static const AVClass sdp_demuxer_class = {
2333  .class_name = "SDP demuxer",
2334  .item_name = av_default_item_name,
2335  .option = sdp_options,
2336  .version = LIBAVUTIL_VERSION_INT,
2337 };
2338 
2339 AVInputFormat ff_sdp_demuxer = {
2340  .name = "sdp",
2341  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2342  .priv_data_size = sizeof(RTSPState),
2343  .read_probe = sdp_probe,
2344  .read_header = sdp_read_header,
2346  .read_close = sdp_read_close,
2347  .priv_class = &sdp_demuxer_class,
2348 };
2349 #endif /* CONFIG_SDP_DEMUXER */
2350 
2351 #if CONFIG_RTP_DEMUXER
2352 static int rtp_probe(AVProbeData *p)
2353 {
2354  if (av_strstart(p->filename, "rtp:", NULL))
2355  return AVPROBE_SCORE_MAX;
2356  return 0;
2357 }
2358 
2359 static int rtp_read_header(AVFormatContext *s)
2360 {
2361  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2362  char host[500], sdp[500];
2363  int ret, port;
2364  URLContext* in = NULL;
2365  int payload_type;
2366  AVCodecContext codec = { 0 };
2367  struct sockaddr_storage addr;
2368  AVIOContext pb;
2369  socklen_t addrlen = sizeof(addr);
2370  RTSPState *rt = s->priv_data;
2371 
2372  if (!ff_network_init())
2373  return AVERROR(EIO);
2374 
2375  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2376  &s->interrupt_callback, NULL);
2377  if (ret)
2378  goto fail;
2379 
2380  while (1) {
2381  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2382  if (ret == AVERROR(EAGAIN))
2383  continue;
2384  if (ret < 0)
2385  goto fail;
2386  if (ret < 12) {
2387  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2388  continue;
2389  }
2390 
2391  if ((recvbuf[0] & 0xc0) != 0x80) {
2392  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2393  "received\n");
2394  continue;
2395  }
2396 
2397  if (RTP_PT_IS_RTCP(recvbuf[1]))
2398  continue;
2399 
2400  payload_type = recvbuf[1] & 0x7f;
2401  break;
2402  }
2403  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2404  ffurl_close(in);
2405  in = NULL;
2406 
2407  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2408  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2409  "without an SDP file describing it\n",
2410  payload_type);
2411  goto fail;
2412  }
2413  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2414  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2415  "properly you need an SDP file "
2416  "describing it\n");
2417  }
2418 
2419  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2420  NULL, 0, s->filename);
2421 
2422  snprintf(sdp, sizeof(sdp),
2423  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2424  addr.ss_family == AF_INET ? 4 : 6, host,
2425  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2426  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2427  port, payload_type);
2428  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2429 
2430  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2431  s->pb = &pb;
2432 
2433  /* sdp_read_header initializes this again */
2434  ff_network_close();
2435 
2436  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2437 
2438  ret = sdp_read_header(s);
2439  s->pb = NULL;
2440  return ret;
2441 
2442 fail:
2443  if (in)
2444  ffurl_close(in);
2445  ff_network_close();
2446  return ret;
2447 }
2448 
2449 static const AVClass rtp_demuxer_class = {
2450  .class_name = "RTP demuxer",
2451  .item_name = av_default_item_name,
2452  .option = rtp_options,
2453  .version = LIBAVUTIL_VERSION_INT,
2454 };
2455 
2456 AVInputFormat ff_rtp_demuxer = {
2457  .name = "rtp",
2458  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2459  .priv_data_size = sizeof(RTSPState),
2460  .read_probe = rtp_probe,
2461  .read_header = rtp_read_header,
2463  .read_close = sdp_read_close,
2464  .flags = AVFMT_NOFILE,
2465  .priv_class = &rtp_demuxer_class,
2466 };
2467 #endif /* CONFIG_RTP_DEMUXER */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:273
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:93
void av_url_split(char *proto, int proto_size, char *authorization, int authorization_size, char *hostname, int hostname_size, int *port_ptr, char *path, int path_size, const char *url)
Split a URL string into components.
Definition: utils.c:3978
char crypto_suite[40]
Definition: rtsp.h:472
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:387
#define NULL
Definition: coverity.c:32
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:100
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:548
float v
const char * s
Definition: avisynth_c.h:631
Bytestream IO Context.
Definition: avio.h:111
Realmedia Data Transport.
Definition: rtsp.h:58
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:567
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
Definition: avformat.h:1510
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1535
AVOption.
Definition: opt.h:255
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:115
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:187
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:62
const char * filename
Definition: avformat.h:461
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
Definition: rtsp.c:165
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:317
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4098
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:558
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:374
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
#define CONFIG_RTPDEC
Definition: config.h:594
Windows Media server.
Definition: rtsp.h:209
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:354
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:788
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2787
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:202
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:131
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
Definition: rdt.c:335
int num
numerator
Definition: rational.h:44
int index
stream index in AVFormatContext
Definition: avformat.h:855
#define AVIO_FLAG_READ
read-only
Definition: avio.h:485
char * user_agent
User-Agent string.
Definition: rtsp.h:407
char location[4096]
the "Location:" field.
Definition: rtsp.h:152
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:486
int mode_record
transport set to record data
Definition: rtsp.h:112
enum AVMediaType codec_type
Definition: rtp.c:37
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:223
void ff_network_close(void)
Definition: network.c:102
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:144
initialized and sending/receiving data
Definition: rtsp.h:197
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:270
static av_always_inline void interleave(IDWTELEM *dst, IDWTELEM *src0, IDWTELEM *src1, int w2, int add, int shift)
Definition: dirac_dwt.c:40
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:420
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:79
#define freeaddrinfo
Definition: network.h:208
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:452
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1334
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:418
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:245
int auth_type
The currently chosen auth type.
Definition: httpauth.h:59
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:239
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:132
AVCodec.
Definition: avcodec.h:3482
#define AI_NUMERICHOST
Definition: network.h:177
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:121
This describes the server response to each RTSP command.
Definition: rtsp.h:127
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
Definition: rtpdec.c:509
#define RECVBUF_SIZE
Definition: rtsp.c:59
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:142
Format I/O context.
Definition: avformat.h:1285
#define RTP_PT_PRIVATE
Definition: rtp.h:77
#define COMMON_OPTS()
Definition: rtsp.c:77
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
Definition: rtp.c:143
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:207
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:402
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
#define RTSP_FLAG_PREFER_TCP
Try RTP via TCP first if possible.
Definition: rtsp.h:423
int recvbuf_len
Definition: rtsp.h:323
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:180
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Public dictionary API.
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:45
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:359
#define CONFIG_RTSP_DEMUXER
Definition: config.h:1213
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:148
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:74
#define av_malloc(s)
Opaque data information usually continuous.
Definition: avutil.h:195
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:109
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
Definition: rtpdec.h:126
int ff_network_init(void)
Definition: network.c:55
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1244
AVOptions.
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:470
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:106
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int id
Format-specific stream ID.
Definition: avformat.h:861
enum AVStreamParseType need_parsing
Definition: avformat.h:1046
#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)
static void handler(vbi_event *ev, void *user_data)
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:3761
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1353
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:372
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:328
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:435
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
#define CONFIG_RTSP_MUXER
Definition: config.h:1870
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:131
#define AVERROR_EOF
End of file.
Definition: error.h:55
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:156
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
ptrdiff_t size
Definition: opengl_enc.c:101
static const uint8_t header[24]
Definition: sdr2.c:67
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:464
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
Normal RTSP.
Definition: rtsp.h:68
const OptionDef options[]
Definition: ffserver.c:3810
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
#define av_log(a,...)
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:134
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:542
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1304
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:177
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:77
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:147
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:455
Private data for the RTSP demuxer.
Definition: rtsp.h:218
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:255
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
Definition: avio.c:277
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVDictionary * metadata
Metadata that applies to the whole file.
Definition: avformat.h:1497
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:585
int profile
Definition: mxfenc.c:1820
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
av_default_item_name
#define AV_RB16
Definition: intreadwrite.h:53
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:178
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const AVOption ff_rtsp_options[]
Definition: rtsp.c:82
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:199
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:182
Definition: graph2dot.c:48
URLContext * rtsp_hd
Definition: rtsp.h:220
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
Definition: avcodec.h:3489
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:331
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:453
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:610
GLsizei count
Definition: opengl_enc.c:109
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
Definition: base64.c:138
int64_t rtcp_ts_offset
Definition: rtpdec.h:182
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:83
#define fail()
Definition: checkasm.h:57
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:131
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:225
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:164
int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:71
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:440
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:873
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:463
int seq
RTSP command sequence number.
Definition: rtsp.h:241
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:462
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:339
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1341
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:419
#define NI_NUMERICHOST
Definition: network.h:185
#define th
Definition: regdef.h:75
#define LIBAVFORMAT_IDENT
Definition: version.h:44
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:307
int recvbuf_pos
Definition: rtsp.h:322
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:119
char filename[1024]
input or output filename
Definition: avformat.h:1361
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:223
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:134
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes to a null-terminated string.
Definition: base64.h:61
#define FFMIN(a, b)
Definition: common.h:92
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:283
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:55
int content_length
length of the data following this header
Definition: rtsp.h:129
int max_streams
The maximum number of streams.
Definition: avformat.h:1840
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: [;ti...
Definition: rtsp.h:172
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:75
enum AVStreamParseType need_parsing
Definition: rtpdec.h:119
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:88
RTSP over HTTP (tunneling)
Definition: rtsp.h:69
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:130
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:140
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:156
int n
Definition: avisynth_c.h:547
AVDictionary * metadata
Definition: avformat.h:928
char crypto_params[100]
Definition: rtsp.h:473
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:192
int(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
Definition: rtpdec.h:128
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:578
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define ENC
Definition: rtsp.c:64
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:450
Raw data (over UDP)
Definition: rtsp.h:59
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:321
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:23
int sdp_payload_type
payload type
Definition: rtsp.h:457
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:541
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:454
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1178
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:187
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:634
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:441
Stream structure.
Definition: avformat.h:854
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
Definition: url.c:36
int nb_byes
Definition: rtsp.h:336
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:262
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:426
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:451
enum AVMediaType codec_type
Definition: avcodec.h:1520
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:718
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
Definition: avio.c:607
enum AVCodecID codec_id
Definition: avcodec.h:1529
int rtp_port_max
Definition: rtsp.h:387
#define NTP_OFFSET
Definition: internal.h:159
Definition: rtp.h:100
int sample_rate
samples per second
Definition: avcodec.h:2272
AVIOContext * pb
I/O context.
Definition: avformat.h:1327
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:382
int server_port_max
Definition: rtsp.h:105
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
main external API structure.
Definition: avcodec.h:1512
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:3038
#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
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:413
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:258
void * buf
Definition: avisynth_c.h:553
Definition: url.h:39
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:73
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:487
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:69
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:377
int client_port_max
Definition: rtsp.h:101
Describe the class of an AVClass context structure.
Definition: log.h:67
#define SDP_MAX_SIZE
Definition: rtsp.c:58
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:515
#define SPACE_CHARS
Definition: internal.h:225
void * priv_data
Definition: url.h:42
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:466
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:279
not initialized
Definition: rtsp.h:196
int64_t range_end
Definition: rtsp.h:138
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:118
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:2806
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:155
AVMediaType
Definition: avutil.h:191
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:70
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:720
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:750
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:752
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count)
some rtp servers assume client is dead if they don't hear from them...
Definition: rtpdec.c:270
#define s1
Definition: regdef.h:38
#define snprintf
Definition: snprintf.h:34
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:470
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:3698
int buffer_size
Definition: rtsp.h:410
This structure contains the data a format has to probe a file.
Definition: avformat.h:460
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:76
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
Definition: avstring.c:93
int interleaved_max
Definition: rtsp.h:93
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:267
static int flags
Definition: cpu.c:47
int ffurl_close(URLContext *h)
Definition: avio.c:419
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:138
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1370
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:231
#define DEC
Definition: rtsp.c:63
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:472
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:34
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:207
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:103
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
uint32_t ssrc
Definition: rtpdec.h:153
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:119
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:477
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:118
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:72
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:299
static double c[64]
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:288
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:463
int ffurl_read_complete(URLContext *h, unsigned char *buf, int size)
Read as many bytes as possible (up to size), calling the read function multiple times if necessary...
Definition: avio.c:367
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
Definition: rdt.c:94
int den
denominator
Definition: rational.h:45
char default_lang[4]
Definition: rtsp.h:409
struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:1297
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:3733
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
uint32_t base_timestamp
Definition: rtpdec.h:156
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:397
#define getnameinfo
Definition: network.h:209
#define av_free(p)
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:149
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:276
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:78
int channels
number of audio channels
Definition: avcodec.h:2273
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:446
void * priv_data
Format private data.
Definition: avformat.h:1313
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:593
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:843
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:456
#define MAX_TIMEOUTS
Definition: rtsp.c:57
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:986
int ai_flags
Definition: network.h:128
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1380
Realmedia-style server.
Definition: rtsp.h:208
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:344
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:553
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:640
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:116
#define av_malloc_array(a, b)
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:896
int interleaved_max
Definition: rtsp.h:444
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
Definition: rtpdec.c:830
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:114
int ff_rtp_set_remote_url(URLContext *h, const char *uri)
If no filename is given to av_open_input_file because you want to get the local port first...
Definition: rtpproto.c:98
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:2831
AVStream * st
Definition: rtpdec.h:151
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:444
This structure stores compressed data.
Definition: avcodec.h:1410
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
Definition: aviobuf.c:963
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:105
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:369
static const AVOption rtp_options[]
Definition: rtsp.c:112
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
Definition: avio.c:360
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:436
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:240
#define OFFSET(x)
Definition: rtsp.c:62
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:97
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:437
No authentication specified.
Definition: httpauth.h:29
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:101
const char * name
Definition: opengl_enc.c:103