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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 || !(rt->media_type_mask & (1 << codec_type))) {
443  s1->skip_media = 1;
444  return;
445  }
446  rtsp_st = av_mallocz(sizeof(RTSPStream));
447  if (!rtsp_st)
448  return;
449  rtsp_st->stream_index = -1;
450  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
451 
452  rtsp_st->sdp_ip = s1->default_ip;
453  rtsp_st->sdp_ttl = s1->default_ttl;
454 
455  copy_default_source_addrs(s1->default_include_source_addrs,
456  s1->nb_default_include_source_addrs,
457  &rtsp_st->include_source_addrs,
458  &rtsp_st->nb_include_source_addrs);
459  copy_default_source_addrs(s1->default_exclude_source_addrs,
460  s1->nb_default_exclude_source_addrs,
461  &rtsp_st->exclude_source_addrs,
462  &rtsp_st->nb_exclude_source_addrs);
463 
464  get_word(buf1, sizeof(buf1), &p); /* port */
465  rtsp_st->sdp_port = atoi(buf1);
466 
467  get_word(buf1, sizeof(buf1), &p); /* protocol */
468  if (!strcmp(buf1, "udp"))
470  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
471  rtsp_st->feedback = 1;
472 
473  /* XXX: handle list of formats */
474  get_word(buf1, sizeof(buf1), &p); /* format list */
475  rtsp_st->sdp_payload_type = atoi(buf1);
476 
477  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
478  /* no corresponding stream */
479  if (rt->transport == RTSP_TRANSPORT_RAW) {
480  if (CONFIG_RTPDEC && !rt->ts)
481  rt->ts = avpriv_mpegts_parse_open(s);
482  } else {
484  handler = ff_rtp_handler_find_by_id(
486  init_rtp_handler(handler, rtsp_st, NULL);
487  finalize_rtp_handler_init(s, rtsp_st, NULL);
488  }
489  } else if (rt->server_type == RTSP_SERVER_WMS &&
490  codec_type == AVMEDIA_TYPE_DATA) {
491  /* RTX stream, a stream that carries all the other actual
492  * audio/video streams. Don't expose this to the callers. */
493  } else {
494  st = avformat_new_stream(s, NULL);
495  if (!st)
496  return;
497  st->id = rt->nb_rtsp_streams - 1;
498  rtsp_st->stream_index = st->index;
499  st->codec->codec_type = codec_type;
500  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
502  /* if standard payload type, we can find the codec right now */
504  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
505  st->codec->sample_rate > 0)
506  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
507  /* Even static payload types may need a custom depacketizer */
508  handler = ff_rtp_handler_find_by_id(
509  rtsp_st->sdp_payload_type, st->codec->codec_type);
510  init_rtp_handler(handler, rtsp_st, st);
511  finalize_rtp_handler_init(s, rtsp_st, st);
512  }
513  if (rt->default_lang[0])
514  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
515  }
516  /* put a default control url */
517  av_strlcpy(rtsp_st->control_url, rt->control_uri,
518  sizeof(rtsp_st->control_url));
519  break;
520  case 'a':
521  if (av_strstart(p, "control:", &p)) {
522  if (s->nb_streams == 0) {
523  if (!strncmp(p, "rtsp://", 7))
524  av_strlcpy(rt->control_uri, p,
525  sizeof(rt->control_uri));
526  } else {
527  char proto[32];
528  /* get the control url */
529  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
530 
531  /* XXX: may need to add full url resolution */
532  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
533  NULL, NULL, 0, p);
534  if (proto[0] == '\0') {
535  /* relative control URL */
536  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
537  av_strlcat(rtsp_st->control_url, "/",
538  sizeof(rtsp_st->control_url));
539  av_strlcat(rtsp_st->control_url, p,
540  sizeof(rtsp_st->control_url));
541  } else
542  av_strlcpy(rtsp_st->control_url, p,
543  sizeof(rtsp_st->control_url));
544  }
545  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
546  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
547  get_word(buf1, sizeof(buf1), &p);
548  payload_type = atoi(buf1);
549  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
550  if (rtsp_st->stream_index >= 0) {
551  st = s->streams[rtsp_st->stream_index];
552  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
553  }
554  s1->seen_rtpmap = 1;
555  if (s1->seen_fmtp) {
556  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
557  }
558  } else if (av_strstart(p, "fmtp:", &p) ||
559  av_strstart(p, "framesize:", &p)) {
560  // let dynamic protocol handlers have a stab at the line.
561  get_word(buf1, sizeof(buf1), &p);
562  payload_type = atoi(buf1);
563  if (s1->seen_rtpmap) {
564  parse_fmtp(s, rt, payload_type, buf);
565  } else {
566  s1->seen_fmtp = 1;
567  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
568  }
569  } else if (av_strstart(p, "range:", &p)) {
570  int64_t start, end;
571 
572  // this is so that seeking on a streamed file can work.
573  rtsp_parse_range_npt(p, &start, &end);
574  s->start_time = start;
575  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
576  s->duration = (end == AV_NOPTS_VALUE) ?
577  AV_NOPTS_VALUE : end - start;
578  } else if (av_strstart(p, "lang:", &p)) {
579  if (s->nb_streams > 0) {
580  get_word(buf1, sizeof(buf1), &p);
581  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
582  if (rtsp_st->stream_index >= 0) {
583  st = s->streams[rtsp_st->stream_index];
584  av_dict_set(&st->metadata, "language", buf1, 0);
585  }
586  } else
587  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
588  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
589  if (atoi(p) == 1)
591  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
592  s->nb_streams > 0) {
593  st = s->streams[s->nb_streams - 1];
594  st->codec->sample_rate = atoi(p);
595  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
596  // RFC 4568
597  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
598  get_word(buf1, sizeof(buf1), &p); // ignore tag
599  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
600  p += strspn(p, SPACE_CHARS);
601  if (av_strstart(p, "inline:", &p))
602  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
603  } else if (av_strstart(p, "source-filter:", &p)) {
604  int exclude = 0;
605  get_word(buf1, sizeof(buf1), &p);
606  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
607  return;
608  exclude = !strcmp(buf1, "excl");
609 
610  get_word(buf1, sizeof(buf1), &p);
611  if (strcmp(buf1, "IN") != 0)
612  return;
613  get_word(buf1, sizeof(buf1), &p);
614  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
615  return;
616  // not checking that the destination address actually matches or is wildcard
617  get_word(buf1, sizeof(buf1), &p);
618 
619  while (*p != '\0') {
620  rtsp_src = av_mallocz(sizeof(*rtsp_src));
621  if (!rtsp_src)
622  return;
623  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
624  if (exclude) {
625  if (s->nb_streams == 0) {
626  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
627  } else {
628  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
629  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
630  }
631  } else {
632  if (s->nb_streams == 0) {
633  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
634  } else {
635  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
636  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
637  }
638  }
639  }
640  } else {
641  if (rt->server_type == RTSP_SERVER_WMS)
643  if (s->nb_streams > 0) {
644  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
645 
646  if (rt->server_type == RTSP_SERVER_REAL)
647  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
648 
649  if (rtsp_st->dynamic_handler &&
651  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
652  rtsp_st->stream_index,
653  rtsp_st->dynamic_protocol_context, buf);
654  }
655  }
656  break;
657  }
658 }
659 
660 int ff_sdp_parse(AVFormatContext *s, const char *content)
661 {
662  RTSPState *rt = s->priv_data;
663  const char *p;
664  int letter, i;
665  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
666  * contain long SDP lines containing complete ASF Headers (several
667  * kB) or arrays of MDPR (RM stream descriptor) headers plus
668  * "rulebooks" describing their properties. Therefore, the SDP line
669  * buffer is large.
670  *
671  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
672  * in rtpdec_xiph.c. */
673  char buf[16384], *q;
674  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
675 
676  p = content;
677  for (;;) {
678  p += strspn(p, SPACE_CHARS);
679  letter = *p;
680  if (letter == '\0')
681  break;
682  p++;
683  if (*p != '=')
684  goto next_line;
685  p++;
686  /* get the content */
687  q = buf;
688  while (*p != '\n' && *p != '\r' && *p != '\0') {
689  if ((q - buf) < sizeof(buf) - 1)
690  *q++ = *p;
691  p++;
692  }
693  *q = '\0';
694  sdp_parse_line(s, s1, letter, buf);
695  next_line:
696  while (*p != '\n' && *p != '\0')
697  p++;
698  if (*p == '\n')
699  p++;
700  }
701 
702  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
703  av_freep(&s1->default_include_source_addrs[i]);
704  av_freep(&s1->default_include_source_addrs);
705  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
706  av_freep(&s1->default_exclude_source_addrs[i]);
707  av_freep(&s1->default_exclude_source_addrs);
708 
709  rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
710  if (!rt->p) return AVERROR(ENOMEM);
711  return 0;
712 }
713 #endif /* CONFIG_RTPDEC */
714 
715 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
716 {
717  RTSPState *rt = s->priv_data;
718  int i;
719 
720  for (i = 0; i < rt->nb_rtsp_streams; i++) {
721  RTSPStream *rtsp_st = rt->rtsp_streams[i];
722  if (!rtsp_st)
723  continue;
724  if (rtsp_st->transport_priv) {
725  if (s->oformat) {
726  AVFormatContext *rtpctx = rtsp_st->transport_priv;
727  av_write_trailer(rtpctx);
729  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
730  ff_rtsp_tcp_write_packet(s, rtsp_st);
731  ffio_free_dyn_buf(&rtpctx->pb);
732  } else {
733  avio_closep(&rtpctx->pb);
734  }
735  avformat_free_context(rtpctx);
736  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
738  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
740  }
741  rtsp_st->transport_priv = NULL;
742  if (rtsp_st->rtp_handle)
743  ffurl_close(rtsp_st->rtp_handle);
744  rtsp_st->rtp_handle = NULL;
745  }
746 }
747 
748 /* close and free RTSP streams */
750 {
751  RTSPState *rt = s->priv_data;
752  int i, j;
753  RTSPStream *rtsp_st;
754 
755  ff_rtsp_undo_setup(s, 0);
756  for (i = 0; i < rt->nb_rtsp_streams; i++) {
757  rtsp_st = rt->rtsp_streams[i];
758  if (rtsp_st) {
759  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
760  if (rtsp_st->dynamic_handler->close)
761  rtsp_st->dynamic_handler->close(
762  rtsp_st->dynamic_protocol_context);
764  }
765  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
766  av_freep(&rtsp_st->include_source_addrs[j]);
767  av_freep(&rtsp_st->include_source_addrs);
768  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
769  av_freep(&rtsp_st->exclude_source_addrs[j]);
770  av_freep(&rtsp_st->exclude_source_addrs);
771 
772  av_freep(&rtsp_st);
773  }
774  }
775  av_freep(&rt->rtsp_streams);
776  if (rt->asf_ctx) {
778  }
779  if (CONFIG_RTPDEC && rt->ts)
781  av_freep(&rt->p);
782  av_freep(&rt->recvbuf);
783 }
784 
786 {
787  RTSPState *rt = s->priv_data;
788  AVStream *st = NULL;
789  int reordering_queue_size = rt->reordering_queue_size;
790  if (reordering_queue_size < 0) {
792  reordering_queue_size = 0;
793  else
794  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
795  }
796 
797  /* open the RTP context */
798  if (rtsp_st->stream_index >= 0)
799  st = s->streams[rtsp_st->stream_index];
800  if (!st)
802 
803  if (CONFIG_RTSP_MUXER && s->oformat && st) {
804  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
805  s, st, rtsp_st->rtp_handle,
807  rtsp_st->stream_index);
808  /* Ownership of rtp_handle is passed to the rtp mux context */
809  rtsp_st->rtp_handle = NULL;
810  if (ret < 0)
811  return ret;
812  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
813  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
814  return 0; // Don't need to open any parser here
815  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
816  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
817  rtsp_st->dynamic_protocol_context,
818  rtsp_st->dynamic_handler);
819  else if (CONFIG_RTPDEC)
820  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
821  rtsp_st->sdp_payload_type,
822  reordering_queue_size);
823 
824  if (!rtsp_st->transport_priv) {
825  return AVERROR(ENOMEM);
826  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
827  if (rtsp_st->dynamic_handler) {
829  rtsp_st->dynamic_protocol_context,
830  rtsp_st->dynamic_handler);
831  }
832  if (rtsp_st->crypto_suite[0])
834  rtsp_st->crypto_suite,
835  rtsp_st->crypto_params);
836  }
837 
838  return 0;
839 }
840 
841 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
842 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
843 {
844  const char *q;
845  char *p;
846  int v;
847 
848  q = *pp;
849  q += strspn(q, SPACE_CHARS);
850  v = strtol(q, &p, 10);
851  if (*p == '-') {
852  p++;
853  *min_ptr = v;
854  v = strtol(p, &p, 10);
855  *max_ptr = v;
856  } else {
857  *min_ptr = v;
858  *max_ptr = v;
859  }
860  *pp = p;
861 }
862 
863 /* XXX: only one transport specification is parsed */
864 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
865 {
866  char transport_protocol[16];
867  char profile[16];
868  char lower_transport[16];
869  char parameter[16];
871  char buf[256];
872 
873  reply->nb_transports = 0;
874 
875  for (;;) {
876  p += strspn(p, SPACE_CHARS);
877  if (*p == '\0')
878  break;
879 
880  th = &reply->transports[reply->nb_transports];
881 
882  get_word_sep(transport_protocol, sizeof(transport_protocol),
883  "/", &p);
884  if (!av_strcasecmp (transport_protocol, "rtp")) {
885  get_word_sep(profile, sizeof(profile), "/;,", &p);
886  lower_transport[0] = '\0';
887  /* rtp/avp/<protocol> */
888  if (*p == '/') {
889  get_word_sep(lower_transport, sizeof(lower_transport),
890  ";,", &p);
891  }
893  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
894  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
895  /* x-pn-tng/<protocol> */
896  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
897  profile[0] = '\0';
899  } else if (!av_strcasecmp(transport_protocol, "raw")) {
900  get_word_sep(profile, sizeof(profile), "/;,", &p);
901  lower_transport[0] = '\0';
902  /* raw/raw/<protocol> */
903  if (*p == '/') {
904  get_word_sep(lower_transport, sizeof(lower_transport),
905  ";,", &p);
906  }
908  }
909  if (!av_strcasecmp(lower_transport, "TCP"))
911  else
913 
914  if (*p == ';')
915  p++;
916  /* get each parameter */
917  while (*p != '\0' && *p != ',') {
918  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
919  if (!strcmp(parameter, "port")) {
920  if (*p == '=') {
921  p++;
922  rtsp_parse_range(&th->port_min, &th->port_max, &p);
923  }
924  } else if (!strcmp(parameter, "client_port")) {
925  if (*p == '=') {
926  p++;
927  rtsp_parse_range(&th->client_port_min,
928  &th->client_port_max, &p);
929  }
930  } else if (!strcmp(parameter, "server_port")) {
931  if (*p == '=') {
932  p++;
933  rtsp_parse_range(&th->server_port_min,
934  &th->server_port_max, &p);
935  }
936  } else if (!strcmp(parameter, "interleaved")) {
937  if (*p == '=') {
938  p++;
939  rtsp_parse_range(&th->interleaved_min,
940  &th->interleaved_max, &p);
941  }
942  } else if (!strcmp(parameter, "multicast")) {
945  } else if (!strcmp(parameter, "ttl")) {
946  if (*p == '=') {
947  char *end;
948  p++;
949  th->ttl = strtol(p, &end, 10);
950  p = end;
951  }
952  } else if (!strcmp(parameter, "destination")) {
953  if (*p == '=') {
954  p++;
955  get_word_sep(buf, sizeof(buf), ";,", &p);
956  get_sockaddr(buf, &th->destination);
957  }
958  } else if (!strcmp(parameter, "source")) {
959  if (*p == '=') {
960  p++;
961  get_word_sep(buf, sizeof(buf), ";,", &p);
962  av_strlcpy(th->source, buf, sizeof(th->source));
963  }
964  } else if (!strcmp(parameter, "mode")) {
965  if (*p == '=') {
966  p++;
967  get_word_sep(buf, sizeof(buf), ";, ", &p);
968  if (!strcmp(buf, "record") ||
969  !strcmp(buf, "receive"))
970  th->mode_record = 1;
971  }
972  }
973 
974  while (*p != ';' && *p != '\0' && *p != ',')
975  p++;
976  if (*p == ';')
977  p++;
978  }
979  if (*p == ',')
980  p++;
981 
982  reply->nb_transports++;
983  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
984  break;
985  }
986 }
987 
988 static void handle_rtp_info(RTSPState *rt, const char *url,
989  uint32_t seq, uint32_t rtptime)
990 {
991  int i;
992  if (!rtptime || !url[0])
993  return;
994  if (rt->transport != RTSP_TRANSPORT_RTP)
995  return;
996  for (i = 0; i < rt->nb_rtsp_streams; i++) {
997  RTSPStream *rtsp_st = rt->rtsp_streams[i];
998  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
999  if (!rtpctx)
1000  continue;
1001  if (!strcmp(rtsp_st->control_url, url)) {
1002  rtpctx->base_timestamp = rtptime;
1003  break;
1004  }
1005  }
1006 }
1007 
1008 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1009 {
1010  int read = 0;
1011  char key[20], value[1024], url[1024] = "";
1012  uint32_t seq = 0, rtptime = 0;
1013 
1014  for (;;) {
1015  p += strspn(p, SPACE_CHARS);
1016  if (!*p)
1017  break;
1018  get_word_sep(key, sizeof(key), "=", &p);
1019  if (*p != '=')
1020  break;
1021  p++;
1022  get_word_sep(value, sizeof(value), ";, ", &p);
1023  read++;
1024  if (!strcmp(key, "url"))
1025  av_strlcpy(url, value, sizeof(url));
1026  else if (!strcmp(key, "seq"))
1027  seq = strtoul(value, NULL, 10);
1028  else if (!strcmp(key, "rtptime"))
1029  rtptime = strtoul(value, NULL, 10);
1030  if (*p == ',') {
1031  handle_rtp_info(rt, url, seq, rtptime);
1032  url[0] = '\0';
1033  seq = rtptime = 0;
1034  read = 0;
1035  }
1036  if (*p)
1037  p++;
1038  }
1039  if (read > 0)
1040  handle_rtp_info(rt, url, seq, rtptime);
1041 }
1042 
1043 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1044  RTSPState *rt, const char *method)
1045 {
1046  const char *p;
1047 
1048  /* NOTE: we do case independent match for broken servers */
1049  p = buf;
1050  if (av_stristart(p, "Session:", &p)) {
1051  int t;
1052  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1053  if (av_stristart(p, ";timeout=", &p) &&
1054  (t = strtol(p, NULL, 10)) > 0) {
1055  reply->timeout = t;
1056  }
1057  } else if (av_stristart(p, "Content-Length:", &p)) {
1058  reply->content_length = strtol(p, NULL, 10);
1059  } else if (av_stristart(p, "Transport:", &p)) {
1060  rtsp_parse_transport(reply, p);
1061  } else if (av_stristart(p, "CSeq:", &p)) {
1062  reply->seq = strtol(p, NULL, 10);
1063  } else if (av_stristart(p, "Range:", &p)) {
1064  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1065  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1066  p += strspn(p, SPACE_CHARS);
1067  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1068  } else if (av_stristart(p, "Server:", &p)) {
1069  p += strspn(p, SPACE_CHARS);
1070  av_strlcpy(reply->server, p, sizeof(reply->server));
1071  } else if (av_stristart(p, "Notice:", &p) ||
1072  av_stristart(p, "X-Notice:", &p)) {
1073  reply->notice = strtol(p, NULL, 10);
1074  } else if (av_stristart(p, "Location:", &p)) {
1075  p += strspn(p, SPACE_CHARS);
1076  av_strlcpy(reply->location, p , sizeof(reply->location));
1077  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1078  p += strspn(p, SPACE_CHARS);
1079  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1080  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1081  p += strspn(p, SPACE_CHARS);
1082  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1083  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1084  p += strspn(p, SPACE_CHARS);
1085  if (method && !strcmp(method, "DESCRIBE"))
1086  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1087  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1088  p += strspn(p, SPACE_CHARS);
1089  if (method && !strcmp(method, "PLAY"))
1090  rtsp_parse_rtp_info(rt, p);
1091  } else if (av_stristart(p, "Public:", &p) && rt) {
1092  if (strstr(p, "GET_PARAMETER") &&
1093  method && !strcmp(method, "OPTIONS"))
1094  rt->get_parameter_supported = 1;
1095  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1096  p += strspn(p, SPACE_CHARS);
1097  rt->accept_dynamic_rate = atoi(p);
1098  } else if (av_stristart(p, "Content-Type:", &p)) {
1099  p += strspn(p, SPACE_CHARS);
1100  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1101  }
1102 }
1103 
1104 /* skip a RTP/TCP interleaved packet */
1106 {
1107  RTSPState *rt = s->priv_data;
1108  int ret, len, len1;
1109  uint8_t buf[1024];
1110 
1111  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1112  if (ret != 3)
1113  return;
1114  len = AV_RB16(buf + 1);
1115 
1116  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1117 
1118  /* skip payload */
1119  while (len > 0) {
1120  len1 = len;
1121  if (len1 > sizeof(buf))
1122  len1 = sizeof(buf);
1123  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1124  if (ret != len1)
1125  return;
1126  len -= len1;
1127  }
1128 }
1129 
1131  unsigned char **content_ptr,
1132  int return_on_interleaved_data, const char *method)
1133 {
1134  RTSPState *rt = s->priv_data;
1135  char buf[4096], buf1[1024], *q;
1136  unsigned char ch;
1137  const char *p;
1138  int ret, content_length, line_count = 0, request = 0;
1139  int first_line = 1;
1140  unsigned char *content = NULL;
1141 
1142 start:
1143  line_count = 0;
1144  request = 0;
1145  content = NULL;
1146  memset(reply, 0, sizeof(*reply));
1147 
1148  /* parse reply (XXX: use buffers) */
1149  rt->last_reply[0] = '\0';
1150  for (;;) {
1151  q = buf;
1152  for (;;) {
1153  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1154  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1155  if (ret != 1)
1156  return AVERROR_EOF;
1157  if (ch == '\n')
1158  break;
1159  if (ch == '$' && first_line && q == buf) {
1160  if (return_on_interleaved_data) {
1161  return 1;
1162  } else
1164  } else if (ch != '\r') {
1165  if ((q - buf) < sizeof(buf) - 1)
1166  *q++ = ch;
1167  }
1168  }
1169  *q = '\0';
1170  first_line = 0;
1171 
1172  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1173 
1174  /* test if last line */
1175  if (buf[0] == '\0')
1176  break;
1177  p = buf;
1178  if (line_count == 0) {
1179  /* get reply code */
1180  get_word(buf1, sizeof(buf1), &p);
1181  if (!strncmp(buf1, "RTSP/", 5)) {
1182  get_word(buf1, sizeof(buf1), &p);
1183  reply->status_code = atoi(buf1);
1184  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1185  } else {
1186  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1187  get_word(buf1, sizeof(buf1), &p); // object
1188  request = 1;
1189  }
1190  } else {
1191  ff_rtsp_parse_line(reply, p, rt, method);
1192  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1193  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1194  }
1195  line_count++;
1196  }
1197 
1198  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1199  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1200 
1201  content_length = reply->content_length;
1202  if (content_length > 0) {
1203  /* leave some room for a trailing '\0' (useful for simple parsing) */
1204  content = av_malloc(content_length + 1);
1205  if (!content)
1206  return AVERROR(ENOMEM);
1207  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1208  content[content_length] = '\0';
1209  }
1210  if (content_ptr)
1211  *content_ptr = content;
1212  else
1213  av_freep(&content);
1214 
1215  if (request) {
1216  char buf[1024];
1217  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1218  const char* ptr = buf;
1219 
1220  if (!strcmp(reply->reason, "OPTIONS")) {
1221  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1222  if (reply->seq)
1223  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1224  if (reply->session_id[0])
1225  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1226  reply->session_id);
1227  } else {
1228  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1229  }
1230  av_strlcat(buf, "\r\n", sizeof(buf));
1231 
1232  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1233  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1234  ptr = base64buf;
1235  }
1236  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1237 
1239  /* Even if the request from the server had data, it is not the data
1240  * that the caller wants or expects. The memory could also be leaked
1241  * if the actual following reply has content data. */
1242  if (content_ptr)
1243  av_freep(content_ptr);
1244  /* If method is set, this is called from ff_rtsp_send_cmd,
1245  * where a reply to exactly this request is awaited. For
1246  * callers from within packet receiving, we just want to
1247  * return to the caller and go back to receiving packets. */
1248  if (method)
1249  goto start;
1250  return 0;
1251  }
1252 
1253  if (rt->seq != reply->seq) {
1254  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1255  rt->seq, reply->seq);
1256  }
1257 
1258  /* EOS */
1259  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1260  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1261  reply->notice == 2306 /* Continuous Feed Terminated */) {
1262  rt->state = RTSP_STATE_IDLE;
1263  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1264  return AVERROR(EIO); /* data or server error */
1265  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1266  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1267  return AVERROR(EPERM);
1268 
1269  return 0;
1270 }
1271 
1272 /**
1273  * Send a command to the RTSP server without waiting for the reply.
1274  *
1275  * @param s RTSP (de)muxer context
1276  * @param method the method for the request
1277  * @param url the target url for the request
1278  * @param headers extra header lines to include in the request
1279  * @param send_content if non-null, the data to send as request body content
1280  * @param send_content_length the length of the send_content data, or 0 if
1281  * send_content is null
1282  *
1283  * @return zero if success, nonzero otherwise
1284  */
1285 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1286  const char *method, const char *url,
1287  const char *headers,
1288  const unsigned char *send_content,
1289  int send_content_length)
1290 {
1291  RTSPState *rt = s->priv_data;
1292  char buf[4096], *out_buf;
1293  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1294 
1295  /* Add in RTSP headers */
1296  out_buf = buf;
1297  rt->seq++;
1298  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1299  if (headers)
1300  av_strlcat(buf, headers, sizeof(buf));
1301  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1302  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1303  if (rt->session_id[0] != '\0' && (!headers ||
1304  !strstr(headers, "\nIf-Match:"))) {
1305  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1306  }
1307  if (rt->auth[0]) {
1308  char *str = ff_http_auth_create_response(&rt->auth_state,
1309  rt->auth, url, method);
1310  if (str)
1311  av_strlcat(buf, str, sizeof(buf));
1312  av_free(str);
1313  }
1314  if (send_content_length > 0 && send_content)
1315  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1316  av_strlcat(buf, "\r\n", sizeof(buf));
1317 
1318  /* base64 encode rtsp if tunneling */
1319  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1320  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1321  out_buf = base64buf;
1322  }
1323 
1324  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1325 
1326  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1327  if (send_content_length > 0 && send_content) {
1328  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1329  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1330  "with content data not supported\n");
1331  return AVERROR_PATCHWELCOME;
1332  }
1333  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1334  }
1336 
1337  return 0;
1338 }
1339 
1340 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1341  const char *url, const char *headers)
1342 {
1343  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1344 }
1345 
1346 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1347  const char *headers, RTSPMessageHeader *reply,
1348  unsigned char **content_ptr)
1349 {
1350  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1351  content_ptr, NULL, 0);
1352 }
1353 
1355  const char *method, const char *url,
1356  const char *header,
1357  RTSPMessageHeader *reply,
1358  unsigned char **content_ptr,
1359  const unsigned char *send_content,
1360  int send_content_length)
1361 {
1362  RTSPState *rt = s->priv_data;
1363  HTTPAuthType cur_auth_type;
1364  int ret, attempts = 0;
1365 
1366 retry:
1367  cur_auth_type = rt->auth_state.auth_type;
1368  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1369  send_content,
1370  send_content_length)))
1371  return ret;
1372 
1373  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1374  return ret;
1375  attempts++;
1376 
1377  if (reply->status_code == 401 &&
1378  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1379  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1380  goto retry;
1381 
1382  if (reply->status_code > 400){
1383  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1384  method,
1385  reply->status_code,
1386  reply->reason);
1387  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1388  }
1389 
1390  return 0;
1391 }
1392 
1393 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1394  int lower_transport, const char *real_challenge)
1395 {
1396  RTSPState *rt = s->priv_data;
1397  int rtx = 0, j, i, err, interleave = 0, port_off;
1398  RTSPStream *rtsp_st;
1399  RTSPMessageHeader reply1, *reply = &reply1;
1400  char cmd[2048];
1401  const char *trans_pref;
1402 
1403  if (rt->transport == RTSP_TRANSPORT_RDT)
1404  trans_pref = "x-pn-tng";
1405  else if (rt->transport == RTSP_TRANSPORT_RAW)
1406  trans_pref = "RAW/RAW";
1407  else
1408  trans_pref = "RTP/AVP";
1409 
1410  /* default timeout: 1 minute */
1411  rt->timeout = 60;
1412 
1413  /* Choose a random starting offset within the first half of the
1414  * port range, to allow for a number of ports to try even if the offset
1415  * happens to be at the end of the random range. */
1416  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1417  /* even random offset */
1418  port_off -= port_off & 0x01;
1419 
1420  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1421  char transport[2048];
1422 
1423  /*
1424  * WMS serves all UDP data over a single connection, the RTX, which
1425  * isn't necessarily the first in the SDP but has to be the first
1426  * to be set up, else the second/third SETUP will fail with a 461.
1427  */
1428  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1429  rt->server_type == RTSP_SERVER_WMS) {
1430  if (i == 0) {
1431  /* rtx first */
1432  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1433  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1434  if (len >= 4 &&
1435  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1436  "/rtx"))
1437  break;
1438  }
1439  if (rtx == rt->nb_rtsp_streams)
1440  return -1; /* no RTX found */
1441  rtsp_st = rt->rtsp_streams[rtx];
1442  } else
1443  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1444  } else
1445  rtsp_st = rt->rtsp_streams[i];
1446 
1447  /* RTP/UDP */
1448  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1449  char buf[256];
1450 
1451  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1452  port = reply->transports[0].client_port_min;
1453  goto have_port;
1454  }
1455 
1456  /* first try in specified port range */
1457  while (j <= rt->rtp_port_max) {
1458  AVDictionary *opts = map_to_opts(rt);
1459 
1460  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1461  "?localport=%d", j);
1462  /* we will use two ports per rtp stream (rtp and rtcp) */
1463  j += 2;
1464  err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1465  &s->interrupt_callback, &opts);
1466 
1467  av_dict_free(&opts);
1468 
1469  if (!err)
1470  goto rtp_opened;
1471  }
1472  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1473  err = AVERROR(EIO);
1474  goto fail;
1475 
1476  rtp_opened:
1477  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1478  have_port:
1479  snprintf(transport, sizeof(transport) - 1,
1480  "%s/UDP;", trans_pref);
1481  if (rt->server_type != RTSP_SERVER_REAL)
1482  av_strlcat(transport, "unicast;", sizeof(transport));
1483  av_strlcatf(transport, sizeof(transport),
1484  "client_port=%d", port);
1485  if (rt->transport == RTSP_TRANSPORT_RTP &&
1486  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1487  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1488  }
1489 
1490  /* RTP/TCP */
1491  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1492  /* For WMS streams, the application streams are only used for
1493  * UDP. When trying to set it up for TCP streams, the server
1494  * will return an error. Therefore, we skip those streams. */
1495  if (rt->server_type == RTSP_SERVER_WMS &&
1496  (rtsp_st->stream_index < 0 ||
1497  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1499  continue;
1500  snprintf(transport, sizeof(transport) - 1,
1501  "%s/TCP;", trans_pref);
1502  if (rt->transport != RTSP_TRANSPORT_RDT)
1503  av_strlcat(transport, "unicast;", sizeof(transport));
1504  av_strlcatf(transport, sizeof(transport),
1505  "interleaved=%d-%d",
1506  interleave, interleave + 1);
1507  interleave += 2;
1508  }
1509 
1510  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1511  snprintf(transport, sizeof(transport) - 1,
1512  "%s/UDP;multicast", trans_pref);
1513  }
1514  if (s->oformat) {
1515  av_strlcat(transport, ";mode=record", sizeof(transport));
1516  } else if (rt->server_type == RTSP_SERVER_REAL ||
1518  av_strlcat(transport, ";mode=play", sizeof(transport));
1519  snprintf(cmd, sizeof(cmd),
1520  "Transport: %s\r\n",
1521  transport);
1522  if (rt->accept_dynamic_rate)
1523  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1524  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1525  char real_res[41], real_csum[9];
1526  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1527  real_challenge);
1528  av_strlcatf(cmd, sizeof(cmd),
1529  "If-Match: %s\r\n"
1530  "RealChallenge2: %s, sd=%s\r\n",
1531  rt->session_id, real_res, real_csum);
1532  }
1533  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1534  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1535  err = 1;
1536  goto fail;
1537  } else if (reply->status_code != RTSP_STATUS_OK ||
1538  reply->nb_transports != 1) {
1540  goto fail;
1541  }
1542 
1543  /* XXX: same protocol for all streams is required */
1544  if (i > 0) {
1545  if (reply->transports[0].lower_transport != rt->lower_transport ||
1546  reply->transports[0].transport != rt->transport) {
1547  err = AVERROR_INVALIDDATA;
1548  goto fail;
1549  }
1550  } else {
1551  rt->lower_transport = reply->transports[0].lower_transport;
1552  rt->transport = reply->transports[0].transport;
1553  }
1554 
1555  /* Fail if the server responded with another lower transport mode
1556  * than what we requested. */
1557  if (reply->transports[0].lower_transport != lower_transport) {
1558  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1559  err = AVERROR_INVALIDDATA;
1560  goto fail;
1561  }
1562 
1563  switch(reply->transports[0].lower_transport) {
1565  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1566  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1567  break;
1568 
1569  case RTSP_LOWER_TRANSPORT_UDP: {
1570  char url[1024], options[30] = "";
1571  const char *peer = host;
1572 
1573  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1574  av_strlcpy(options, "?connect=1", sizeof(options));
1575  /* Use source address if specified */
1576  if (reply->transports[0].source[0])
1577  peer = reply->transports[0].source;
1578  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1579  reply->transports[0].server_port_min, "%s", options);
1580  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1581  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1582  err = AVERROR_INVALIDDATA;
1583  goto fail;
1584  }
1585  break;
1586  }
1588  char url[1024], namebuf[50], optbuf[20] = "";
1589  struct sockaddr_storage addr;
1590  int port, ttl;
1591 
1592  if (reply->transports[0].destination.ss_family) {
1593  addr = reply->transports[0].destination;
1594  port = reply->transports[0].port_min;
1595  ttl = reply->transports[0].ttl;
1596  } else {
1597  addr = rtsp_st->sdp_ip;
1598  port = rtsp_st->sdp_port;
1599  ttl = rtsp_st->sdp_ttl;
1600  }
1601  if (ttl > 0)
1602  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1603  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1604  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1605  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1606  port, "%s", optbuf);
1607  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1608  &s->interrupt_callback, NULL) < 0) {
1609  err = AVERROR_INVALIDDATA;
1610  goto fail;
1611  }
1612  break;
1613  }
1614  }
1615 
1616  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1617  goto fail;
1618  }
1619 
1620  if (rt->nb_rtsp_streams && reply->timeout > 0)
1621  rt->timeout = reply->timeout;
1622 
1623  if (rt->server_type == RTSP_SERVER_REAL)
1624  rt->need_subscription = 1;
1625 
1626  return 0;
1627 
1628 fail:
1629  ff_rtsp_undo_setup(s, 0);
1630  return err;
1631 }
1632 
1634 {
1635  RTSPState *rt = s->priv_data;
1636  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1637  ffurl_close(rt->rtsp_hd);
1638  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1639 }
1640 
1642 {
1643  RTSPState *rt = s->priv_data;
1644  char proto[128], host[1024], path[1024];
1645  char tcpname[1024], cmd[2048], auth[128];
1646  const char *lower_rtsp_proto = "tcp";
1647  int port, err, tcp_fd;
1648  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1649  int lower_transport_mask = 0;
1650  int default_port = RTSP_DEFAULT_PORT;
1651  char real_challenge[64] = "";
1652  struct sockaddr_storage peer;
1653  socklen_t peer_len = sizeof(peer);
1654 
1655  if (rt->rtp_port_max < rt->rtp_port_min) {
1656  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1657  "than min port %d\n", rt->rtp_port_max,
1658  rt->rtp_port_min);
1659  return AVERROR(EINVAL);
1660  }
1661 
1662  if (!ff_network_init())
1663  return AVERROR(EIO);
1664 
1665  if (s->max_delay < 0) /* Not set by the caller */
1667 
1672  }
1673  /* Only pass through valid flags from here */
1675 
1676 redirect:
1677  /* extract hostname and port */
1678  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1679  host, sizeof(host), &port, path, sizeof(path), s->filename);
1680 
1681  if (!strcmp(proto, "rtsps")) {
1682  lower_rtsp_proto = "tls";
1683  default_port = RTSPS_DEFAULT_PORT;
1685  }
1686 
1687  if (*auth) {
1688  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1689  }
1690  if (port < 0)
1691  port = default_port;
1692 
1693  lower_transport_mask = rt->lower_transport_mask;
1694 
1695  if (!lower_transport_mask)
1696  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1697 
1698  if (s->oformat) {
1699  /* Only UDP or TCP - UDP multicast isn't supported. */
1700  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1701  (1 << RTSP_LOWER_TRANSPORT_TCP);
1702  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1703  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1704  "only UDP and TCP are supported for output.\n");
1705  err = AVERROR(EINVAL);
1706  goto fail;
1707  }
1708  }
1709 
1710  /* Construct the URI used in request; this is similar to s->filename,
1711  * but with authentication credentials removed and RTSP specific options
1712  * stripped out. */
1713  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1714  host, port, "%s", path);
1715 
1716  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1717  /* set up initial handshake for tunneling */
1718  char httpname[1024];
1719  char sessioncookie[17];
1720  char headers[1024];
1721 
1722  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1723  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1725 
1726  /* GET requests */
1727  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1728  &s->interrupt_callback) < 0) {
1729  err = AVERROR(EIO);
1730  goto fail;
1731  }
1732 
1733  /* generate GET headers */
1734  snprintf(headers, sizeof(headers),
1735  "x-sessioncookie: %s\r\n"
1736  "Accept: application/x-rtsp-tunnelled\r\n"
1737  "Pragma: no-cache\r\n"
1738  "Cache-Control: no-cache\r\n",
1739  sessioncookie);
1740  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1741 
1742  /* complete the connection */
1743  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1744  err = AVERROR(EIO);
1745  goto fail;
1746  }
1747 
1748  /* POST requests */
1749  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1750  &s->interrupt_callback) < 0 ) {
1751  err = AVERROR(EIO);
1752  goto fail;
1753  }
1754 
1755  /* generate POST headers */
1756  snprintf(headers, sizeof(headers),
1757  "x-sessioncookie: %s\r\n"
1758  "Content-Type: application/x-rtsp-tunnelled\r\n"
1759  "Pragma: no-cache\r\n"
1760  "Cache-Control: no-cache\r\n"
1761  "Content-Length: 32767\r\n"
1762  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1763  sessioncookie);
1764  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1765  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1766 
1767  /* Initialize the authentication state for the POST session. The HTTP
1768  * protocol implementation doesn't properly handle multi-pass
1769  * authentication for POST requests, since it would require one of
1770  * the following:
1771  * - implementing Expect: 100-continue, which many HTTP servers
1772  * don't support anyway, even less the RTSP servers that do HTTP
1773  * tunneling
1774  * - sending the whole POST data until getting a 401 reply specifying
1775  * what authentication method to use, then resending all that data
1776  * - waiting for potential 401 replies directly after sending the
1777  * POST header (waiting for some unspecified time)
1778  * Therefore, we copy the full auth state, which works for both basic
1779  * and digest. (For digest, we would have to synchronize the nonce
1780  * count variable between the two sessions, if we'd do more requests
1781  * with the original session, though.)
1782  */
1784 
1785  /* complete the connection */
1786  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1787  err = AVERROR(EIO);
1788  goto fail;
1789  }
1790  } else {
1791  int ret;
1792  /* open the tcp connection */
1793  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1794  host, port,
1795  "?timeout=%d", rt->stimeout);
1796  if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1797  &s->interrupt_callback, NULL)) < 0) {
1798  err = ret;
1799  goto fail;
1800  }
1801  rt->rtsp_hd_out = rt->rtsp_hd;
1802  }
1803  rt->seq = 0;
1804 
1805  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1806  if (tcp_fd < 0) {
1807  err = tcp_fd;
1808  goto fail;
1809  }
1810  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1811  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1812  NULL, 0, NI_NUMERICHOST);
1813  }
1814 
1815  /* request options supported by the server; this also detects server
1816  * type */
1817  for (rt->server_type = RTSP_SERVER_RTP;;) {
1818  cmd[0] = 0;
1819  if (rt->server_type == RTSP_SERVER_REAL)
1820  av_strlcat(cmd,
1821  /*
1822  * The following entries are required for proper
1823  * streaming from a Realmedia server. They are
1824  * interdependent in some way although we currently
1825  * don't quite understand how. Values were copied
1826  * from mplayer SVN r23589.
1827  * ClientChallenge is a 16-byte ID in hex
1828  * CompanyID is a 16-byte ID in base64
1829  */
1830  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1831  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1832  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1833  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1834  sizeof(cmd));
1835  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1836  if (reply->status_code != RTSP_STATUS_OK) {
1838  goto fail;
1839  }
1840 
1841  /* detect server type if not standard-compliant RTP */
1842  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1844  continue;
1845  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1847  } else if (rt->server_type == RTSP_SERVER_REAL)
1848  strcpy(real_challenge, reply->real_challenge);
1849  break;
1850  }
1851 
1852  if (CONFIG_RTSP_DEMUXER && s->iformat)
1853  err = ff_rtsp_setup_input_streams(s, reply);
1854  else if (CONFIG_RTSP_MUXER)
1855  err = ff_rtsp_setup_output_streams(s, host);
1856  else
1857  av_assert0(0);
1858  if (err)
1859  goto fail;
1860 
1861  do {
1862  int lower_transport = ff_log2_tab[lower_transport_mask &
1863  ~(lower_transport_mask - 1)];
1864 
1865  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1866  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1867  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1868 
1869  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1870  rt->server_type == RTSP_SERVER_REAL ?
1871  real_challenge : NULL);
1872  if (err < 0)
1873  goto fail;
1874  lower_transport_mask &= ~(1 << lower_transport);
1875  if (lower_transport_mask == 0 && err == 1) {
1876  err = AVERROR(EPROTONOSUPPORT);
1877  goto fail;
1878  }
1879  } while (err);
1880 
1881  rt->lower_transport_mask = lower_transport_mask;
1882  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1883  rt->state = RTSP_STATE_IDLE;
1884  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1885  return 0;
1886  fail:
1889  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1890  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1891  rt->session_id[0] = '\0';
1892  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1893  reply->status_code,
1894  s->filename);
1895  goto redirect;
1896  }
1897  ff_network_close();
1898  return err;
1899 }
1900 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1901 
1902 #if CONFIG_RTPDEC
1903 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1904  uint8_t *buf, int buf_size, int64_t wait_end)
1905 {
1906  RTSPState *rt = s->priv_data;
1907  RTSPStream *rtsp_st;
1908  int n, i, ret, tcp_fd, timeout_cnt = 0;
1909  int max_p = 0;
1910  struct pollfd *p = rt->p;
1911  int *fds = NULL, fdsnum, fdsidx;
1912 
1913  for (;;) {
1915  return AVERROR_EXIT;
1916  if (wait_end && wait_end - av_gettime_relative() < 0)
1917  return AVERROR(EAGAIN);
1918  max_p = 0;
1919  if (rt->rtsp_hd) {
1920  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1921  p[max_p].fd = tcp_fd;
1922  p[max_p++].events = POLLIN;
1923  } else {
1924  tcp_fd = -1;
1925  }
1926  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1927  rtsp_st = rt->rtsp_streams[i];
1928  if (rtsp_st->rtp_handle) {
1929  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1930  &fds, &fdsnum)) {
1931  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1932  return ret;
1933  }
1934  if (fdsnum != 2) {
1935  av_log(s, AV_LOG_ERROR,
1936  "Number of fds %d not supported\n", fdsnum);
1937  return AVERROR_INVALIDDATA;
1938  }
1939  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1940  p[max_p].fd = fds[fdsidx];
1941  p[max_p++].events = POLLIN;
1942  }
1943  av_freep(&fds);
1944  }
1945  }
1946  n = poll(p, max_p, POLL_TIMEOUT_MS);
1947  if (n > 0) {
1948  int j = 1 - (tcp_fd == -1);
1949  timeout_cnt = 0;
1950  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1951  rtsp_st = rt->rtsp_streams[i];
1952  if (rtsp_st->rtp_handle) {
1953  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1954  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1955  if (ret > 0) {
1956  *prtsp_st = rtsp_st;
1957  return ret;
1958  }
1959  }
1960  j+=2;
1961  }
1962  }
1963 #if CONFIG_RTSP_DEMUXER
1964  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1965  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1966  if (rt->state == RTSP_STATE_STREAMING) {
1968  return AVERROR_EOF;
1969  else
1971  "Unable to answer to TEARDOWN\n");
1972  } else
1973  return 0;
1974  } else {
1975  RTSPMessageHeader reply;
1976  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1977  if (ret < 0)
1978  return ret;
1979  /* XXX: parse message */
1980  if (rt->state != RTSP_STATE_STREAMING)
1981  return 0;
1982  }
1983  }
1984 #endif
1985  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1986  return AVERROR(ETIMEDOUT);
1987  } else if (n < 0 && errno != EINTR)
1988  return AVERROR(errno);
1989  }
1990 }
1991 
1992 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1993  const uint8_t *buf, int len)
1994 {
1995  RTSPState *rt = s->priv_data;
1996  int i;
1997  if (len < 0)
1998  return len;
1999  if (rt->nb_rtsp_streams == 1) {
2000  *rtsp_st = rt->rtsp_streams[0];
2001  return len;
2002  }
2003  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2004  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2005  int no_ssrc = 0;
2006  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2007  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2008  if (!rtpctx)
2009  continue;
2010  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2011  *rtsp_st = rt->rtsp_streams[i];
2012  return len;
2013  }
2014  if (!rtpctx->ssrc)
2015  no_ssrc = 1;
2016  }
2017  if (no_ssrc) {
2019  "Unable to pick stream for packet - SSRC not known for "
2020  "all streams\n");
2021  return AVERROR(EAGAIN);
2022  }
2023  } else {
2024  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2025  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2026  *rtsp_st = rt->rtsp_streams[i];
2027  return len;
2028  }
2029  }
2030  }
2031  }
2032  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2033  return AVERROR(EAGAIN);
2034 }
2035 
2037 {
2038  RTSPState *rt = s->priv_data;
2039  int ret, len;
2040  RTSPStream *rtsp_st, *first_queue_st = NULL;
2041  int64_t wait_end = 0;
2042 
2043  if (rt->nb_byes == rt->nb_rtsp_streams)
2044  return AVERROR_EOF;
2045 
2046  /* get next frames from the same RTP packet */
2047  if (rt->cur_transport_priv) {
2048  if (rt->transport == RTSP_TRANSPORT_RDT) {
2049  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2050  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2051  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2052  } else if (CONFIG_RTPDEC && rt->ts) {
2053  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2054  if (ret >= 0) {
2055  rt->recvbuf_pos += ret;
2056  ret = rt->recvbuf_pos < rt->recvbuf_len;
2057  }
2058  } else
2059  ret = -1;
2060  if (ret == 0) {
2061  rt->cur_transport_priv = NULL;
2062  return 0;
2063  } else if (ret == 1) {
2064  return 0;
2065  } else
2066  rt->cur_transport_priv = NULL;
2067  }
2068 
2069 redo:
2070  if (rt->transport == RTSP_TRANSPORT_RTP) {
2071  int i;
2072  int64_t first_queue_time = 0;
2073  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2075  int64_t queue_time;
2076  if (!rtpctx)
2077  continue;
2078  queue_time = ff_rtp_queued_packet_time(rtpctx);
2079  if (queue_time && (queue_time - first_queue_time < 0 ||
2080  !first_queue_time)) {
2081  first_queue_time = queue_time;
2082  first_queue_st = rt->rtsp_streams[i];
2083  }
2084  }
2085  if (first_queue_time) {
2086  wait_end = first_queue_time + s->max_delay;
2087  } else {
2088  wait_end = 0;
2089  first_queue_st = NULL;
2090  }
2091  }
2092 
2093  /* read next RTP packet */
2094  if (!rt->recvbuf) {
2096  if (!rt->recvbuf)
2097  return AVERROR(ENOMEM);
2098  }
2099 
2100  switch(rt->lower_transport) {
2101  default:
2102 #if CONFIG_RTSP_DEMUXER
2104  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2105  break;
2106 #endif
2109  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2110  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2111  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2112  break;
2114  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2115  wait_end && wait_end < av_gettime_relative())
2116  len = AVERROR(EAGAIN);
2117  else
2118  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2119  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2120  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2122  break;
2123  }
2124  if (len == AVERROR(EAGAIN) && first_queue_st &&
2125  rt->transport == RTSP_TRANSPORT_RTP) {
2126  rtsp_st = first_queue_st;
2127  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2128  goto end;
2129  }
2130  if (len < 0)
2131  return len;
2132  if (len == 0)
2133  return AVERROR_EOF;
2134  if (rt->transport == RTSP_TRANSPORT_RDT) {
2135  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2136  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2137  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2138  if (rtsp_st->feedback) {
2139  AVIOContext *pb = NULL;
2141  pb = s->pb;
2142  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2143  }
2144  if (ret < 0) {
2145  /* Either bad packet, or a RTCP packet. Check if the
2146  * first_rtcp_ntp_time field was initialized. */
2147  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2148  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2149  /* first_rtcp_ntp_time has been initialized for this stream,
2150  * copy the same value to all other uninitialized streams,
2151  * in order to map their timestamp origin to the same ntp time
2152  * as this one. */
2153  int i;
2154  AVStream *st = NULL;
2155  if (rtsp_st->stream_index >= 0)
2156  st = s->streams[rtsp_st->stream_index];
2157  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2158  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2159  AVStream *st2 = NULL;
2160  if (rt->rtsp_streams[i]->stream_index >= 0)
2161  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2162  if (rtpctx2 && st && st2 &&
2163  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2164  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2165  rtpctx2->rtcp_ts_offset = av_rescale_q(
2166  rtpctx->rtcp_ts_offset, st->time_base,
2167  st2->time_base);
2168  }
2169  }
2170  // Make real NTP start time available in AVFormatContext
2171  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2172  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2173  if (rtpctx->st) {
2174  s->start_time_realtime -=
2175  av_rescale (rtpctx->rtcp_ts_offset,
2176  (uint64_t) rtpctx->st->time_base.num * 1000000,
2177  rtpctx->st->time_base.den);
2178  }
2179  }
2180  }
2181  if (ret == -RTCP_BYE) {
2182  rt->nb_byes++;
2183 
2184  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2185  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2186 
2187  if (rt->nb_byes == rt->nb_rtsp_streams)
2188  return AVERROR_EOF;
2189  }
2190  }
2191  } else if (CONFIG_RTPDEC && rt->ts) {
2192  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2193  if (ret >= 0) {
2194  if (ret < len) {
2195  rt->recvbuf_len = len;
2196  rt->recvbuf_pos = ret;
2197  rt->cur_transport_priv = rt->ts;
2198  return 1;
2199  } else {
2200  ret = 0;
2201  }
2202  }
2203  } else {
2204  return AVERROR_INVALIDDATA;
2205  }
2206 end:
2207  if (ret < 0)
2208  goto redo;
2209  if (ret == 1)
2210  /* more packets may follow, so we save the RTP context */
2211  rt->cur_transport_priv = rtsp_st->transport_priv;
2212 
2213  return ret;
2214 }
2215 #endif /* CONFIG_RTPDEC */
2216 
2217 #if CONFIG_SDP_DEMUXER
2218 static int sdp_probe(AVProbeData *p1)
2219 {
2220  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2221 
2222  /* we look for a line beginning "c=IN IP" */
2223  while (p < p_end && *p != '\0') {
2224  if (sizeof("c=IN IP") - 1 < p_end - p &&
2225  av_strstart(p, "c=IN IP", NULL))
2226  return AVPROBE_SCORE_EXTENSION;
2227 
2228  while (p < p_end - 1 && *p != '\n') p++;
2229  if (++p >= p_end)
2230  break;
2231  if (*p == '\r')
2232  p++;
2233  }
2234  return 0;
2235 }
2236 
2237 static void append_source_addrs(char *buf, int size, const char *name,
2238  int count, struct RTSPSource **addrs)
2239 {
2240  int i;
2241  if (!count)
2242  return;
2243  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2244  for (i = 1; i < count; i++)
2245  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2246 }
2247 
2248 static int sdp_read_header(AVFormatContext *s)
2249 {
2250  RTSPState *rt = s->priv_data;
2251  RTSPStream *rtsp_st;
2252  int size, i, err;
2253  char *content;
2254  char url[1024];
2255 
2256  if (!ff_network_init())
2257  return AVERROR(EIO);
2258 
2259  if (s->max_delay < 0) /* Not set by the caller */
2261  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2263 
2264  /* read the whole sdp file */
2265  /* XXX: better loading */
2266  content = av_malloc(SDP_MAX_SIZE);
2267  if (!content)
2268  return AVERROR(ENOMEM);
2269  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2270  if (size <= 0) {
2271  av_free(content);
2272  return AVERROR_INVALIDDATA;
2273  }
2274  content[size] ='\0';
2275 
2276  err = ff_sdp_parse(s, content);
2277  av_freep(&content);
2278  if (err) goto fail;
2279 
2280  /* open each RTP stream */
2281  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2282  char namebuf[50];
2283  rtsp_st = rt->rtsp_streams[i];
2284 
2285  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2286  AVDictionary *opts = map_to_opts(rt);
2287 
2288  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2289  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2290  ff_url_join(url, sizeof(url), "rtp", NULL,
2291  namebuf, rtsp_st->sdp_port,
2292  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2293  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2294  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2295  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2296 
2297  append_source_addrs(url, sizeof(url), "sources",
2298  rtsp_st->nb_include_source_addrs,
2299  rtsp_st->include_source_addrs);
2300  append_source_addrs(url, sizeof(url), "block",
2301  rtsp_st->nb_exclude_source_addrs,
2302  rtsp_st->exclude_source_addrs);
2303  err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2304  &s->interrupt_callback, &opts);
2305 
2306  av_dict_free(&opts);
2307 
2308  if (err < 0) {
2309  err = AVERROR_INVALIDDATA;
2310  goto fail;
2311  }
2312  }
2313  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2314  goto fail;
2315  }
2316  return 0;
2317 fail:
2319  ff_network_close();
2320  return err;
2321 }
2322 
2323 static int sdp_read_close(AVFormatContext *s)
2324 {
2326  ff_network_close();
2327  return 0;
2328 }
2329 
2330 static const AVClass sdp_demuxer_class = {
2331  .class_name = "SDP demuxer",
2332  .item_name = av_default_item_name,
2333  .option = sdp_options,
2334  .version = LIBAVUTIL_VERSION_INT,
2335 };
2336 
2337 AVInputFormat ff_sdp_demuxer = {
2338  .name = "sdp",
2339  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2340  .priv_data_size = sizeof(RTSPState),
2341  .read_probe = sdp_probe,
2342  .read_header = sdp_read_header,
2344  .read_close = sdp_read_close,
2345  .priv_class = &sdp_demuxer_class,
2346 };
2347 #endif /* CONFIG_SDP_DEMUXER */
2348 
2349 #if CONFIG_RTP_DEMUXER
2350 static int rtp_probe(AVProbeData *p)
2351 {
2352  if (av_strstart(p->filename, "rtp:", NULL))
2353  return AVPROBE_SCORE_MAX;
2354  return 0;
2355 }
2356 
2357 static int rtp_read_header(AVFormatContext *s)
2358 {
2359  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2360  char host[500], sdp[500];
2361  int ret, port;
2362  URLContext* in = NULL;
2363  int payload_type;
2364  AVCodecContext codec = { 0 };
2365  struct sockaddr_storage addr;
2366  AVIOContext pb;
2367  socklen_t addrlen = sizeof(addr);
2368  RTSPState *rt = s->priv_data;
2369 
2370  if (!ff_network_init())
2371  return AVERROR(EIO);
2372 
2373  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2374  &s->interrupt_callback, NULL);
2375  if (ret)
2376  goto fail;
2377 
2378  while (1) {
2379  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2380  if (ret == AVERROR(EAGAIN))
2381  continue;
2382  if (ret < 0)
2383  goto fail;
2384  if (ret < 12) {
2385  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2386  continue;
2387  }
2388 
2389  if ((recvbuf[0] & 0xc0) != 0x80) {
2390  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2391  "received\n");
2392  continue;
2393  }
2394 
2395  if (RTP_PT_IS_RTCP(recvbuf[1]))
2396  continue;
2397 
2398  payload_type = recvbuf[1] & 0x7f;
2399  break;
2400  }
2401  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2402  ffurl_close(in);
2403  in = NULL;
2404 
2405  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2406  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2407  "without an SDP file describing it\n",
2408  payload_type);
2409  goto fail;
2410  }
2411  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2412  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2413  "properly you need an SDP file "
2414  "describing it\n");
2415  }
2416 
2417  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2418  NULL, 0, s->filename);
2419 
2420  snprintf(sdp, sizeof(sdp),
2421  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2422  addr.ss_family == AF_INET ? 4 : 6, host,
2423  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2424  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2425  port, payload_type);
2426  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2427 
2428  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2429  s->pb = &pb;
2430 
2431  /* sdp_read_header initializes this again */
2432  ff_network_close();
2433 
2434  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2435 
2436  ret = sdp_read_header(s);
2437  s->pb = NULL;
2438  return ret;
2439 
2440 fail:
2441  if (in)
2442  ffurl_close(in);
2443  ff_network_close();
2444  return ret;
2445 }
2446 
2447 static const AVClass rtp_demuxer_class = {
2448  .class_name = "RTP demuxer",
2449  .item_name = av_default_item_name,
2450  .option = rtp_options,
2451  .version = LIBAVUTIL_VERSION_INT,
2452 };
2453 
2454 AVInputFormat ff_rtp_demuxer = {
2455  .name = "rtp",
2456  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2457  .priv_data_size = sizeof(RTSPState),
2458  .read_probe = rtp_probe,
2459  .read_header = rtp_read_header,
2461  .read_close = sdp_read_close,
2462  .flags = AVFMT_NOFILE,
2463  .priv_class = &rtp_demuxer_class,
2464 };
2465 #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:3963
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:1498
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1523
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:449
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:4083
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:367
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
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:785
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2776
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:195
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:843
#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:1322
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
#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:3472
#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:1273
#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
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:1232
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
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:849
enum AVStreamParseType need_parsing
Definition: avformat.h:1034
#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:3749
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1341
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
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:3807
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:538
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1292
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:140
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:270
#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:1485
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:578
int profile
Definition: mxfenc.c:1806
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
av_default_item_name
#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:175
#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:3479
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:606
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:861
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:451
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:450
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:1329
#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:1349
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:127
#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:81
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
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: <xyz>[;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:916
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:571
#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:1174
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:187
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:623
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:441
Stream structure.
Definition: avformat.h:842
#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:1510
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:715
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
Definition: avio.c:600
enum AVCodecID codec_id
Definition: avcodec.h:1519
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:2262
AVIOContext * pb
I/O context.
Definition: avformat.h:1315
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:1502
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:3016
#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:2795
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:749
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:458
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:3686
int buffer_size
Definition: rtsp.h:410
This structure contains the data a format has to probe a file.
Definition: avformat.h:448
#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:412
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:1358
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:460
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:465
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:292
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:360
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:1285
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:3721
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:2263
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:446
void * priv_data
Format private data.
Definition: avformat.h:1301
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:1368
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:628
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:109
#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:884
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:2820
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:1400
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:959
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:353
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