FFmpeg  2.8.15
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Modules Pages
rtsp.c
Go to the documentation of this file.
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  unsigned char *content = NULL;
1140 
1141 start:
1142  line_count = 0;
1143  request = 0;
1144  content = NULL;
1145  memset(reply, 0, sizeof(*reply));
1146 
1147  /* parse reply (XXX: use buffers) */
1148  rt->last_reply[0] = '\0';
1149  for (;;) {
1150  q = buf;
1151  for (;;) {
1152  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1153  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1154  if (ret != 1)
1155  return AVERROR_EOF;
1156  if (ch == '\n')
1157  break;
1158  if (ch == '$' && q == buf) {
1159  if (return_on_interleaved_data) {
1160  return 1;
1161  } else
1163  } else if (ch != '\r') {
1164  if ((q - buf) < sizeof(buf) - 1)
1165  *q++ = ch;
1166  }
1167  }
1168  *q = '\0';
1169 
1170  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1171 
1172  /* test if last line */
1173  if (buf[0] == '\0')
1174  break;
1175  p = buf;
1176  if (line_count == 0) {
1177  /* get reply code */
1178  get_word(buf1, sizeof(buf1), &p);
1179  if (!strncmp(buf1, "RTSP/", 5)) {
1180  get_word(buf1, sizeof(buf1), &p);
1181  reply->status_code = atoi(buf1);
1182  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1183  } else {
1184  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1185  get_word(buf1, sizeof(buf1), &p); // object
1186  request = 1;
1187  }
1188  } else {
1189  ff_rtsp_parse_line(reply, p, rt, method);
1190  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1191  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1192  }
1193  line_count++;
1194  }
1195 
1196  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1197  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1198 
1199  content_length = reply->content_length;
1200  if (content_length > 0) {
1201  /* leave some room for a trailing '\0' (useful for simple parsing) */
1202  content = av_malloc(content_length + 1);
1203  if (!content)
1204  return AVERROR(ENOMEM);
1205  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1206  content[content_length] = '\0';
1207  }
1208  if (content_ptr)
1209  *content_ptr = content;
1210  else
1211  av_freep(&content);
1212 
1213  if (request) {
1214  char buf[1024];
1215  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1216  const char* ptr = buf;
1217 
1218  if (!strcmp(reply->reason, "OPTIONS")) {
1219  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1220  if (reply->seq)
1221  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1222  if (reply->session_id[0])
1223  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1224  reply->session_id);
1225  } else {
1226  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1227  }
1228  av_strlcat(buf, "\r\n", sizeof(buf));
1229 
1230  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1231  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1232  ptr = base64buf;
1233  }
1234  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1235 
1237  /* Even if the request from the server had data, it is not the data
1238  * that the caller wants or expects. The memory could also be leaked
1239  * if the actual following reply has content data. */
1240  if (content_ptr)
1241  av_freep(content_ptr);
1242  /* If method is set, this is called from ff_rtsp_send_cmd,
1243  * where a reply to exactly this request is awaited. For
1244  * callers from within packet receiving, we just want to
1245  * return to the caller and go back to receiving packets. */
1246  if (method)
1247  goto start;
1248  return 0;
1249  }
1250 
1251  if (rt->seq != reply->seq) {
1252  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1253  rt->seq, reply->seq);
1254  }
1255 
1256  /* EOS */
1257  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1258  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1259  reply->notice == 2306 /* Continuous Feed Terminated */) {
1260  rt->state = RTSP_STATE_IDLE;
1261  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1262  return AVERROR(EIO); /* data or server error */
1263  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1264  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1265  return AVERROR(EPERM);
1266 
1267  return 0;
1268 }
1269 
1270 /**
1271  * Send a command to the RTSP server without waiting for the reply.
1272  *
1273  * @param s RTSP (de)muxer context
1274  * @param method the method for the request
1275  * @param url the target url for the request
1276  * @param headers extra header lines to include in the request
1277  * @param send_content if non-null, the data to send as request body content
1278  * @param send_content_length the length of the send_content data, or 0 if
1279  * send_content is null
1280  *
1281  * @return zero if success, nonzero otherwise
1282  */
1283 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1284  const char *method, const char *url,
1285  const char *headers,
1286  const unsigned char *send_content,
1287  int send_content_length)
1288 {
1289  RTSPState *rt = s->priv_data;
1290  char buf[4096], *out_buf;
1291  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1292 
1293  /* Add in RTSP headers */
1294  out_buf = buf;
1295  rt->seq++;
1296  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1297  if (headers)
1298  av_strlcat(buf, headers, sizeof(buf));
1299  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1300  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1301  if (rt->session_id[0] != '\0' && (!headers ||
1302  !strstr(headers, "\nIf-Match:"))) {
1303  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1304  }
1305  if (rt->auth[0]) {
1306  char *str = ff_http_auth_create_response(&rt->auth_state,
1307  rt->auth, url, method);
1308  if (str)
1309  av_strlcat(buf, str, sizeof(buf));
1310  av_free(str);
1311  }
1312  if (send_content_length > 0 && send_content)
1313  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1314  av_strlcat(buf, "\r\n", sizeof(buf));
1315 
1316  /* base64 encode rtsp if tunneling */
1317  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1318  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1319  out_buf = base64buf;
1320  }
1321 
1322  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1323 
1324  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1325  if (send_content_length > 0 && send_content) {
1326  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1327  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1328  "with content data not supported\n");
1329  return AVERROR_PATCHWELCOME;
1330  }
1331  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1332  }
1334 
1335  return 0;
1336 }
1337 
1338 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1339  const char *url, const char *headers)
1340 {
1341  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1342 }
1343 
1344 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1345  const char *headers, RTSPMessageHeader *reply,
1346  unsigned char **content_ptr)
1347 {
1348  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1349  content_ptr, NULL, 0);
1350 }
1351 
1353  const char *method, const char *url,
1354  const char *header,
1355  RTSPMessageHeader *reply,
1356  unsigned char **content_ptr,
1357  const unsigned char *send_content,
1358  int send_content_length)
1359 {
1360  RTSPState *rt = s->priv_data;
1361  HTTPAuthType cur_auth_type;
1362  int ret, attempts = 0;
1363 
1364 retry:
1365  cur_auth_type = rt->auth_state.auth_type;
1366  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1367  send_content,
1368  send_content_length)))
1369  return ret;
1370 
1371  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1372  return ret;
1373  attempts++;
1374 
1375  if (reply->status_code == 401 &&
1376  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1377  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1378  goto retry;
1379 
1380  if (reply->status_code > 400){
1381  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1382  method,
1383  reply->status_code,
1384  reply->reason);
1385  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1386  }
1387 
1388  return 0;
1389 }
1390 
1391 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1392  int lower_transport, const char *real_challenge)
1393 {
1394  RTSPState *rt = s->priv_data;
1395  int rtx = 0, j, i, err, interleave = 0, port_off;
1396  RTSPStream *rtsp_st;
1397  RTSPMessageHeader reply1, *reply = &reply1;
1398  char cmd[2048];
1399  const char *trans_pref;
1400 
1401  if (rt->transport == RTSP_TRANSPORT_RDT)
1402  trans_pref = "x-pn-tng";
1403  else if (rt->transport == RTSP_TRANSPORT_RAW)
1404  trans_pref = "RAW/RAW";
1405  else
1406  trans_pref = "RTP/AVP";
1407 
1408  /* default timeout: 1 minute */
1409  rt->timeout = 60;
1410 
1411  /* Choose a random starting offset within the first half of the
1412  * port range, to allow for a number of ports to try even if the offset
1413  * happens to be at the end of the random range. */
1414  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1415  /* even random offset */
1416  port_off -= port_off & 0x01;
1417 
1418  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1419  char transport[2048];
1420 
1421  /*
1422  * WMS serves all UDP data over a single connection, the RTX, which
1423  * isn't necessarily the first in the SDP but has to be the first
1424  * to be set up, else the second/third SETUP will fail with a 461.
1425  */
1426  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1427  rt->server_type == RTSP_SERVER_WMS) {
1428  if (i == 0) {
1429  /* rtx first */
1430  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1431  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1432  if (len >= 4 &&
1433  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1434  "/rtx"))
1435  break;
1436  }
1437  if (rtx == rt->nb_rtsp_streams)
1438  return -1; /* no RTX found */
1439  rtsp_st = rt->rtsp_streams[rtx];
1440  } else
1441  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1442  } else
1443  rtsp_st = rt->rtsp_streams[i];
1444 
1445  /* RTP/UDP */
1446  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1447  char buf[256];
1448 
1449  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1450  port = reply->transports[0].client_port_min;
1451  goto have_port;
1452  }
1453 
1454  /* first try in specified port range */
1455  while (j <= rt->rtp_port_max) {
1456  AVDictionary *opts = map_to_opts(rt);
1457 
1458  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1459  "?localport=%d", j);
1460  /* we will use two ports per rtp stream (rtp and rtcp) */
1461  j += 2;
1462  err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1463  &s->interrupt_callback, &opts);
1464 
1465  av_dict_free(&opts);
1466 
1467  if (!err)
1468  goto rtp_opened;
1469  }
1470  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1471  err = AVERROR(EIO);
1472  goto fail;
1473 
1474  rtp_opened:
1475  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1476  have_port:
1477  snprintf(transport, sizeof(transport) - 1,
1478  "%s/UDP;", trans_pref);
1479  if (rt->server_type != RTSP_SERVER_REAL)
1480  av_strlcat(transport, "unicast;", sizeof(transport));
1481  av_strlcatf(transport, sizeof(transport),
1482  "client_port=%d", port);
1483  if (rt->transport == RTSP_TRANSPORT_RTP &&
1484  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1485  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1486  }
1487 
1488  /* RTP/TCP */
1489  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1490  /* For WMS streams, the application streams are only used for
1491  * UDP. When trying to set it up for TCP streams, the server
1492  * will return an error. Therefore, we skip those streams. */
1493  if (rt->server_type == RTSP_SERVER_WMS &&
1494  (rtsp_st->stream_index < 0 ||
1495  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1497  continue;
1498  snprintf(transport, sizeof(transport) - 1,
1499  "%s/TCP;", trans_pref);
1500  if (rt->transport != RTSP_TRANSPORT_RDT)
1501  av_strlcat(transport, "unicast;", sizeof(transport));
1502  av_strlcatf(transport, sizeof(transport),
1503  "interleaved=%d-%d",
1504  interleave, interleave + 1);
1505  interleave += 2;
1506  }
1507 
1508  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1509  snprintf(transport, sizeof(transport) - 1,
1510  "%s/UDP;multicast", trans_pref);
1511  }
1512  if (s->oformat) {
1513  av_strlcat(transport, ";mode=record", sizeof(transport));
1514  } else if (rt->server_type == RTSP_SERVER_REAL ||
1516  av_strlcat(transport, ";mode=play", sizeof(transport));
1517  snprintf(cmd, sizeof(cmd),
1518  "Transport: %s\r\n",
1519  transport);
1520  if (rt->accept_dynamic_rate)
1521  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1522  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1523  char real_res[41], real_csum[9];
1524  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1525  real_challenge);
1526  av_strlcatf(cmd, sizeof(cmd),
1527  "If-Match: %s\r\n"
1528  "RealChallenge2: %s, sd=%s\r\n",
1529  rt->session_id, real_res, real_csum);
1530  }
1531  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1532  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1533  err = 1;
1534  goto fail;
1535  } else if (reply->status_code != RTSP_STATUS_OK ||
1536  reply->nb_transports != 1) {
1538  goto fail;
1539  }
1540 
1541  /* XXX: same protocol for all streams is required */
1542  if (i > 0) {
1543  if (reply->transports[0].lower_transport != rt->lower_transport ||
1544  reply->transports[0].transport != rt->transport) {
1545  err = AVERROR_INVALIDDATA;
1546  goto fail;
1547  }
1548  } else {
1549  rt->lower_transport = reply->transports[0].lower_transport;
1550  rt->transport = reply->transports[0].transport;
1551  }
1552 
1553  /* Fail if the server responded with another lower transport mode
1554  * than what we requested. */
1555  if (reply->transports[0].lower_transport != lower_transport) {
1556  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1557  err = AVERROR_INVALIDDATA;
1558  goto fail;
1559  }
1560 
1561  switch(reply->transports[0].lower_transport) {
1563  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1564  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1565  break;
1566 
1567  case RTSP_LOWER_TRANSPORT_UDP: {
1568  char url[1024], options[30] = "";
1569  const char *peer = host;
1570 
1571  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1572  av_strlcpy(options, "?connect=1", sizeof(options));
1573  /* Use source address if specified */
1574  if (reply->transports[0].source[0])
1575  peer = reply->transports[0].source;
1576  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1577  reply->transports[0].server_port_min, "%s", options);
1578  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1579  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1580  err = AVERROR_INVALIDDATA;
1581  goto fail;
1582  }
1583  break;
1584  }
1586  char url[1024], namebuf[50], optbuf[20] = "";
1587  struct sockaddr_storage addr;
1588  int port, ttl;
1589 
1590  if (reply->transports[0].destination.ss_family) {
1591  addr = reply->transports[0].destination;
1592  port = reply->transports[0].port_min;
1593  ttl = reply->transports[0].ttl;
1594  } else {
1595  addr = rtsp_st->sdp_ip;
1596  port = rtsp_st->sdp_port;
1597  ttl = rtsp_st->sdp_ttl;
1598  }
1599  if (ttl > 0)
1600  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1601  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1602  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1603  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1604  port, "%s", optbuf);
1605  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1606  &s->interrupt_callback, NULL) < 0) {
1607  err = AVERROR_INVALIDDATA;
1608  goto fail;
1609  }
1610  break;
1611  }
1612  }
1613 
1614  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1615  goto fail;
1616  }
1617 
1618  if (rt->nb_rtsp_streams && reply->timeout > 0)
1619  rt->timeout = reply->timeout;
1620 
1621  if (rt->server_type == RTSP_SERVER_REAL)
1622  rt->need_subscription = 1;
1623 
1624  return 0;
1625 
1626 fail:
1627  ff_rtsp_undo_setup(s, 0);
1628  return err;
1629 }
1630 
1632 {
1633  RTSPState *rt = s->priv_data;
1634  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1635  ffurl_close(rt->rtsp_hd);
1636  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1637 }
1638 
1640 {
1641  RTSPState *rt = s->priv_data;
1642  char proto[128], host[1024], path[1024];
1643  char tcpname[1024], cmd[2048], auth[128];
1644  const char *lower_rtsp_proto = "tcp";
1645  int port, err, tcp_fd;
1646  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1647  int lower_transport_mask = 0;
1648  int default_port = RTSP_DEFAULT_PORT;
1649  char real_challenge[64] = "";
1650  struct sockaddr_storage peer;
1651  socklen_t peer_len = sizeof(peer);
1652 
1653  if (rt->rtp_port_max < rt->rtp_port_min) {
1654  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1655  "than min port %d\n", rt->rtp_port_max,
1656  rt->rtp_port_min);
1657  return AVERROR(EINVAL);
1658  }
1659 
1660  if (!ff_network_init())
1661  return AVERROR(EIO);
1662 
1663  if (s->max_delay < 0) /* Not set by the caller */
1665 
1670  }
1671  /* Only pass through valid flags from here */
1673 
1674 redirect:
1675  /* extract hostname and port */
1676  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1677  host, sizeof(host), &port, path, sizeof(path), s->filename);
1678 
1679  if (!strcmp(proto, "rtsps")) {
1680  lower_rtsp_proto = "tls";
1681  default_port = RTSPS_DEFAULT_PORT;
1683  }
1684 
1685  if (*auth) {
1686  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1687  }
1688  if (port < 0)
1689  port = default_port;
1690 
1691  lower_transport_mask = rt->lower_transport_mask;
1692 
1693  if (!lower_transport_mask)
1694  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1695 
1696  if (s->oformat) {
1697  /* Only UDP or TCP - UDP multicast isn't supported. */
1698  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1699  (1 << RTSP_LOWER_TRANSPORT_TCP);
1700  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1701  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1702  "only UDP and TCP are supported for output.\n");
1703  err = AVERROR(EINVAL);
1704  goto fail;
1705  }
1706  }
1707 
1708  /* Construct the URI used in request; this is similar to s->filename,
1709  * but with authentication credentials removed and RTSP specific options
1710  * stripped out. */
1711  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1712  host, port, "%s", path);
1713 
1714  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1715  /* set up initial handshake for tunneling */
1716  char httpname[1024];
1717  char sessioncookie[17];
1718  char headers[1024];
1719 
1720  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1721  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1723 
1724  /* GET requests */
1725  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1726  &s->interrupt_callback) < 0) {
1727  err = AVERROR(EIO);
1728  goto fail;
1729  }
1730 
1731  /* generate GET headers */
1732  snprintf(headers, sizeof(headers),
1733  "x-sessioncookie: %s\r\n"
1734  "Accept: application/x-rtsp-tunnelled\r\n"
1735  "Pragma: no-cache\r\n"
1736  "Cache-Control: no-cache\r\n",
1737  sessioncookie);
1738  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1739 
1740  /* complete the connection */
1741  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1742  err = AVERROR(EIO);
1743  goto fail;
1744  }
1745 
1746  /* POST requests */
1747  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1748  &s->interrupt_callback) < 0 ) {
1749  err = AVERROR(EIO);
1750  goto fail;
1751  }
1752 
1753  /* generate POST headers */
1754  snprintf(headers, sizeof(headers),
1755  "x-sessioncookie: %s\r\n"
1756  "Content-Type: application/x-rtsp-tunnelled\r\n"
1757  "Pragma: no-cache\r\n"
1758  "Cache-Control: no-cache\r\n"
1759  "Content-Length: 32767\r\n"
1760  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1761  sessioncookie);
1762  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1763  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1764 
1765  /* Initialize the authentication state for the POST session. The HTTP
1766  * protocol implementation doesn't properly handle multi-pass
1767  * authentication for POST requests, since it would require one of
1768  * the following:
1769  * - implementing Expect: 100-continue, which many HTTP servers
1770  * don't support anyway, even less the RTSP servers that do HTTP
1771  * tunneling
1772  * - sending the whole POST data until getting a 401 reply specifying
1773  * what authentication method to use, then resending all that data
1774  * - waiting for potential 401 replies directly after sending the
1775  * POST header (waiting for some unspecified time)
1776  * Therefore, we copy the full auth state, which works for both basic
1777  * and digest. (For digest, we would have to synchronize the nonce
1778  * count variable between the two sessions, if we'd do more requests
1779  * with the original session, though.)
1780  */
1782 
1783  /* complete the connection */
1784  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1785  err = AVERROR(EIO);
1786  goto fail;
1787  }
1788  } else {
1789  int ret;
1790  /* open the tcp connection */
1791  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1792  host, port,
1793  "?timeout=%d", rt->stimeout);
1794  if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1795  &s->interrupt_callback, NULL)) < 0) {
1796  err = ret;
1797  goto fail;
1798  }
1799  rt->rtsp_hd_out = rt->rtsp_hd;
1800  }
1801  rt->seq = 0;
1802 
1803  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1804  if (tcp_fd < 0) {
1805  err = tcp_fd;
1806  goto fail;
1807  }
1808  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1809  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1810  NULL, 0, NI_NUMERICHOST);
1811  }
1812 
1813  /* request options supported by the server; this also detects server
1814  * type */
1815  for (rt->server_type = RTSP_SERVER_RTP;;) {
1816  cmd[0] = 0;
1817  if (rt->server_type == RTSP_SERVER_REAL)
1818  av_strlcat(cmd,
1819  /*
1820  * The following entries are required for proper
1821  * streaming from a Realmedia server. They are
1822  * interdependent in some way although we currently
1823  * don't quite understand how. Values were copied
1824  * from mplayer SVN r23589.
1825  * ClientChallenge is a 16-byte ID in hex
1826  * CompanyID is a 16-byte ID in base64
1827  */
1828  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1829  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1830  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1831  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1832  sizeof(cmd));
1833  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1834  if (reply->status_code != RTSP_STATUS_OK) {
1836  goto fail;
1837  }
1838 
1839  /* detect server type if not standard-compliant RTP */
1840  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1842  continue;
1843  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1845  } else if (rt->server_type == RTSP_SERVER_REAL)
1846  strcpy(real_challenge, reply->real_challenge);
1847  break;
1848  }
1849 
1850  if (CONFIG_RTSP_DEMUXER && s->iformat)
1851  err = ff_rtsp_setup_input_streams(s, reply);
1852  else if (CONFIG_RTSP_MUXER)
1853  err = ff_rtsp_setup_output_streams(s, host);
1854  else
1855  av_assert0(0);
1856  if (err)
1857  goto fail;
1858 
1859  do {
1860  int lower_transport = ff_log2_tab[lower_transport_mask &
1861  ~(lower_transport_mask - 1)];
1862 
1863  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1864  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1865  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1866 
1867  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1868  rt->server_type == RTSP_SERVER_REAL ?
1869  real_challenge : NULL);
1870  if (err < 0)
1871  goto fail;
1872  lower_transport_mask &= ~(1 << lower_transport);
1873  if (lower_transport_mask == 0 && err == 1) {
1874  err = AVERROR(EPROTONOSUPPORT);
1875  goto fail;
1876  }
1877  } while (err);
1878 
1879  rt->lower_transport_mask = lower_transport_mask;
1880  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1881  rt->state = RTSP_STATE_IDLE;
1882  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1883  return 0;
1884  fail:
1887  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1888  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1889  rt->session_id[0] = '\0';
1890  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1891  reply->status_code,
1892  s->filename);
1893  goto redirect;
1894  }
1895  ff_network_close();
1896  return err;
1897 }
1898 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1899 
1900 #if CONFIG_RTPDEC
1901 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1902  uint8_t *buf, int buf_size, int64_t wait_end)
1903 {
1904  RTSPState *rt = s->priv_data;
1905  RTSPStream *rtsp_st;
1906  int n, i, ret, tcp_fd, timeout_cnt = 0;
1907  int max_p = 0;
1908  struct pollfd *p = rt->p;
1909  int *fds = NULL, fdsnum, fdsidx;
1910 
1911  for (;;) {
1913  return AVERROR_EXIT;
1914  if (wait_end && wait_end - av_gettime_relative() < 0)
1915  return AVERROR(EAGAIN);
1916  max_p = 0;
1917  if (rt->rtsp_hd) {
1918  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1919  p[max_p].fd = tcp_fd;
1920  p[max_p++].events = POLLIN;
1921  } else {
1922  tcp_fd = -1;
1923  }
1924  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1925  rtsp_st = rt->rtsp_streams[i];
1926  if (rtsp_st->rtp_handle) {
1927  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1928  &fds, &fdsnum)) {
1929  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1930  return ret;
1931  }
1932  if (fdsnum != 2) {
1933  av_log(s, AV_LOG_ERROR,
1934  "Number of fds %d not supported\n", fdsnum);
1935  return AVERROR_INVALIDDATA;
1936  }
1937  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1938  p[max_p].fd = fds[fdsidx];
1939  p[max_p++].events = POLLIN;
1940  }
1941  av_freep(&fds);
1942  }
1943  }
1944  n = poll(p, max_p, POLL_TIMEOUT_MS);
1945  if (n > 0) {
1946  int j = 1 - (tcp_fd == -1);
1947  timeout_cnt = 0;
1948  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1949  rtsp_st = rt->rtsp_streams[i];
1950  if (rtsp_st->rtp_handle) {
1951  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1952  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1953  if (ret > 0) {
1954  *prtsp_st = rtsp_st;
1955  return ret;
1956  }
1957  }
1958  j+=2;
1959  }
1960  }
1961 #if CONFIG_RTSP_DEMUXER
1962  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1963  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1964  if (rt->state == RTSP_STATE_STREAMING) {
1966  return AVERROR_EOF;
1967  else
1969  "Unable to answer to TEARDOWN\n");
1970  } else
1971  return 0;
1972  } else {
1973  RTSPMessageHeader reply;
1974  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1975  if (ret < 0)
1976  return ret;
1977  /* XXX: parse message */
1978  if (rt->state != RTSP_STATE_STREAMING)
1979  return 0;
1980  }
1981  }
1982 #endif
1983  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1984  return AVERROR(ETIMEDOUT);
1985  } else if (n < 0 && errno != EINTR)
1986  return AVERROR(errno);
1987  }
1988 }
1989 
1990 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1991  const uint8_t *buf, int len)
1992 {
1993  RTSPState *rt = s->priv_data;
1994  int i;
1995  if (len < 0)
1996  return len;
1997  if (rt->nb_rtsp_streams == 1) {
1998  *rtsp_st = rt->rtsp_streams[0];
1999  return len;
2000  }
2001  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2002  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2003  int no_ssrc = 0;
2004  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2005  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2006  if (!rtpctx)
2007  continue;
2008  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2009  *rtsp_st = rt->rtsp_streams[i];
2010  return len;
2011  }
2012  if (!rtpctx->ssrc)
2013  no_ssrc = 1;
2014  }
2015  if (no_ssrc) {
2017  "Unable to pick stream for packet - SSRC not known for "
2018  "all streams\n");
2019  return AVERROR(EAGAIN);
2020  }
2021  } else {
2022  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2023  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2024  *rtsp_st = rt->rtsp_streams[i];
2025  return len;
2026  }
2027  }
2028  }
2029  }
2030  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2031  return AVERROR(EAGAIN);
2032 }
2033 
2035 {
2036  RTSPState *rt = s->priv_data;
2037  int ret, len;
2038  RTSPStream *rtsp_st, *first_queue_st = NULL;
2039  int64_t wait_end = 0;
2040 
2041  if (rt->nb_byes == rt->nb_rtsp_streams)
2042  return AVERROR_EOF;
2043 
2044  /* get next frames from the same RTP packet */
2045  if (rt->cur_transport_priv) {
2046  if (rt->transport == RTSP_TRANSPORT_RDT) {
2047  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2048  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2049  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2050  } else if (CONFIG_RTPDEC && rt->ts) {
2051  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2052  if (ret >= 0) {
2053  rt->recvbuf_pos += ret;
2054  ret = rt->recvbuf_pos < rt->recvbuf_len;
2055  }
2056  } else
2057  ret = -1;
2058  if (ret == 0) {
2059  rt->cur_transport_priv = NULL;
2060  return 0;
2061  } else if (ret == 1) {
2062  return 0;
2063  } else
2064  rt->cur_transport_priv = NULL;
2065  }
2066 
2067 redo:
2068  if (rt->transport == RTSP_TRANSPORT_RTP) {
2069  int i;
2070  int64_t first_queue_time = 0;
2071  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2072  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2073  int64_t queue_time;
2074  if (!rtpctx)
2075  continue;
2076  queue_time = ff_rtp_queued_packet_time(rtpctx);
2077  if (queue_time && (queue_time - first_queue_time < 0 ||
2078  !first_queue_time)) {
2079  first_queue_time = queue_time;
2080  first_queue_st = rt->rtsp_streams[i];
2081  }
2082  }
2083  if (first_queue_time) {
2084  wait_end = first_queue_time + s->max_delay;
2085  } else {
2086  wait_end = 0;
2087  first_queue_st = NULL;
2088  }
2089  }
2090 
2091  /* read next RTP packet */
2092  if (!rt->recvbuf) {
2094  if (!rt->recvbuf)
2095  return AVERROR(ENOMEM);
2096  }
2097 
2098  switch(rt->lower_transport) {
2099  default:
2100 #if CONFIG_RTSP_DEMUXER
2102  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2103  break;
2104 #endif
2107  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2108  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2109  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2110  break;
2112  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2113  wait_end && wait_end < av_gettime_relative())
2114  len = AVERROR(EAGAIN);
2115  else
2116  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2117  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2118  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2120  break;
2121  }
2122  if (len == AVERROR(EAGAIN) && first_queue_st &&
2123  rt->transport == RTSP_TRANSPORT_RTP) {
2124  rtsp_st = first_queue_st;
2125  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2126  goto end;
2127  }
2128  if (len < 0)
2129  return len;
2130  if (len == 0)
2131  return AVERROR_EOF;
2132  if (rt->transport == RTSP_TRANSPORT_RDT) {
2133  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2134  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2135  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2136  if (rtsp_st->feedback) {
2137  AVIOContext *pb = NULL;
2139  pb = s->pb;
2140  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2141  }
2142  if (ret < 0) {
2143  /* Either bad packet, or a RTCP packet. Check if the
2144  * first_rtcp_ntp_time field was initialized. */
2145  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2146  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2147  /* first_rtcp_ntp_time has been initialized for this stream,
2148  * copy the same value to all other uninitialized streams,
2149  * in order to map their timestamp origin to the same ntp time
2150  * as this one. */
2151  int i;
2152  AVStream *st = NULL;
2153  if (rtsp_st->stream_index >= 0)
2154  st = s->streams[rtsp_st->stream_index];
2155  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2156  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2157  AVStream *st2 = NULL;
2158  if (rt->rtsp_streams[i]->stream_index >= 0)
2159  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2160  if (rtpctx2 && st && st2 &&
2161  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2162  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2163  rtpctx2->rtcp_ts_offset = av_rescale_q(
2164  rtpctx->rtcp_ts_offset, st->time_base,
2165  st2->time_base);
2166  }
2167  }
2168  // Make real NTP start time available in AVFormatContext
2169  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2170  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2171  if (rtpctx->st) {
2172  s->start_time_realtime -=
2173  av_rescale (rtpctx->rtcp_ts_offset,
2174  (uint64_t) rtpctx->st->time_base.num * 1000000,
2175  rtpctx->st->time_base.den);
2176  }
2177  }
2178  }
2179  if (ret == -RTCP_BYE) {
2180  rt->nb_byes++;
2181 
2182  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2183  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2184 
2185  if (rt->nb_byes == rt->nb_rtsp_streams)
2186  return AVERROR_EOF;
2187  }
2188  }
2189  } else if (CONFIG_RTPDEC && rt->ts) {
2190  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2191  if (ret >= 0) {
2192  if (ret < len) {
2193  rt->recvbuf_len = len;
2194  rt->recvbuf_pos = ret;
2195  rt->cur_transport_priv = rt->ts;
2196  return 1;
2197  } else {
2198  ret = 0;
2199  }
2200  }
2201  } else {
2202  return AVERROR_INVALIDDATA;
2203  }
2204 end:
2205  if (ret < 0)
2206  goto redo;
2207  if (ret == 1)
2208  /* more packets may follow, so we save the RTP context */
2209  rt->cur_transport_priv = rtsp_st->transport_priv;
2210 
2211  return ret;
2212 }
2213 #endif /* CONFIG_RTPDEC */
2214 
2215 #if CONFIG_SDP_DEMUXER
2216 static int sdp_probe(AVProbeData *p1)
2217 {
2218  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2219 
2220  /* we look for a line beginning "c=IN IP" */
2221  while (p < p_end && *p != '\0') {
2222  if (sizeof("c=IN IP") - 1 < p_end - p &&
2223  av_strstart(p, "c=IN IP", NULL))
2224  return AVPROBE_SCORE_EXTENSION;
2225 
2226  while (p < p_end - 1 && *p != '\n') p++;
2227  if (++p >= p_end)
2228  break;
2229  if (*p == '\r')
2230  p++;
2231  }
2232  return 0;
2233 }
2234 
2235 static void append_source_addrs(char *buf, int size, const char *name,
2236  int count, struct RTSPSource **addrs)
2237 {
2238  int i;
2239  if (!count)
2240  return;
2241  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2242  for (i = 1; i < count; i++)
2243  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2244 }
2245 
2246 static int sdp_read_header(AVFormatContext *s)
2247 {
2248  RTSPState *rt = s->priv_data;
2249  RTSPStream *rtsp_st;
2250  int size, i, err;
2251  char *content;
2252  char url[1024];
2253 
2254  if (!ff_network_init())
2255  return AVERROR(EIO);
2256 
2257  if (s->max_delay < 0) /* Not set by the caller */
2259  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2261 
2262  /* read the whole sdp file */
2263  /* XXX: better loading */
2264  content = av_malloc(SDP_MAX_SIZE);
2265  if (!content)
2266  return AVERROR(ENOMEM);
2267  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2268  if (size <= 0) {
2269  av_free(content);
2270  return AVERROR_INVALIDDATA;
2271  }
2272  content[size] ='\0';
2273 
2274  err = ff_sdp_parse(s, content);
2275  av_freep(&content);
2276  if (err) goto fail;
2277 
2278  /* open each RTP stream */
2279  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2280  char namebuf[50];
2281  rtsp_st = rt->rtsp_streams[i];
2282 
2283  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2284  AVDictionary *opts = map_to_opts(rt);
2285 
2286  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2287  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2288  ff_url_join(url, sizeof(url), "rtp", NULL,
2289  namebuf, rtsp_st->sdp_port,
2290  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2291  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2292  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2293  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2294 
2295  append_source_addrs(url, sizeof(url), "sources",
2296  rtsp_st->nb_include_source_addrs,
2297  rtsp_st->include_source_addrs);
2298  append_source_addrs(url, sizeof(url), "block",
2299  rtsp_st->nb_exclude_source_addrs,
2300  rtsp_st->exclude_source_addrs);
2301  err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2302  &s->interrupt_callback, &opts);
2303 
2304  av_dict_free(&opts);
2305 
2306  if (err < 0) {
2307  err = AVERROR_INVALIDDATA;
2308  goto fail;
2309  }
2310  }
2311  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2312  goto fail;
2313  }
2314  return 0;
2315 fail:
2317  ff_network_close();
2318  return err;
2319 }
2320 
2321 static int sdp_read_close(AVFormatContext *s)
2322 {
2324  ff_network_close();
2325  return 0;
2326 }
2327 
2328 static const AVClass sdp_demuxer_class = {
2329  .class_name = "SDP demuxer",
2330  .item_name = av_default_item_name,
2331  .option = sdp_options,
2332  .version = LIBAVUTIL_VERSION_INT,
2333 };
2334 
2335 AVInputFormat ff_sdp_demuxer = {
2336  .name = "sdp",
2337  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2338  .priv_data_size = sizeof(RTSPState),
2339  .read_probe = sdp_probe,
2340  .read_header = sdp_read_header,
2342  .read_close = sdp_read_close,
2343  .priv_class = &sdp_demuxer_class,
2344 };
2345 #endif /* CONFIG_SDP_DEMUXER */
2346 
2347 #if CONFIG_RTP_DEMUXER
2348 static int rtp_probe(AVProbeData *p)
2349 {
2350  if (av_strstart(p->filename, "rtp:", NULL))
2351  return AVPROBE_SCORE_MAX;
2352  return 0;
2353 }
2354 
2355 static int rtp_read_header(AVFormatContext *s)
2356 {
2357  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2358  char host[500], sdp[500];
2359  int ret, port;
2360  URLContext* in = NULL;
2361  int payload_type;
2362  AVCodecContext codec = { 0 };
2363  struct sockaddr_storage addr;
2364  AVIOContext pb;
2365  socklen_t addrlen = sizeof(addr);
2366  RTSPState *rt = s->priv_data;
2367 
2368  if (!ff_network_init())
2369  return AVERROR(EIO);
2370 
2371  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2372  &s->interrupt_callback, NULL);
2373  if (ret)
2374  goto fail;
2375 
2376  while (1) {
2377  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2378  if (ret == AVERROR(EAGAIN))
2379  continue;
2380  if (ret < 0)
2381  goto fail;
2382  if (ret < 12) {
2383  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2384  continue;
2385  }
2386 
2387  if ((recvbuf[0] & 0xc0) != 0x80) {
2388  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2389  "received\n");
2390  continue;
2391  }
2392 
2393  if (RTP_PT_IS_RTCP(recvbuf[1]))
2394  continue;
2395 
2396  payload_type = recvbuf[1] & 0x7f;
2397  break;
2398  }
2399  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2400  ffurl_close(in);
2401  in = NULL;
2402 
2403  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2404  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2405  "without an SDP file describing it\n",
2406  payload_type);
2407  goto fail;
2408  }
2409  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2410  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2411  "properly you need an SDP file "
2412  "describing it\n");
2413  }
2414 
2415  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2416  NULL, 0, s->filename);
2417 
2418  snprintf(sdp, sizeof(sdp),
2419  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2420  addr.ss_family == AF_INET ? 4 : 6, host,
2421  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2422  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2423  port, payload_type);
2424  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2425 
2426  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2427  s->pb = &pb;
2428 
2429  /* sdp_read_header initializes this again */
2430  ff_network_close();
2431 
2432  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2433 
2434  ret = sdp_read_header(s);
2435  s->pb = NULL;
2436  return ret;
2437 
2438 fail:
2439  if (in)
2440  ffurl_close(in);
2441  ff_network_close();
2442  return ret;
2443 }
2444 
2445 static const AVClass rtp_demuxer_class = {
2446  .class_name = "RTP demuxer",
2447  .item_name = av_default_item_name,
2448  .option = rtp_options,
2449  .version = LIBAVUTIL_VERSION_INT,
2450 };
2451 
2452 AVInputFormat ff_rtp_demuxer = {
2453  .name = "rtp",
2454  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2455  .priv_data_size = sizeof(RTSPState),
2456  .read_probe = rtp_probe,
2457  .read_header = rtp_read_header,
2459  .read_close = sdp_read_close,
2460  .flags = AVFMT_NOFILE,
2461  .priv_class = &rtp_demuxer_class,
2462 };
2463 #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:3973
char crypto_suite[40]
Definition: rtsp.h:472
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:387
#define NULL
Definition: coverity.c:32
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:100
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:548
float v
const char * s
Definition: avisynth_c.h:631
Bytestream IO Context.
Definition: avio.h:111
Realmedia Data Transport.
Definition: rtsp.h:58
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:567
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
Definition: avformat.h:1510
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1535
AVOption.
Definition: opt.h:255
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:115
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:187
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:62
const char * filename
Definition: avformat.h:461
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
Definition: rtsp.c:165
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:317
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4093
int av_parse_time(int64_t *timeval, const char *timestr, int duration)
Parse timestr and return in *time a corresponding number of microseconds.
Definition: parseutils.c:558
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:374
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
#define CONFIG_RTPDEC
Definition: config.h:594
Windows Media server.
Definition: rtsp.h:209
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:354
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:785
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2789
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:202
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:131
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
Definition: rdt.c:335
int num
numerator
Definition: rational.h:44
int index
stream index in AVFormatContext
Definition: avformat.h:855
#define AVIO_FLAG_READ
read-only
Definition: avio.h:485
char * user_agent
User-Agent string.
Definition: rtsp.h:407
char location[4096]
the "Location:" field.
Definition: rtsp.h:152
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:486
int mode_record
transport set to record data
Definition: rtsp.h:112
enum AVMediaType codec_type
Definition: rtp.c:37
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:223
void ff_network_close(void)
Definition: network.c:102
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:144
initialized and sending/receiving data
Definition: rtsp.h:197
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:270
static av_always_inline void interleave(IDWTELEM *dst, IDWTELEM *src0, IDWTELEM *src1, int w2, int add, int shift)
Definition: dirac_dwt.c:40
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:420
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:79
#define freeaddrinfo
Definition: network.h:208
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:452
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1334
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:418
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:245
int auth_type
The currently chosen auth type.
Definition: httpauth.h:59
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:239
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:132
AVCodec.
Definition: avcodec.h:3482
#define AI_NUMERICHOST
Definition: network.h:177
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:121
This describes the server response to each RTSP command.
Definition: rtsp.h:127
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
Definition: rtpdec.c:509
#define RECVBUF_SIZE
Definition: rtsp.c:59
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:142
Format I/O context.
Definition: avformat.h:1285
#define RTP_PT_PRIVATE
Definition: rtp.h:77
#define COMMON_OPTS()
Definition: rtsp.c:77
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
Definition: rtp.c:143
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:207
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:402
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
#define RTSP_FLAG_PREFER_TCP
Try RTP via TCP first if possible.
Definition: rtsp.h:423
int recvbuf_len
Definition: rtsp.h:323
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:180
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Public dictionary API.
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:45
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:359
#define CONFIG_RTSP_DEMUXER
Definition: config.h:1213
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:148
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:74
#define av_malloc(s)
Opaque data information usually continuous.
Definition: avutil.h:195
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:109
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
Definition: rtpdec.h:126
int ff_network_init(void)
Definition: network.c:55
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1244
AVOptions.
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:470
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:106
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int id
Format-specific stream ID.
Definition: avformat.h:861
enum AVStreamParseType need_parsing
Definition: avformat.h:1046
#define POLL_TIMEOUT_MS
Definition: rtsp.c:55
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:60
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
static void handler(vbi_event *ev, void *user_data)
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:3756
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1353
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:372
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:328
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:435
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
#define CONFIG_RTSP_MUXER
Definition: config.h:1870
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:131
#define AVERROR_EOF
End of file.
Definition: error.h:55
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:156
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
ptrdiff_t size
Definition: opengl_enc.c:101
static const uint8_t header[24]
Definition: sdr2.c:67
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:464
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
Normal RTSP.
Definition: rtsp.h:68
const OptionDef options[]
Definition: ffserver.c:3810
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
#define av_log(a,...)
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:134
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:542
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1304
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:177
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:77
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:147
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:455
Private data for the RTSP demuxer.
Definition: rtsp.h:218
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:255
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
Definition: avio.c:277
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVDictionary * metadata
Metadata that applies to the whole file.
Definition: avformat.h:1497
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:585
int profile
Definition: mxfenc.c:1820
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
av_default_item_name
#define AV_RB16
Definition: intreadwrite.h:53
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:178
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const AVOption ff_rtsp_options[]
Definition: rtsp.c:82
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:199
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:182
Definition: graph2dot.c:48
URLContext * rtsp_hd
Definition: rtsp.h:220
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
Definition: avcodec.h:3489
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:331
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:453
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:610
GLsizei count
Definition: opengl_enc.c:109
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
Definition: base64.c:138
int64_t rtcp_ts_offset
Definition: rtpdec.h:182
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:83
#define fail()
Definition: checkasm.h:57
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:131
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:225
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:164
int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:71
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:440
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:873
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:463
int seq
RTSP command sequence number.
Definition: rtsp.h:241
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:462
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:339
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1341
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:419
#define NI_NUMERICHOST
Definition: network.h:185
#define th
Definition: regdef.h:75
#define LIBAVFORMAT_IDENT
Definition: version.h:44
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:307
int recvbuf_pos
Definition: rtsp.h:322
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:119
char filename[1024]
input or output filename
Definition: avformat.h:1361
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:223
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:134
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes to a null-terminated string.
Definition: base64.h:61
#define FFMIN(a, b)
Definition: common.h:92
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:283
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:55
int content_length
length of the data following this header
Definition: rtsp.h:129
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: [;ti...
Definition: rtsp.h:172
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:75
enum AVStreamParseType need_parsing
Definition: rtpdec.h:119
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:88
RTSP over HTTP (tunneling)
Definition: rtsp.h:69
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:130
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:140
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:156
int n
Definition: avisynth_c.h:547
AVDictionary * metadata
Definition: avformat.h:928
char crypto_params[100]
Definition: rtsp.h:473
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:192
int(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
Definition: rtpdec.h:128
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:578
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define ENC
Definition: rtsp.c:64
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:450
Raw data (over UDP)
Definition: rtsp.h:59
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:321
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:23
int sdp_payload_type
payload type
Definition: rtsp.h:457
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:541
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:454
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1178
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:187
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:634
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:441
Stream structure.
Definition: avformat.h:854
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
Definition: url.c:36
int nb_byes
Definition: rtsp.h:336
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:262
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:426
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:451
enum AVMediaType codec_type
Definition: avcodec.h:1520
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:715
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
Definition: avio.c:607
enum AVCodecID codec_id
Definition: avcodec.h:1529
int rtp_port_max
Definition: rtsp.h:387
#define NTP_OFFSET
Definition: internal.h:159
Definition: rtp.h:100
int sample_rate
samples per second
Definition: avcodec.h:2272
AVIOContext * pb
I/O context.
Definition: avformat.h:1327
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:382
int server_port_max
Definition: rtsp.h:105
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
main external API structure.
Definition: avcodec.h:1512
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:3028
#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:2808
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:470
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:3693
int buffer_size
Definition: rtsp.h:410
This structure contains the data a format has to probe a file.
Definition: avformat.h:460
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:76
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
Definition: avstring.c:93
int interleaved_max
Definition: rtsp.h:93
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:267
static int flags
Definition: cpu.c:47
int ffurl_close(URLContext *h)
Definition: avio.c:419
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:138
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1370
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:231
#define DEC
Definition: rtsp.c:63
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:472
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:34
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:207
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:103
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
uint32_t ssrc
Definition: rtpdec.h:153
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:119
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:477
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:118
int ffio_init_context(AVIOContext *s, unsigned char *buffer, int buffer_size, int write_flag, void *opaque, int(*read_packet)(void *opaque, uint8_t *buf, int buf_size), int(*write_packet)(void *opaque, uint8_t *buf, int buf_size), int64_t(*seek)(void *opaque, int64_t offset, int whence))
Definition: aviobuf.c:72
int ffurl_open(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options)
Create an URLContext for accessing to the resource indicated by url, and open it. ...
Definition: avio.c:299
static double c[64]
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:288
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:463
int ffurl_read_complete(URLContext *h, unsigned char *buf, int size)
Read as many bytes as possible (up to size), calling the read function multiple times if necessary...
Definition: avio.c:367
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
Definition: rdt.c:94
int den
denominator
Definition: rational.h:45
char default_lang[4]
Definition: rtsp.h:409
struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:1297
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:3728
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
uint32_t base_timestamp
Definition: rtpdec.h:156
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:397
#define getnameinfo
Definition: network.h:209
#define av_free(p)
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:149
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:276
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:78
int channels
number of audio channels
Definition: avcodec.h:2273
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:446
void * priv_data
Format private data.
Definition: avformat.h:1313
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:593
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:843
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:456
#define MAX_TIMEOUTS
Definition: rtsp.c:57
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:986
int ai_flags
Definition: network.h:128
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1380
Realmedia-style server.
Definition: rtsp.h:208
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:344
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:553
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:640
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:116
#define av_malloc_array(a, b)
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:896
int interleaved_max
Definition: rtsp.h:444
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
Definition: rtpdec.c:830
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:114
int ff_rtp_set_remote_url(URLContext *h, const char *uri)
If no filename is given to av_open_input_file because you want to get the local port first...
Definition: rtpproto.c:98
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:2833
AVStream * st
Definition: rtpdec.h:151
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:444
This structure stores compressed data.
Definition: avcodec.h:1410
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
Definition: aviobuf.c:963
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:105
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:369
static const AVOption rtp_options[]
Definition: rtsp.c:112
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
Definition: avio.c:360
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:436
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:240
#define OFFSET(x)
Definition: rtsp.c:62
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:97
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:437
No authentication specified.
Definition: httpauth.h:29
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:101
const char * name
Definition: opengl_enc.c:103