FFmpeg
rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "config_components.h"
23 
24 #include "libavutil/avassert.h"
25 #include "libavutil/base64.h"
26 #include "libavutil/bprint.h"
27 #include "libavutil/avstring.h"
28 #include "libavutil/intreadwrite.h"
29 #include "libavutil/mathematics.h"
30 #include "libavutil/parseutils.h"
31 #include "libavutil/random_seed.h"
32 #include "libavutil/dict.h"
33 #include "libavutil/opt.h"
34 #include "libavutil/time.h"
35 #include "avformat.h"
36 #include "avio_internal.h"
37 
38 #if HAVE_POLL_H
39 #include <poll.h>
40 #endif
41 #include "internal.h"
42 #include "network.h"
43 #include "os_support.h"
44 #include "http.h"
45 #include "rtsp.h"
46 
47 #include "rtpdec.h"
48 #include "rtpproto.h"
49 #include "rdt.h"
50 #include "rtpdec_formats.h"
51 #include "rtpenc_chain.h"
52 #include "url.h"
53 #include "rtpenc.h"
54 #include "mpegts.h"
55 #include "version.h"
56 
57 /* Default timeout values for read packet in seconds */
58 #define READ_PACKET_TIMEOUT_S 10
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  { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = 1472 }, -1, INT_MAX, ENC } \
81 
82 
84  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
85  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
86  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
87  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
89  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
90  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
91  { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
92  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
93  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
94  { "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" },
95  { "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_SATIP_RAW}, 0, 0, DEC, "rtsp_flags" },
96  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
97  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
98  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
99  { "listen_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 },
100  { "timeout", "set timeout (in microseconds) of socket I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT64, {.i64 = 0}, INT_MIN, INT64_MAX, DEC },
101  COMMON_OPTS(),
102  { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
103  { NULL },
104 };
105 
106 static const AVOption sdp_options[] = {
107  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
108  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
109  { "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" },
110  { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(stimeout), AV_OPT_TYPE_DURATION, {.i64 = READ_PACKET_TIMEOUT_S*1000000}, INT_MIN, INT64_MAX, DEC },
111  { "localaddr", "local address", OFFSET(localaddr),AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, DEC }, \
112  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
113  COMMON_OPTS(),
114  { NULL },
115 };
116 
117 static const AVOption rtp_options[] = {
118  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
119  { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(stimeout), AV_OPT_TYPE_DURATION, {.i64 = READ_PACKET_TIMEOUT_S*1000000}, INT_MIN, INT64_MAX, DEC },
120  { "localaddr", "local address", OFFSET(localaddr),AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, DEC }, \
121  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
122  COMMON_OPTS(),
123  { NULL },
124 };
125 
126 
128 {
130 
131  av_dict_set_int(&opts, "buffer_size", rt->buffer_size, 0);
132  av_dict_set_int(&opts, "pkt_size", rt->pkt_size, 0);
133  if (rt->localaddr && rt->localaddr[0])
134  av_dict_set(&opts, "localaddr", rt->localaddr, 0);
135 
136  return opts;
137 }
138 
139 static void get_word_until_chars(char *buf, int buf_size,
140  const char *sep, const char **pp)
141 {
142  const char *p;
143  char *q;
144 
145  p = *pp;
146  p += strspn(p, SPACE_CHARS);
147  q = buf;
148  while (!strchr(sep, *p) && *p != '\0') {
149  if ((q - buf) < buf_size - 1)
150  *q++ = *p;
151  p++;
152  }
153  if (buf_size > 0)
154  *q = '\0';
155  *pp = p;
156 }
157 
158 static void get_word_sep(char *buf, int buf_size, const char *sep,
159  const char **pp)
160 {
161  if (**pp == '/') (*pp)++;
162  get_word_until_chars(buf, buf_size, sep, pp);
163 }
164 
165 static void get_word(char *buf, int buf_size, const char **pp)
166 {
167  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
168 }
169 
170 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
171  * and end time.
172  * Used for seeking in the rtp stream.
173  */
174 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
175 {
176  char buf[256];
177 
178  p += strspn(p, SPACE_CHARS);
179  if (!av_stristart(p, "npt=", &p))
180  return;
181 
182  *start = AV_NOPTS_VALUE;
183  *end = AV_NOPTS_VALUE;
184 
185  get_word_sep(buf, sizeof(buf), "-", &p);
186  if (av_parse_time(start, buf, 1) < 0)
187  return;
188  if (*p == '-') {
189  p++;
190  get_word_sep(buf, sizeof(buf), "-", &p);
191  if (av_parse_time(end, buf, 1) < 0)
192  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
193  }
194 }
195 
197  const char *buf, struct sockaddr_storage *sock)
198 {
199  struct addrinfo hints = { 0 }, *ai = NULL;
200  int ret;
201 
202  hints.ai_flags = AI_NUMERICHOST;
203  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
204  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
205  buf,
206  gai_strerror(ret));
207  return -1;
208  }
209  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
210  freeaddrinfo(ai);
211  return 0;
212 }
213 
214 #if CONFIG_RTPDEC
215 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
216  RTSPStream *rtsp_st, AVStream *st)
217 {
218  AVCodecParameters *par = st ? st->codecpar : NULL;
219  if (!handler)
220  return;
221  if (par)
222  par->codec_id = handler->codec_id;
223  rtsp_st->dynamic_handler = handler;
224  if (st)
225  ffstream(st)->need_parsing = handler->need_parsing;
226  if (handler->priv_data_size) {
227  rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
228  if (!rtsp_st->dynamic_protocol_context)
229  rtsp_st->dynamic_handler = NULL;
230  }
231 }
232 
233 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
234  AVStream *st)
235 {
236  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
237  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
238  rtsp_st->dynamic_protocol_context);
239  if (ret < 0) {
240  if (rtsp_st->dynamic_protocol_context) {
241  if (rtsp_st->dynamic_handler->close)
242  rtsp_st->dynamic_handler->close(
243  rtsp_st->dynamic_protocol_context);
245  }
246  rtsp_st->dynamic_protocol_context = NULL;
247  rtsp_st->dynamic_handler = NULL;
248  }
249  }
250 }
251 
252 #if CONFIG_RTSP_DEMUXER
253 static int init_satip_stream(AVFormatContext *s)
254 {
255  RTSPState *rt = s->priv_data;
256  RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream));
257  if (!rtsp_st)
258  return AVERROR(ENOMEM);
260  &rt->nb_rtsp_streams, rtsp_st);
261 
262  rtsp_st->sdp_payload_type = 33; // MP2T
263  av_strlcpy(rtsp_st->control_url,
264  rt->control_uri, sizeof(rtsp_st->control_url));
265 
266  if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) {
268  if (!st)
269  return AVERROR(ENOMEM);
270  st->id = rt->nb_rtsp_streams - 1;
271  rtsp_st->stream_index = st->index;
274  } else {
275  rtsp_st->stream_index = -1;
276  init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
277  finalize_rtp_handler_init(s, rtsp_st, NULL);
278  }
279  return 0;
280 }
281 #endif
282 
283 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
284 static int sdp_parse_rtpmap(AVFormatContext *s,
285  AVStream *st, RTSPStream *rtsp_st,
286  int payload_type, const char *p)
287 {
288  AVCodecParameters *par = st->codecpar;
289  char buf[256];
290  int i;
291  const AVCodecDescriptor *desc;
292  const char *c_name;
293 
294  /* See if we can handle this kind of payload.
295  * The space should normally not be there but some Real streams or
296  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
297  * have a trailing space. */
298  get_word_sep(buf, sizeof(buf), "/ ", &p);
299  if (payload_type < RTP_PT_PRIVATE) {
300  /* We are in a standard case
301  * (from http://www.iana.org/assignments/rtp-parameters). */
302  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
303  }
304 
305  if (par->codec_id == AV_CODEC_ID_NONE) {
308  init_rtp_handler(handler, rtsp_st, st);
309  /* If no dynamic handler was found, check with the list of standard
310  * allocated types, if such a stream for some reason happens to
311  * use a private payload type. This isn't handled in rtpdec.c, since
312  * the format name from the rtpmap line never is passed into rtpdec. */
313  if (!rtsp_st->dynamic_handler)
314  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
315  }
316 
318  if (desc && desc->name)
319  c_name = desc->name;
320  else
321  c_name = "(null)";
322 
323  get_word_sep(buf, sizeof(buf), "/", &p);
324  i = atoi(buf);
325  switch (par->codec_type) {
326  case AVMEDIA_TYPE_AUDIO:
327  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
330  if (i > 0) {
331  par->sample_rate = i;
332  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
333  get_word_sep(buf, sizeof(buf), "/", &p);
334  i = atoi(buf);
335  if (i > 0)
337  }
338  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
339  par->sample_rate);
340  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
341  par->ch_layout.nb_channels);
342  break;
343  case AVMEDIA_TYPE_VIDEO:
344  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
345  if (i > 0)
346  avpriv_set_pts_info(st, 32, 1, i);
347  break;
348  default:
349  break;
350  }
351  finalize_rtp_handler_init(s, rtsp_st, st);
352  return 0;
353 }
354 
355 /* parse the attribute line from the fmtp a line of an sdp response. This
356  * is broken out as a function because it is used in rtp_h264.c, which is
357  * forthcoming. */
358 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
359  char *value, int value_size)
360 {
361  *p += strspn(*p, SPACE_CHARS);
362  if (**p) {
363  get_word_sep(attr, attr_size, "=", p);
364  if (**p == '=')
365  (*p)++;
366  get_word_sep(value, value_size, ";", p);
367  if (**p == ';')
368  (*p)++;
369  return 1;
370  }
371  return 0;
372 }
373 
374 typedef struct SDPParseState {
375  /* SDP only */
376  struct sockaddr_storage default_ip;
377  int default_ttl;
378  int skip_media; ///< set if an unknown m= line occurs
379  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
380  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
381  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
382  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
383  int seen_rtpmap;
384  int seen_fmtp;
385  char delayed_fmtp[2048];
386 } SDPParseState;
387 
388 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
389  struct RTSPSource ***dest, int *dest_count)
390 {
391  RTSPSource *rtsp_src, *rtsp_src2;
392  int i;
393  for (i = 0; i < count; i++) {
394  rtsp_src = addrs[i];
395  rtsp_src2 = av_memdup(rtsp_src, sizeof(*rtsp_src));
396  if (!rtsp_src2)
397  continue;
398  dynarray_add(dest, dest_count, rtsp_src2);
399  }
400 }
401 
402 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
403  int payload_type, const char *line)
404 {
405  int i;
406 
407  for (i = 0; i < rt->nb_rtsp_streams; i++) {
408  RTSPStream *rtsp_st = rt->rtsp_streams[i];
409  if (rtsp_st->sdp_payload_type == payload_type &&
410  rtsp_st->dynamic_handler &&
411  rtsp_st->dynamic_handler->parse_sdp_a_line) {
412  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
413  rtsp_st->dynamic_protocol_context, line);
414  }
415  }
416 }
417 
418 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
419  int letter, const char *buf)
420 {
421  RTSPState *rt = s->priv_data;
422  char buf1[64], st_type[64];
423  const char *p;
424  enum AVMediaType codec_type;
425  int payload_type;
426  AVStream *st;
427  RTSPStream *rtsp_st;
428  RTSPSource *rtsp_src;
429  struct sockaddr_storage sdp_ip;
430  int ttl;
431 
432  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
433 
434  p = buf;
435  if (s1->skip_media && letter != 'm')
436  return;
437  switch (letter) {
438  case 'c':
439  get_word(buf1, sizeof(buf1), &p);
440  if (strcmp(buf1, "IN") != 0)
441  return;
442  get_word(buf1, sizeof(buf1), &p);
443  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
444  return;
445  get_word_sep(buf1, sizeof(buf1), "/", &p);
446  if (get_sockaddr(s, buf1, &sdp_ip))
447  return;
448  ttl = 16;
449  if (*p == '/') {
450  p++;
451  get_word_sep(buf1, sizeof(buf1), "/", &p);
452  ttl = atoi(buf1);
453  }
454  if (s->nb_streams == 0) {
455  s1->default_ip = sdp_ip;
456  s1->default_ttl = ttl;
457  } else {
458  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
459  rtsp_st->sdp_ip = sdp_ip;
460  rtsp_st->sdp_ttl = ttl;
461  }
462  break;
463  case 's':
464  av_dict_set(&s->metadata, "title", p, 0);
465  break;
466  case 'i':
467  if (s->nb_streams == 0) {
468  av_dict_set(&s->metadata, "comment", p, 0);
469  break;
470  }
471  break;
472  case 'm':
473  /* new stream */
474  s1->skip_media = 0;
475  s1->seen_fmtp = 0;
476  s1->seen_rtpmap = 0;
478  get_word(st_type, sizeof(st_type), &p);
479  if (!strcmp(st_type, "audio")) {
481  } else if (!strcmp(st_type, "video")) {
483  } else if (!strcmp(st_type, "application")) {
485  } else if (!strcmp(st_type, "text")) {
487  }
489  !(rt->media_type_mask & (1 << codec_type)) ||
490  rt->nb_rtsp_streams >= s->max_streams
491  ) {
492  s1->skip_media = 1;
493  return;
494  }
495  rtsp_st = av_mallocz(sizeof(RTSPStream));
496  if (!rtsp_st)
497  return;
498  rtsp_st->stream_index = -1;
499  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
500 
501  rtsp_st->sdp_ip = s1->default_ip;
502  rtsp_st->sdp_ttl = s1->default_ttl;
503 
504  copy_default_source_addrs(s1->default_include_source_addrs,
505  s1->nb_default_include_source_addrs,
506  &rtsp_st->include_source_addrs,
507  &rtsp_st->nb_include_source_addrs);
508  copy_default_source_addrs(s1->default_exclude_source_addrs,
509  s1->nb_default_exclude_source_addrs,
510  &rtsp_st->exclude_source_addrs,
511  &rtsp_st->nb_exclude_source_addrs);
512 
513  get_word(buf1, sizeof(buf1), &p); /* port */
514  rtsp_st->sdp_port = atoi(buf1);
515 
516  get_word(buf1, sizeof(buf1), &p); /* protocol */
517  if (!strcmp(buf1, "udp"))
519  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
520  rtsp_st->feedback = 1;
521 
522  /* XXX: handle list of formats */
523  get_word(buf1, sizeof(buf1), &p); /* format list */
524  rtsp_st->sdp_payload_type = atoi(buf1);
525 
526  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
527  /* no corresponding stream */
528  if (rt->transport == RTSP_TRANSPORT_RAW) {
529  if (CONFIG_RTPDEC && !rt->ts)
531  } else {
535  init_rtp_handler(handler, rtsp_st, NULL);
536  finalize_rtp_handler_init(s, rtsp_st, NULL);
537  }
538  } else if (rt->server_type == RTSP_SERVER_WMS &&
540  /* RTX stream, a stream that carries all the other actual
541  * audio/video streams. Don't expose this to the callers. */
542  } else {
543  st = avformat_new_stream(s, NULL);
544  if (!st)
545  return;
546  st->id = rt->nb_rtsp_streams - 1;
547  rtsp_st->stream_index = st->index;
549  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
551  /* if standard payload type, we can find the codec right now */
553  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
554  st->codecpar->sample_rate > 0)
555  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
556  /* Even static payload types may need a custom depacketizer */
558  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
559  init_rtp_handler(handler, rtsp_st, st);
560  finalize_rtp_handler_init(s, rtsp_st, st);
561  }
562  if (rt->default_lang[0])
563  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
564  }
565  /* put a default control url */
566  av_strlcpy(rtsp_st->control_url, rt->control_uri,
567  sizeof(rtsp_st->control_url));
568  break;
569  case 'a':
570  if (av_strstart(p, "control:", &p)) {
571  if (rt->nb_rtsp_streams == 0) {
572  if (!strncmp(p, "rtsp://", 7))
573  av_strlcpy(rt->control_uri, p,
574  sizeof(rt->control_uri));
575  } else {
576  char proto[32];
577  /* get the control url */
578  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
579 
580  /* XXX: may need to add full url resolution */
581  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
582  NULL, NULL, 0, p);
583  if (proto[0] == '\0') {
584  /* relative control URL */
585  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
586  av_strlcat(rtsp_st->control_url, "/",
587  sizeof(rtsp_st->control_url));
588  av_strlcat(rtsp_st->control_url, p,
589  sizeof(rtsp_st->control_url));
590  } else
591  av_strlcpy(rtsp_st->control_url, p,
592  sizeof(rtsp_st->control_url));
593  }
594  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
595  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
596  get_word(buf1, sizeof(buf1), &p);
597  payload_type = atoi(buf1);
598  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
599  if (rtsp_st->stream_index >= 0) {
600  st = s->streams[rtsp_st->stream_index];
601  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
602  }
603  s1->seen_rtpmap = 1;
604  if (s1->seen_fmtp) {
605  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
606  }
607  } else if (av_strstart(p, "fmtp:", &p) ||
608  av_strstart(p, "framesize:", &p)) {
609  // let dynamic protocol handlers have a stab at the line.
610  get_word(buf1, sizeof(buf1), &p);
611  payload_type = atoi(buf1);
612  if (s1->seen_rtpmap) {
613  parse_fmtp(s, rt, payload_type, buf);
614  } else {
615  s1->seen_fmtp = 1;
616  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
617  }
618  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
619  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
620  get_word(buf1, sizeof(buf1), &p);
621  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
622  } else if (av_strstart(p, "range:", &p)) {
623  int64_t start, end;
624 
625  // this is so that seeking on a streamed file can work.
626  rtsp_parse_range_npt(p, &start, &end);
627  s->start_time = start;
628  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
629  s->duration = (end == AV_NOPTS_VALUE) ?
630  AV_NOPTS_VALUE : end - start;
631  } else if (av_strstart(p, "lang:", &p)) {
632  if (s->nb_streams > 0) {
633  get_word(buf1, sizeof(buf1), &p);
634  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
635  if (rtsp_st->stream_index >= 0) {
636  st = s->streams[rtsp_st->stream_index];
637  av_dict_set(&st->metadata, "language", buf1, 0);
638  }
639  } else
640  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
641  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
642  if (atoi(p) == 1)
644  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
645  s->nb_streams > 0) {
646  st = s->streams[s->nb_streams - 1];
647  st->codecpar->sample_rate = atoi(p);
648  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
649  // RFC 4568
650  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
651  get_word(buf1, sizeof(buf1), &p); // ignore tag
652  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
653  p += strspn(p, SPACE_CHARS);
654  if (av_strstart(p, "inline:", &p))
655  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
656  } else if (av_strstart(p, "source-filter:", &p)) {
657  int exclude = 0;
658  get_word(buf1, sizeof(buf1), &p);
659  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
660  return;
661  exclude = !strcmp(buf1, "excl");
662 
663  get_word(buf1, sizeof(buf1), &p);
664  if (strcmp(buf1, "IN") != 0)
665  return;
666  get_word(buf1, sizeof(buf1), &p);
667  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
668  return;
669  // not checking that the destination address actually matches or is wildcard
670  get_word(buf1, sizeof(buf1), &p);
671 
672  while (*p != '\0') {
673  rtsp_src = av_mallocz(sizeof(*rtsp_src));
674  if (!rtsp_src)
675  return;
676  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
677  if (exclude) {
678  if (s->nb_streams == 0) {
679  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
680  } else {
681  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
682  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
683  }
684  } else {
685  if (s->nb_streams == 0) {
686  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
687  } else {
688  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
689  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
690  }
691  }
692  }
693  } else {
694  if (rt->server_type == RTSP_SERVER_WMS)
696  if (s->nb_streams > 0) {
697  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
698 
699  if (rt->server_type == RTSP_SERVER_REAL)
701 
702  if (rtsp_st->dynamic_handler &&
705  rtsp_st->stream_index,
706  rtsp_st->dynamic_protocol_context, buf);
707  }
708  }
709  break;
710  }
711 }
712 
713 int ff_sdp_parse(AVFormatContext *s, const char *content)
714 {
715  const char *p;
716  int letter, i;
717  char buf[SDP_MAX_SIZE], *q;
718  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
719 
720  p = content;
721  for (;;) {
722  p += strspn(p, SPACE_CHARS);
723  letter = *p;
724  if (letter == '\0')
725  break;
726  p++;
727  if (*p != '=')
728  goto next_line;
729  p++;
730  /* get the content */
731  q = buf;
732  while (*p != '\n' && *p != '\r' && *p != '\0') {
733  if ((q - buf) < sizeof(buf) - 1)
734  *q++ = *p;
735  p++;
736  }
737  *q = '\0';
738  sdp_parse_line(s, s1, letter, buf);
739  next_line:
740  while (*p != '\n' && *p != '\0')
741  p++;
742  if (*p == '\n')
743  p++;
744  }
745 
746  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
747  av_freep(&s1->default_include_source_addrs[i]);
748  av_freep(&s1->default_include_source_addrs);
749  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
750  av_freep(&s1->default_exclude_source_addrs[i]);
751  av_freep(&s1->default_exclude_source_addrs);
752 
753  return 0;
754 }
755 #endif /* CONFIG_RTPDEC */
756 
757 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
758 {
759  RTSPState *rt = s->priv_data;
760  int i;
761 
762  for (i = 0; i < rt->nb_rtsp_streams; i++) {
763  RTSPStream *rtsp_st = rt->rtsp_streams[i];
764  if (!rtsp_st)
765  continue;
766  if (rtsp_st->transport_priv) {
767  if (s->oformat) {
768  AVFormatContext *rtpctx = rtsp_st->transport_priv;
769  av_write_trailer(rtpctx);
771  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
772  ff_rtsp_tcp_write_packet(s, rtsp_st);
773  ffio_free_dyn_buf(&rtpctx->pb);
774  } else {
775  avio_closep(&rtpctx->pb);
776  }
777  avformat_free_context(rtpctx);
778  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
780  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
782  }
783  rtsp_st->transport_priv = NULL;
784  ffurl_closep(&rtsp_st->rtp_handle);
785  }
786 }
787 
788 /* close and free RTSP streams */
790 {
791  RTSPState *rt = s->priv_data;
792  int i, j;
793  RTSPStream *rtsp_st;
794 
795  ff_rtsp_undo_setup(s, 0);
796  for (i = 0; i < rt->nb_rtsp_streams; i++) {
797  rtsp_st = rt->rtsp_streams[i];
798  if (rtsp_st) {
799  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
800  if (rtsp_st->dynamic_handler->close)
801  rtsp_st->dynamic_handler->close(
802  rtsp_st->dynamic_protocol_context);
804  }
805  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
806  av_freep(&rtsp_st->include_source_addrs[j]);
807  av_freep(&rtsp_st->include_source_addrs);
808  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
809  av_freep(&rtsp_st->exclude_source_addrs[j]);
810  av_freep(&rtsp_st->exclude_source_addrs);
811 
812  av_freep(&rtsp_st);
813  }
814  }
815  av_freep(&rt->rtsp_streams);
816  if (rt->asf_ctx) {
818  }
819  if (CONFIG_RTPDEC && rt->ts)
821  av_freep(&rt->p);
822  av_freep(&rt->recvbuf);
823 }
824 
826 {
827  RTSPState *rt = s->priv_data;
828  AVStream *st = NULL;
829  int reordering_queue_size = rt->reordering_queue_size;
830  if (reordering_queue_size < 0) {
831  if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
832  reordering_queue_size = 0;
833  else
834  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
835  }
836 
837  /* open the RTP context */
838  if (rtsp_st->stream_index >= 0)
839  st = s->streams[rtsp_st->stream_index];
840  if (!st)
841  s->ctx_flags |= AVFMTCTX_NOHEADER;
842 
843  if (CONFIG_RTSP_MUXER && s->oformat && st) {
845  s, st, rtsp_st->rtp_handle,
846  rt->pkt_size,
847  rtsp_st->stream_index);
848  /* Ownership of rtp_handle is passed to the rtp mux context */
849  rtsp_st->rtp_handle = NULL;
850  if (ret < 0)
851  return ret;
852  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
853  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
854  return 0; // Don't need to open any parser here
855  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
856  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
857  rtsp_st->dynamic_protocol_context,
858  rtsp_st->dynamic_handler);
859  else if (CONFIG_RTPDEC)
860  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
861  rtsp_st->sdp_payload_type,
862  reordering_queue_size);
863 
864  if (!rtsp_st->transport_priv) {
865  return AVERROR(ENOMEM);
866  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
867  s->iformat) {
868  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
869  rtpctx->ssrc = rtsp_st->ssrc;
870  if (rtsp_st->dynamic_handler) {
872  rtsp_st->dynamic_protocol_context,
873  rtsp_st->dynamic_handler);
874  }
875  if (rtsp_st->crypto_suite[0])
877  rtsp_st->crypto_suite,
878  rtsp_st->crypto_params);
879  }
880 
881  return 0;
882 }
883 
884 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
885 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
886 {
887  const char *q;
888  char *p;
889  int v;
890 
891  q = *pp;
892  q += strspn(q, SPACE_CHARS);
893  v = strtol(q, &p, 10);
894  if (*p == '-') {
895  p++;
896  *min_ptr = v;
897  v = strtol(p, &p, 10);
898  *max_ptr = v;
899  } else {
900  *min_ptr = v;
901  *max_ptr = v;
902  }
903  *pp = p;
904 }
905 
906 /* XXX: only one transport specification is parsed */
907 static void rtsp_parse_transport(AVFormatContext *s,
908  RTSPMessageHeader *reply, const char *p)
909 {
910  char transport_protocol[16];
911  char profile[16];
912  char lower_transport[16];
913  char parameter[16];
915  char buf[256];
916 
917  reply->nb_transports = 0;
918 
919  for (;;) {
920  p += strspn(p, SPACE_CHARS);
921  if (*p == '\0')
922  break;
923 
924  th = &reply->transports[reply->nb_transports];
925 
926  get_word_sep(transport_protocol, sizeof(transport_protocol),
927  "/", &p);
928  if (!av_strcasecmp (transport_protocol, "rtp")) {
929  get_word_sep(profile, sizeof(profile), "/;,", &p);
930  lower_transport[0] = '\0';
931  /* rtp/avp/<protocol> */
932  if (*p == '/') {
933  get_word_sep(lower_transport, sizeof(lower_transport),
934  ";,", &p);
935  }
936  th->transport = RTSP_TRANSPORT_RTP;
937  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
938  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
939  /* x-pn-tng/<protocol> */
940  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
941  profile[0] = '\0';
942  th->transport = RTSP_TRANSPORT_RDT;
943  } else if (!av_strcasecmp(transport_protocol, "raw")) {
944  get_word_sep(profile, sizeof(profile), "/;,", &p);
945  lower_transport[0] = '\0';
946  /* raw/raw/<protocol> */
947  if (*p == '/') {
948  get_word_sep(lower_transport, sizeof(lower_transport),
949  ";,", &p);
950  }
951  th->transport = RTSP_TRANSPORT_RAW;
952  }
953  if (!av_strcasecmp(lower_transport, "TCP"))
954  th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
955  else
956  th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
957 
958  if (*p == ';')
959  p++;
960  /* get each parameter */
961  while (*p != '\0' && *p != ',') {
962  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
963  if (!strcmp(parameter, "port")) {
964  if (*p == '=') {
965  p++;
966  rtsp_parse_range(&th->port_min, &th->port_max, &p);
967  }
968  } else if (!strcmp(parameter, "client_port")) {
969  if (*p == '=') {
970  p++;
971  rtsp_parse_range(&th->client_port_min,
972  &th->client_port_max, &p);
973  }
974  } else if (!strcmp(parameter, "server_port")) {
975  if (*p == '=') {
976  p++;
977  rtsp_parse_range(&th->server_port_min,
978  &th->server_port_max, &p);
979  }
980  } else if (!strcmp(parameter, "interleaved")) {
981  if (*p == '=') {
982  p++;
983  rtsp_parse_range(&th->interleaved_min,
984  &th->interleaved_max, &p);
985  }
986  } else if (!strcmp(parameter, "multicast")) {
987  if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
988  th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
989  } else if (!strcmp(parameter, "ttl")) {
990  if (*p == '=') {
991  char *end;
992  p++;
993  th->ttl = strtol(p, &end, 10);
994  p = end;
995  }
996  } else if (!strcmp(parameter, "destination")) {
997  if (*p == '=') {
998  p++;
999  get_word_sep(buf, sizeof(buf), ";,", &p);
1000  get_sockaddr(s, buf, &th->destination);
1001  }
1002  } else if (!strcmp(parameter, "source")) {
1003  if (*p == '=') {
1004  p++;
1005  get_word_sep(buf, sizeof(buf), ";,", &p);
1006  av_strlcpy(th->source, buf, sizeof(th->source));
1007  }
1008  } else if (!strcmp(parameter, "mode")) {
1009  if (*p == '=') {
1010  p++;
1011  get_word_sep(buf, sizeof(buf), ";, ", &p);
1012  if (!strcmp(buf, "record") ||
1013  !strcmp(buf, "receive"))
1014  th->mode_record = 1;
1015  }
1016  }
1017 
1018  while (*p != ';' && *p != '\0' && *p != ',')
1019  p++;
1020  if (*p == ';')
1021  p++;
1022  }
1023  if (*p == ',')
1024  p++;
1025 
1026  reply->nb_transports++;
1027  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1028  break;
1029  }
1030 }
1031 
1032 static void handle_rtp_info(RTSPState *rt, const char *url,
1033  uint32_t seq, uint32_t rtptime)
1034 {
1035  int i;
1036  if (!rtptime || !url[0])
1037  return;
1038  if (rt->transport != RTSP_TRANSPORT_RTP)
1039  return;
1040  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1041  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1042  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1043  if (!rtpctx)
1044  continue;
1045  if (!strcmp(rtsp_st->control_url, url)) {
1046  rtpctx->base_timestamp = rtptime;
1047  break;
1048  }
1049  }
1050 }
1051 
1052 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1053 {
1054  int read = 0;
1055  char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
1056  uint32_t seq = 0, rtptime = 0;
1057 
1058  for (;;) {
1059  p += strspn(p, SPACE_CHARS);
1060  if (!*p)
1061  break;
1062  get_word_sep(key, sizeof(key), "=", &p);
1063  if (*p != '=')
1064  break;
1065  p++;
1066  get_word_sep(value, sizeof(value), ";, ", &p);
1067  read++;
1068  if (!strcmp(key, "url"))
1069  av_strlcpy(url, value, sizeof(url));
1070  else if (!strcmp(key, "seq"))
1071  seq = strtoul(value, NULL, 10);
1072  else if (!strcmp(key, "rtptime"))
1073  rtptime = strtoul(value, NULL, 10);
1074  if (*p == ',') {
1075  handle_rtp_info(rt, url, seq, rtptime);
1076  url[0] = '\0';
1077  seq = rtptime = 0;
1078  read = 0;
1079  }
1080  if (*p)
1081  p++;
1082  }
1083  if (read > 0)
1084  handle_rtp_info(rt, url, seq, rtptime);
1085 }
1086 
1088  RTSPMessageHeader *reply, const char *buf,
1089  RTSPState *rt, const char *method)
1090 {
1091  const char *p;
1092 
1093  /* NOTE: we do case independent match for broken servers */
1094  p = buf;
1095  if (av_stristart(p, "Session:", &p)) {
1096  int t;
1097  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1098  if (av_stristart(p, ";timeout=", &p) &&
1099  (t = strtol(p, NULL, 10)) > 0) {
1100  reply->timeout = t;
1101  }
1102  } else if (av_stristart(p, "Content-Length:", &p)) {
1103  reply->content_length = strtol(p, NULL, 10);
1104  } else if (av_stristart(p, "Transport:", &p)) {
1105  rtsp_parse_transport(s, reply, p);
1106  } else if (av_stristart(p, "CSeq:", &p)) {
1107  reply->seq = strtol(p, NULL, 10);
1108  } else if (av_stristart(p, "Range:", &p)) {
1109  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1110  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1111  p += strspn(p, SPACE_CHARS);
1112  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1113  } else if (av_stristart(p, "Server:", &p)) {
1114  p += strspn(p, SPACE_CHARS);
1115  av_strlcpy(reply->server, p, sizeof(reply->server));
1116  } else if (av_stristart(p, "Notice:", &p) ||
1117  av_stristart(p, "X-Notice:", &p)) {
1118  reply->notice = strtol(p, NULL, 10);
1119  } else if (av_stristart(p, "Location:", &p)) {
1120  p += strspn(p, SPACE_CHARS);
1121  av_strlcpy(reply->location, p , sizeof(reply->location));
1122  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1123  p += strspn(p, SPACE_CHARS);
1124  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1125  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1126  p += strspn(p, SPACE_CHARS);
1127  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1128  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1129  p += strspn(p, SPACE_CHARS);
1130  if (method && !strcmp(method, "DESCRIBE"))
1131  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1132  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1133  p += strspn(p, SPACE_CHARS);
1134  if (method && !strcmp(method, "PLAY"))
1135  rtsp_parse_rtp_info(rt, p);
1136  } else if (av_stristart(p, "Public:", &p) && rt) {
1137  if (strstr(p, "GET_PARAMETER") &&
1138  method && !strcmp(method, "OPTIONS"))
1139  rt->get_parameter_supported = 1;
1140  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1141  p += strspn(p, SPACE_CHARS);
1142  rt->accept_dynamic_rate = atoi(p);
1143  } else if (av_stristart(p, "Content-Type:", &p)) {
1144  p += strspn(p, SPACE_CHARS);
1145  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1146  } else if (av_stristart(p, "com.ses.streamID:", &p)) {
1147  p += strspn(p, SPACE_CHARS);
1148  av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
1149  }
1150 }
1151 
1152 /* skip a RTP/TCP interleaved packet */
1154 {
1155  RTSPState *rt = s->priv_data;
1156  int ret, len, len1;
1157  uint8_t buf[MAX_URL_SIZE];
1158 
1159  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1160  if (ret != 3)
1161  return ret < 0 ? ret : AVERROR(EIO);
1162  len = AV_RB16(buf + 1);
1163 
1164  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1165 
1166  /* skip payload */
1167  while (len > 0) {
1168  len1 = len;
1169  if (len1 > sizeof(buf))
1170  len1 = sizeof(buf);
1171  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1172  if (ret != len1)
1173  return ret < 0 ? ret : AVERROR(EIO);
1174  len -= len1;
1175  }
1176 
1177  return 0;
1178 }
1179 
1181  unsigned char **content_ptr,
1182  int return_on_interleaved_data, const char *method)
1183 {
1184  RTSPState *rt = s->priv_data;
1185  char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
1186  unsigned char ch;
1187  const char *p;
1188  int ret, content_length, line_count, request;
1189  unsigned char *content;
1190 
1191 start:
1192  line_count = 0;
1193  request = 0;
1194  content = NULL;
1195  memset(reply, 0, sizeof(*reply));
1196 
1197  /* parse reply (XXX: use buffers) */
1198  rt->last_reply[0] = '\0';
1199  for (;;) {
1200  q = buf;
1201  for (;;) {
1202  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1203  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1204  if (ret != 1)
1205  return ret < 0 ? ret : AVERROR(EIO);
1206  if (ch == '\n')
1207  break;
1208  if (ch == '$' && q == buf) {
1209  if (return_on_interleaved_data) {
1210  return 1;
1211  } else {
1213  if (ret < 0)
1214  return ret;
1215  }
1216  } else if (ch != '\r') {
1217  if ((q - buf) < sizeof(buf) - 1)
1218  *q++ = ch;
1219  }
1220  }
1221  *q = '\0';
1222 
1223  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1224 
1225  /* test if last line */
1226  if (buf[0] == '\0')
1227  break;
1228  p = buf;
1229  if (line_count == 0) {
1230  /* get reply code */
1231  get_word(buf1, sizeof(buf1), &p);
1232  if (!strncmp(buf1, "RTSP/", 5)) {
1233  get_word(buf1, sizeof(buf1), &p);
1234  reply->status_code = atoi(buf1);
1235  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1236  } else {
1237  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1238  get_word(buf1, sizeof(buf1), &p); // object
1239  request = 1;
1240  }
1241  } else {
1242  ff_rtsp_parse_line(s, reply, p, rt, method);
1243  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1244  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1245  }
1246  line_count++;
1247  }
1248 
1249  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1250  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1251 
1252  content_length = reply->content_length;
1253  if (content_length > 0) {
1254  /* leave some room for a trailing '\0' (useful for simple parsing) */
1255  content = av_malloc(content_length + 1);
1256  if (!content)
1257  return AVERROR(ENOMEM);
1258  if ((ret = ffurl_read_complete(rt->rtsp_hd, content, content_length)) != content_length) {
1259  av_freep(&content);
1260  return ret < 0 ? ret : AVERROR(EIO);
1261  }
1262  content[content_length] = '\0';
1263  }
1264  if (content_ptr)
1265  *content_ptr = content;
1266  else
1267  av_freep(&content);
1268 
1269  if (request) {
1270  char buf[MAX_URL_SIZE];
1271  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1272  const char* ptr = buf;
1273 
1274  if (!strcmp(reply->reason, "OPTIONS") ||
1275  !strcmp(reply->reason, "GET_PARAMETER")) {
1276  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1277  if (reply->seq)
1278  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1279  if (reply->session_id[0])
1280  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1281  reply->session_id);
1282  } else {
1283  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1284  }
1285  av_strlcat(buf, "\r\n", sizeof(buf));
1286 
1287  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1288  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1289  ptr = base64buf;
1290  }
1291  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1292 
1294  /* Even if the request from the server had data, it is not the data
1295  * that the caller wants or expects. The memory could also be leaked
1296  * if the actual following reply has content data. */
1297  if (content_ptr)
1298  av_freep(content_ptr);
1299  /* If method is set, this is called from ff_rtsp_send_cmd,
1300  * where a reply to exactly this request is awaited. For
1301  * callers from within packet receiving, we just want to
1302  * return to the caller and go back to receiving packets. */
1303  if (method)
1304  goto start;
1305  return 0;
1306  }
1307 
1308  if (rt->seq != reply->seq) {
1309  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1310  rt->seq, reply->seq);
1311  }
1312 
1313  /* EOS */
1314  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1315  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1316  reply->notice == 2306 /* Continuous Feed Terminated */) {
1317  rt->state = RTSP_STATE_IDLE;
1318  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1319  return AVERROR(EIO); /* data or server error */
1320  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1321  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1322  return AVERROR(EPERM);
1323 
1324  return 0;
1325 }
1326 
1327 /**
1328  * Send a command to the RTSP server without waiting for the reply.
1329  *
1330  * @param s RTSP (de)muxer context
1331  * @param method the method for the request
1332  * @param url the target url for the request
1333  * @param headers extra header lines to include in the request
1334  * @param send_content if non-null, the data to send as request body content
1335  * @param send_content_length the length of the send_content data, or 0 if
1336  * send_content is null
1337  *
1338  * @return zero if success, nonzero otherwise
1339  */
1340 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1341  const char *method, const char *url,
1342  const char *headers,
1343  const unsigned char *send_content,
1344  int send_content_length)
1345 {
1346  RTSPState *rt = s->priv_data;
1347  char buf[MAX_URL_SIZE], *out_buf;
1348  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1349 
1350  if (!rt->rtsp_hd_out)
1351  return AVERROR(ENOTCONN);
1352 
1353  /* Add in RTSP headers */
1354  out_buf = buf;
1355  rt->seq++;
1356  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1357  if (headers)
1358  av_strlcat(buf, headers, sizeof(buf));
1359  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1360  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1361  if (rt->session_id[0] != '\0' && (!headers ||
1362  !strstr(headers, "\nIf-Match:"))) {
1363  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1364  }
1365  if (rt->auth[0]) {
1367  rt->auth, url, method);
1368  if (str)
1369  av_strlcat(buf, str, sizeof(buf));
1370  av_free(str);
1371  }
1372  if (send_content_length > 0 && send_content)
1373  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1374  av_strlcat(buf, "\r\n", sizeof(buf));
1375 
1376  /* base64 encode rtsp if tunneling */
1377  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1378  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1379  out_buf = base64buf;
1380  }
1381 
1382  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1383 
1384  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1385  if (send_content_length > 0 && send_content) {
1386  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1387  avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1388  return AVERROR_PATCHWELCOME;
1389  }
1390  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1391  }
1393 
1394  return 0;
1395 }
1396 
1397 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1398  const char *url, const char *headers)
1399 {
1400  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1401 }
1402 
1403 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1404  const char *headers, RTSPMessageHeader *reply,
1405  unsigned char **content_ptr)
1406 {
1407  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1408  content_ptr, NULL, 0);
1409 }
1410 
1412  const char *method, const char *url,
1413  const char *header,
1414  RTSPMessageHeader *reply,
1415  unsigned char **content_ptr,
1416  const unsigned char *send_content,
1417  int send_content_length)
1418 {
1419  RTSPState *rt = s->priv_data;
1420  HTTPAuthType cur_auth_type;
1421  int ret, attempts = 0;
1422 
1423 retry:
1424  cur_auth_type = rt->auth_state.auth_type;
1425  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1426  send_content,
1427  send_content_length)))
1428  return ret;
1429 
1430  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1431  return ret;
1432  attempts++;
1433 
1434  if (reply->status_code == 401 &&
1435  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1436  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1437  goto retry;
1438 
1439  if (reply->status_code > 400){
1440  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1441  method,
1442  reply->status_code,
1443  reply->reason);
1444  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1445  }
1446 
1447  return 0;
1448 }
1449 
1450 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1451  int lower_transport, const char *real_challenge)
1452 {
1453  RTSPState *rt = s->priv_data;
1454  int rtx = 0, j, i, err, interleave = 0, port_off = 0;
1455  RTSPStream *rtsp_st;
1456  RTSPMessageHeader reply1, *reply = &reply1;
1457  char cmd[MAX_URL_SIZE];
1458  const char *trans_pref;
1459 
1460  if (rt->transport == RTSP_TRANSPORT_RDT)
1461  trans_pref = "x-pn-tng";
1462  else if (rt->transport == RTSP_TRANSPORT_RAW)
1463  trans_pref = "RAW/RAW";
1464  else
1465  trans_pref = "RTP/AVP";
1466 
1467  /* default timeout: 1 minute */
1468  rt->timeout = 60;
1469 
1470  /* Choose a random starting offset within the first half of the
1471  * port range, to allow for a number of ports to try even if the offset
1472  * happens to be at the end of the random range. */
1473  if (rt->rtp_port_max - rt->rtp_port_min >= 4) {
1474  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1475  /* even random offset */
1476  port_off -= port_off & 0x01;
1477  }
1478 
1479  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1480  char transport[MAX_URL_SIZE];
1481 
1482  /*
1483  * WMS serves all UDP data over a single connection, the RTX, which
1484  * isn't necessarily the first in the SDP but has to be the first
1485  * to be set up, else the second/third SETUP will fail with a 461.
1486  */
1487  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1488  rt->server_type == RTSP_SERVER_WMS) {
1489  if (i == 0) {
1490  /* rtx first */
1491  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1492  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1493  if (len >= 4 &&
1494  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1495  "/rtx"))
1496  break;
1497  }
1498  if (rtx == rt->nb_rtsp_streams)
1499  return -1; /* no RTX found */
1500  rtsp_st = rt->rtsp_streams[rtx];
1501  } else
1502  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1503  } else
1504  rtsp_st = rt->rtsp_streams[i];
1505 
1506  /* RTP/UDP */
1507  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1508  char buf[256];
1509 
1510  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1511  port = reply->transports[0].client_port_min;
1512  goto have_port;
1513  }
1514 
1515  /* first try in specified port range */
1516  while (j + 1 <= rt->rtp_port_max) {
1517  AVDictionary *opts = map_to_opts(rt);
1518 
1519  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1520  "?localport=%d", j);
1521  /* we will use two ports per rtp stream (rtp and rtcp) */
1522  j += 2;
1524  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1525 
1526  av_dict_free(&opts);
1527 
1528  if (!err)
1529  goto rtp_opened;
1530  }
1531  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1532  err = AVERROR(EIO);
1533  goto fail;
1534 
1535  rtp_opened:
1536  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1537  have_port:
1538  av_strlcpy(transport, trans_pref, sizeof(transport));
1539  av_strlcat(transport,
1540  rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
1541  sizeof(transport));
1542  if (rt->server_type != RTSP_SERVER_REAL)
1543  av_strlcat(transport, "unicast;", sizeof(transport));
1544  av_strlcatf(transport, sizeof(transport),
1545  "client_port=%d", port);
1546  if (rt->transport == RTSP_TRANSPORT_RTP &&
1547  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1548  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1549  }
1550 
1551  /* RTP/TCP */
1552  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1553  /* For WMS streams, the application streams are only used for
1554  * UDP. When trying to set it up for TCP streams, the server
1555  * will return an error. Therefore, we skip those streams. */
1556  if (rt->server_type == RTSP_SERVER_WMS &&
1557  (rtsp_st->stream_index < 0 ||
1558  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1560  continue;
1561  snprintf(transport, sizeof(transport) - 1,
1562  "%s/TCP;", trans_pref);
1563  if (rt->transport != RTSP_TRANSPORT_RDT)
1564  av_strlcat(transport, "unicast;", sizeof(transport));
1565  av_strlcatf(transport, sizeof(transport),
1566  "interleaved=%d-%d",
1567  interleave, interleave + 1);
1568  interleave += 2;
1569  }
1570 
1571  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1572  snprintf(transport, sizeof(transport) - 1,
1573  "%s/UDP;multicast", trans_pref);
1574  }
1575  if (s->oformat) {
1576  av_strlcat(transport, ";mode=record", sizeof(transport));
1577  } else if (rt->server_type == RTSP_SERVER_REAL ||
1579  av_strlcat(transport, ";mode=play", sizeof(transport));
1580  snprintf(cmd, sizeof(cmd),
1581  "Transport: %s\r\n",
1582  transport);
1583  if (rt->accept_dynamic_rate)
1584  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1585  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1586  char real_res[41], real_csum[9];
1587  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1588  real_challenge);
1589  av_strlcatf(cmd, sizeof(cmd),
1590  "If-Match: %s\r\n"
1591  "RealChallenge2: %s, sd=%s\r\n",
1592  rt->session_id, real_res, real_csum);
1593  }
1594  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1595  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1596  err = 1;
1597  goto fail;
1598  } else if (reply->status_code != RTSP_STATUS_OK ||
1599  reply->nb_transports != 1) {
1601  goto fail;
1602  }
1603 
1604  if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
1605  char proto[128], host[128], path[512], auth[128];
1606  int port;
1607  av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
1608  &port, path, sizeof(path), rt->control_uri);
1609  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
1610  port, "/stream=%s", reply->stream_id);
1611  }
1612 
1613  /* XXX: same protocol for all streams is required */
1614  if (i > 0) {
1615  if (reply->transports[0].lower_transport != rt->lower_transport ||
1616  reply->transports[0].transport != rt->transport) {
1617  err = AVERROR_INVALIDDATA;
1618  goto fail;
1619  }
1620  } else {
1621  rt->lower_transport = reply->transports[0].lower_transport;
1622  rt->transport = reply->transports[0].transport;
1623  }
1624 
1625  /* Fail if the server responded with another lower transport mode
1626  * than what we requested. */
1627  if (reply->transports[0].lower_transport != lower_transport) {
1628  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1629  err = AVERROR_INVALIDDATA;
1630  goto fail;
1631  }
1632 
1633  switch(reply->transports[0].lower_transport) {
1635  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1636  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1637  break;
1638 
1639  case RTSP_LOWER_TRANSPORT_UDP: {
1640  char url[MAX_URL_SIZE], options[30] = "";
1641  const char *peer = host;
1642 
1643  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1644  av_strlcpy(options, "?connect=1", sizeof(options));
1645  /* Use source address if specified */
1646  if (reply->transports[0].source[0])
1647  peer = reply->transports[0].source;
1648  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1649  reply->transports[0].server_port_min, "%s", options);
1650  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1651  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1652  err = AVERROR_INVALIDDATA;
1653  goto fail;
1654  }
1655  break;
1656  }
1658  char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
1659  struct sockaddr_storage addr;
1660  int port, ttl;
1661  AVDictionary *opts = map_to_opts(rt);
1662 
1663  if (reply->transports[0].destination.ss_family) {
1664  addr = reply->transports[0].destination;
1665  port = reply->transports[0].port_min;
1666  ttl = reply->transports[0].ttl;
1667  } else {
1668  addr = rtsp_st->sdp_ip;
1669  port = rtsp_st->sdp_port;
1670  ttl = rtsp_st->sdp_ttl;
1671  }
1672  if (ttl > 0)
1673  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1674  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1675  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1676  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1677  port, "%s", optbuf);
1679  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1680  av_dict_free(&opts);
1681 
1682  if (err < 0) {
1683  err = AVERROR_INVALIDDATA;
1684  goto fail;
1685  }
1686  break;
1687  }
1688  }
1689 
1690  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1691  goto fail;
1692  }
1693 
1694  if (rt->nb_rtsp_streams && reply->timeout > 0)
1695  rt->timeout = reply->timeout;
1696 
1697  if (rt->server_type == RTSP_SERVER_REAL)
1698  rt->need_subscription = 1;
1699 
1700  return 0;
1701 
1702 fail:
1703  ff_rtsp_undo_setup(s, 0);
1704  return err;
1705 }
1706 
1708 {
1709  RTSPState *rt = s->priv_data;
1710  if (rt->rtsp_hd_out != rt->rtsp_hd)
1711  ffurl_closep(&rt->rtsp_hd_out);
1712  rt->rtsp_hd_out = NULL;
1713  ffurl_closep(&rt->rtsp_hd);
1714 }
1715 
1717 {
1718  RTSPState *rt = s->priv_data;
1719  char proto[128], host[1024], path[1024];
1720  char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
1721  const char *lower_rtsp_proto = "tcp";
1722  int port, err, tcp_fd;
1723  RTSPMessageHeader reply1, *reply = &reply1;
1724  int lower_transport_mask = 0;
1725  int default_port = RTSP_DEFAULT_PORT;
1726  int https_tunnel = 0;
1727  char real_challenge[64] = "";
1728  struct sockaddr_storage peer;
1729  socklen_t peer_len = sizeof(peer);
1730 
1731  if (rt->rtp_port_max < rt->rtp_port_min) {
1732  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1733  "than min port %d\n", rt->rtp_port_max,
1734  rt->rtp_port_min);
1735  return AVERROR(EINVAL);
1736  }
1737 
1738  if (!ff_network_init())
1739  return AVERROR(EIO);
1740 
1741  if (s->max_delay < 0) /* Not set by the caller */
1742  s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1743 
1746  (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1747  https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1750  }
1751  /* Only pass through valid flags from here */
1753 
1754 redirect:
1755  memset(&reply1, 0, sizeof(reply1));
1756  /* extract hostname and port */
1757  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1758  host, sizeof(host), &port, path, sizeof(path), s->url);
1759 
1760  if (!strcmp(proto, "rtsps")) {
1761  lower_rtsp_proto = "tls";
1762  default_port = RTSPS_DEFAULT_PORT;
1764  } else if (!strcmp(proto, "satip")) {
1765  av_strlcpy(proto, "rtsp", sizeof(proto));
1767  }
1768 
1769  if (*auth) {
1770  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1771  }
1772  if (port < 0)
1773  port = default_port;
1774 
1775  lower_transport_mask = rt->lower_transport_mask;
1776 
1777  if (!lower_transport_mask)
1778  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1779 
1780  if (s->oformat) {
1781  /* Only UDP or TCP - UDP multicast isn't supported. */
1782  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1783  (1 << RTSP_LOWER_TRANSPORT_TCP);
1784  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1785  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1786  "only UDP and TCP are supported for output.\n");
1787  err = AVERROR(EINVAL);
1788  goto fail;
1789  }
1790  }
1791 
1792  /* Construct the URI used in request; this is similar to s->url,
1793  * but with authentication credentials removed and RTSP specific options
1794  * stripped out. */
1795  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1796  host, port, "%s", path);
1797 
1798  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1799  /* set up initial handshake for tunneling */
1800  char httpname[1024];
1801  char sessioncookie[17];
1802  char headers[1024];
1804 
1805  av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1806 
1807  ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1808  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1810 
1811  /* GET requests */
1812  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1813  &s->interrupt_callback) < 0) {
1814  err = AVERROR(EIO);
1815  goto fail;
1816  }
1817 
1818  /* generate GET headers */
1819  snprintf(headers, sizeof(headers),
1820  "x-sessioncookie: %s\r\n"
1821  "Accept: application/x-rtsp-tunnelled\r\n"
1822  "Pragma: no-cache\r\n"
1823  "Cache-Control: no-cache\r\n",
1824  sessioncookie);
1825  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1826 
1827  if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1828  rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1829  if (!rt->rtsp_hd->protocol_whitelist) {
1830  err = AVERROR(ENOMEM);
1831  goto fail;
1832  }
1833  }
1834 
1835  /* complete the connection */
1836  if (ffurl_connect(rt->rtsp_hd, &options)) {
1838  err = AVERROR(EIO);
1839  goto fail;
1840  }
1841 
1842  /* POST requests */
1843  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1844  &s->interrupt_callback) < 0 ) {
1845  err = AVERROR(EIO);
1846  goto fail;
1847  }
1848 
1849  /* generate POST headers */
1850  snprintf(headers, sizeof(headers),
1851  "x-sessioncookie: %s\r\n"
1852  "Content-Type: application/x-rtsp-tunnelled\r\n"
1853  "Pragma: no-cache\r\n"
1854  "Cache-Control: no-cache\r\n"
1855  "Content-Length: 32767\r\n"
1856  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1857  sessioncookie);
1858  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1859  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1860  av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1861 
1862  /* Initialize the authentication state for the POST session. The HTTP
1863  * protocol implementation doesn't properly handle multi-pass
1864  * authentication for POST requests, since it would require one of
1865  * the following:
1866  * - implementing Expect: 100-continue, which many HTTP servers
1867  * don't support anyway, even less the RTSP servers that do HTTP
1868  * tunneling
1869  * - sending the whole POST data until getting a 401 reply specifying
1870  * what authentication method to use, then resending all that data
1871  * - waiting for potential 401 replies directly after sending the
1872  * POST header (waiting for some unspecified time)
1873  * Therefore, we copy the full auth state, which works for both basic
1874  * and digest. (For digest, we would have to synchronize the nonce
1875  * count variable between the two sessions, if we'd do more requests
1876  * with the original session, though.)
1877  */
1879 
1880  /* complete the connection */
1881  if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1883  err = AVERROR(EIO);
1884  goto fail;
1885  }
1887  } else {
1888  int ret;
1889  /* open the tcp connection */
1890  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1891  host, port,
1892  "?timeout=%"PRId64, rt->stimeout);
1893  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1894  &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1895  err = ret;
1896  goto fail;
1897  }
1898  rt->rtsp_hd_out = rt->rtsp_hd;
1899  }
1900  rt->seq = 0;
1901 
1902  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1903  if (tcp_fd < 0) {
1904  err = tcp_fd;
1905  goto fail;
1906  }
1907  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1908  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1909  NULL, 0, NI_NUMERICHOST);
1910  }
1911 
1912  /* request options supported by the server; this also detects server
1913  * type */
1914  if (rt->server_type != RTSP_SERVER_SATIP)
1916  for (;;) {
1917  cmd[0] = 0;
1918  if (rt->server_type == RTSP_SERVER_REAL)
1919  av_strlcat(cmd,
1920  /*
1921  * The following entries are required for proper
1922  * streaming from a Realmedia server. They are
1923  * interdependent in some way although we currently
1924  * don't quite understand how. Values were copied
1925  * from mplayer SVN r23589.
1926  * ClientChallenge is a 16-byte ID in hex
1927  * CompanyID is a 16-byte ID in base64
1928  */
1929  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1930  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1931  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1932  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1933  sizeof(cmd));
1934  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1935  if (reply->status_code != RTSP_STATUS_OK) {
1937  goto fail;
1938  }
1939 
1940  /* detect server type if not standard-compliant RTP */
1941  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1943  continue;
1944  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1946  } else if (rt->server_type == RTSP_SERVER_REAL)
1947  strcpy(real_challenge, reply->real_challenge);
1948  break;
1949  }
1950 
1951 #if CONFIG_RTSP_DEMUXER
1952  if (s->iformat) {
1953  if (rt->server_type == RTSP_SERVER_SATIP)
1954  err = init_satip_stream(s);
1955  else
1956  err = ff_rtsp_setup_input_streams(s, reply);
1957  } else
1958 #endif
1959  if (CONFIG_RTSP_MUXER)
1960  err = ff_rtsp_setup_output_streams(s, host);
1961  else
1962  av_assert0(0);
1963  if (err)
1964  goto fail;
1965 
1966  do {
1967  int lower_transport = ff_log2_tab[lower_transport_mask &
1968  ~(lower_transport_mask - 1)];
1969 
1970  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1971  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1972  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1973 
1974  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1975  rt->server_type == RTSP_SERVER_REAL ?
1976  real_challenge : NULL);
1977  if (err < 0)
1978  goto fail;
1979  lower_transport_mask &= ~(1 << lower_transport);
1980  if (lower_transport_mask == 0 && err == 1) {
1981  err = AVERROR(EPROTONOSUPPORT);
1982  goto fail;
1983  }
1984  } while (err);
1985 
1986  rt->lower_transport_mask = lower_transport_mask;
1987  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1988  rt->state = RTSP_STATE_IDLE;
1989  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1990  return 0;
1991  fail:
1994  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1995  char *new_url = av_strdup(reply->location);
1996  if (!new_url) {
1997  err = AVERROR(ENOMEM);
1998  goto fail2;
1999  }
2000  ff_format_set_url(s, new_url);
2001  rt->session_id[0] = '\0';
2002  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
2003  reply->status_code,
2004  s->url);
2005  goto redirect;
2006  }
2007  fail2:
2008  ff_network_close();
2009  return err;
2010 }
2011 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
2012 
2013 #if CONFIG_RTPDEC
2014 #if CONFIG_RTSP_DEMUXER
2015 static int parse_rtsp_message(AVFormatContext *s)
2016 {
2017  RTSPState *rt = s->priv_data;
2018  int ret;
2019 
2020  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
2021  if (rt->state == RTSP_STATE_STREAMING) {
2023  } else
2024  return AVERROR_EOF;
2025  } else {
2026  RTSPMessageHeader reply;
2027  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
2028  if (ret < 0)
2029  return ret;
2030  /* XXX: parse message */
2031  if (rt->state != RTSP_STATE_STREAMING)
2032  return 0;
2033  }
2034 
2035  return 0;
2036 }
2037 #endif
2038 
2039 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
2040  uint8_t *buf, int buf_size, int64_t wait_end)
2041 {
2042  RTSPState *rt = s->priv_data;
2043  RTSPStream *rtsp_st;
2044  int n, i, ret;
2045  struct pollfd *p = rt->p;
2046  int *fds = NULL, fdsnum, fdsidx;
2047  int64_t runs = rt->stimeout / POLLING_TIME / 1000;
2048 
2049  if (!p) {
2050  p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
2051  if (!p)
2052  return AVERROR(ENOMEM);
2053 
2054  if (rt->rtsp_hd) {
2055  p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2056  p[rt->max_p++].events = POLLIN;
2057  }
2058  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2059  rtsp_st = rt->rtsp_streams[i];
2060  if (rtsp_st->rtp_handle) {
2062  &fds, &fdsnum)) {
2063  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2064  return ret;
2065  }
2066  if (fdsnum != 2) {
2068  "Number of fds %d not supported\n", fdsnum);
2069  return AVERROR_INVALIDDATA;
2070  }
2071  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2072  p[rt->max_p].fd = fds[fdsidx];
2073  p[rt->max_p++].events = POLLIN;
2074  }
2075  av_freep(&fds);
2076  }
2077  }
2078  }
2079 
2080  for (;;) {
2081  if (ff_check_interrupt(&s->interrupt_callback))
2082  return AVERROR_EXIT;
2083  if (wait_end && wait_end - av_gettime_relative() < 0)
2084  return AVERROR(EAGAIN);
2085  n = poll(p, rt->max_p, POLLING_TIME);
2086  if (n > 0) {
2087  int j = rt->rtsp_hd ? 1 : 0;
2088  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2089  rtsp_st = rt->rtsp_streams[i];
2090  if (rtsp_st->rtp_handle) {
2091  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2092  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2093  if (ret > 0) {
2094  *prtsp_st = rtsp_st;
2095  return ret;
2096  }
2097  }
2098  j+=2;
2099  }
2100  }
2101 #if CONFIG_RTSP_DEMUXER
2102  if (rt->rtsp_hd && p[0].revents & POLLIN) {
2103  if ((ret = parse_rtsp_message(s)) < 0) {
2104  return ret;
2105  }
2106  }
2107 #endif
2108  } else if (n == 0 && rt->stimeout > 0 && --runs <= 0) {
2109  return AVERROR(ETIMEDOUT);
2110  } else if (n < 0 && errno != EINTR)
2111  return AVERROR(errno);
2112  }
2113 }
2114 
2115 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2116  const uint8_t *buf, int len)
2117 {
2118  RTSPState *rt = s->priv_data;
2119  int i;
2120  if (len < 0)
2121  return len;
2122  if (rt->nb_rtsp_streams == 1) {
2123  *rtsp_st = rt->rtsp_streams[0];
2124  return len;
2125  }
2126  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2127  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2128  int no_ssrc = 0;
2129  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2130  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2131  if (!rtpctx)
2132  continue;
2133  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2134  *rtsp_st = rt->rtsp_streams[i];
2135  return len;
2136  }
2137  if (!rtpctx->ssrc)
2138  no_ssrc = 1;
2139  }
2140  if (no_ssrc) {
2142  "Unable to pick stream for packet - SSRC not known for "
2143  "all streams\n");
2144  return AVERROR(EAGAIN);
2145  }
2146  } else {
2147  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2148  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2149  *rtsp_st = rt->rtsp_streams[i];
2150  return len;
2151  }
2152  }
2153  }
2154  }
2155  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2156  return AVERROR(EAGAIN);
2157 }
2158 
2159 static int read_packet(AVFormatContext *s,
2160  RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2161  int64_t wait_end)
2162 {
2163  RTSPState *rt = s->priv_data;
2164  int len;
2165 
2166  switch(rt->lower_transport) {
2167  default:
2168 #if CONFIG_RTSP_DEMUXER
2170  len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2171  break;
2172 #endif
2175  len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2176  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2177  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2178  break;
2180  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2181  wait_end && wait_end < av_gettime_relative())
2182  len = AVERROR(EAGAIN);
2183  else
2185  len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2186  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2187  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2188  break;
2189  }
2190 
2191  if (len == 0)
2192  return AVERROR_EOF;
2193 
2194  return len;
2195 }
2196 
2198 {
2199  RTSPState *rt = s->priv_data;
2200  int ret, len;
2201  RTSPStream *rtsp_st, *first_queue_st = NULL;
2202  int64_t wait_end = 0;
2203 
2204  if (rt->nb_byes == rt->nb_rtsp_streams)
2205  return AVERROR_EOF;
2206 
2207  /* get next frames from the same RTP packet */
2208  if (rt->cur_transport_priv) {
2209  if (rt->transport == RTSP_TRANSPORT_RDT) {
2211  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2213  } else if (CONFIG_RTPDEC && rt->ts) {
2215  if (ret >= 0) {
2216  rt->recvbuf_pos += ret;
2217  ret = rt->recvbuf_pos < rt->recvbuf_len;
2218  }
2219  } else
2220  ret = -1;
2221  if (ret == 0) {
2222  rt->cur_transport_priv = NULL;
2223  return 0;
2224  } else if (ret == 1) {
2225  return 0;
2226  } else
2227  rt->cur_transport_priv = NULL;
2228  }
2229 
2230 redo:
2231  if (rt->transport == RTSP_TRANSPORT_RTP) {
2232  int i;
2233  int64_t first_queue_time = 0;
2234  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2235  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2236  int64_t queue_time;
2237  if (!rtpctx)
2238  continue;
2239  queue_time = ff_rtp_queued_packet_time(rtpctx);
2240  if (queue_time && (queue_time - first_queue_time < 0 ||
2241  !first_queue_time)) {
2242  first_queue_time = queue_time;
2243  first_queue_st = rt->rtsp_streams[i];
2244  }
2245  }
2246  if (first_queue_time) {
2247  wait_end = first_queue_time + s->max_delay;
2248  } else {
2249  wait_end = 0;
2250  first_queue_st = NULL;
2251  }
2252  }
2253 
2254  /* read next RTP packet */
2255  if (!rt->recvbuf) {
2257  if (!rt->recvbuf)
2258  return AVERROR(ENOMEM);
2259  }
2260 
2261  len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2262  if (len == AVERROR(EAGAIN) && first_queue_st &&
2263  rt->transport == RTSP_TRANSPORT_RTP) {
2265  "max delay reached. need to consume packet\n");
2266  rtsp_st = first_queue_st;
2267  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2268  goto end;
2269  }
2270  if (len < 0)
2271  return len;
2272 
2273  if (rt->transport == RTSP_TRANSPORT_RDT) {
2274  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2275  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2276  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2277  if (rtsp_st->feedback) {
2278  AVIOContext *pb = NULL;
2280  pb = s->pb;
2281  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2282  }
2283  if (ret < 0) {
2284  /* Either bad packet, or a RTCP packet. Check if the
2285  * first_rtcp_ntp_time field was initialized. */
2286  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2287  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2288  /* first_rtcp_ntp_time has been initialized for this stream,
2289  * copy the same value to all other uninitialized streams,
2290  * in order to map their timestamp origin to the same ntp time
2291  * as this one. */
2292  int i;
2293  AVStream *st = NULL;
2294  if (rtsp_st->stream_index >= 0)
2295  st = s->streams[rtsp_st->stream_index];
2296  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2297  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2298  AVStream *st2 = NULL;
2299  if (rt->rtsp_streams[i]->stream_index >= 0)
2300  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2301  if (rtpctx2 && st && st2 &&
2302  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2303  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2304  rtpctx2->rtcp_ts_offset = av_rescale_q(
2305  rtpctx->rtcp_ts_offset, st->time_base,
2306  st2->time_base);
2307  }
2308  }
2309  // Make real NTP start time available in AVFormatContext
2310  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2311  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2312  if (rtpctx->st) {
2313  s->start_time_realtime -=
2314  av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
2315  }
2316  }
2317  }
2318  if (ret == -RTCP_BYE) {
2319  rt->nb_byes++;
2320 
2321  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2322  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2323 
2324  if (rt->nb_byes == rt->nb_rtsp_streams)
2325  return AVERROR_EOF;
2326  }
2327  }
2328  } else if (CONFIG_RTPDEC && rt->ts) {
2330  if (ret >= 0) {
2331  if (ret < len) {
2332  rt->recvbuf_len = len;
2333  rt->recvbuf_pos = ret;
2334  rt->cur_transport_priv = rt->ts;
2335  return 1;
2336  } else {
2337  ret = 0;
2338  }
2339  }
2340  } else {
2341  return AVERROR_INVALIDDATA;
2342  }
2343 end:
2344  if (ret < 0)
2345  goto redo;
2346  if (ret == 1)
2347  /* more packets may follow, so we save the RTP context */
2348  rt->cur_transport_priv = rtsp_st->transport_priv;
2349 
2350  return ret;
2351 }
2352 #endif /* CONFIG_RTPDEC */
2353 
2354 #if CONFIG_SDP_DEMUXER
2355 static int sdp_probe(const AVProbeData *p1)
2356 {
2357  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2358 
2359  /* we look for a line beginning "c=IN IP" */
2360  while (p < p_end && *p != '\0') {
2361  if (sizeof("c=IN IP") - 1 < p_end - p &&
2362  av_strstart(p, "c=IN IP", NULL))
2363  return AVPROBE_SCORE_EXTENSION;
2364 
2365  while (p < p_end - 1 && *p != '\n') p++;
2366  if (++p >= p_end)
2367  break;
2368  if (*p == '\r')
2369  p++;
2370  }
2371  return 0;
2372 }
2373 
2374 static void append_source_addrs(char *buf, int size, const char *name,
2375  int count, struct RTSPSource **addrs)
2376 {
2377  int i;
2378  if (!count)
2379  return;
2380  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2381  for (i = 1; i < count; i++)
2382  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2383 }
2384 
2385 static int sdp_read_header(AVFormatContext *s)
2386 {
2387  RTSPState *rt = s->priv_data;
2388  RTSPStream *rtsp_st;
2389  int i, err;
2390  char url[MAX_URL_SIZE];
2391  AVBPrint bp;
2392 
2393  if (!ff_network_init())
2394  return AVERROR(EIO);
2395 
2396  if (s->max_delay < 0) /* Not set by the caller */
2397  s->max_delay = DEFAULT_REORDERING_DELAY;
2398  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2400 
2401  /* read the whole sdp file */
2403  err = avio_read_to_bprint(s->pb, &bp, INT_MAX);
2404  if (err < 0 ) {
2405  ff_network_close();
2406  av_bprint_finalize(&bp, NULL);
2407  return err;
2408  }
2409  err = ff_sdp_parse(s, bp.str);
2410  av_bprint_finalize(&bp, NULL);
2411  if (err) goto fail;
2412 
2413  /* open each RTP stream */
2414  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2415  char namebuf[50];
2416  rtsp_st = rt->rtsp_streams[i];
2417 
2418  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2419  AVDictionary *opts = map_to_opts(rt);
2420  char buf[MAX_URL_SIZE];
2421  const char *p;
2422 
2423  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2424  sizeof(rtsp_st->sdp_ip),
2425  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2426  if (err) {
2427  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2428  err = AVERROR(EIO);
2429  av_dict_free(&opts);
2430  goto fail;
2431  }
2432  ff_url_join(url, sizeof(url), "rtp", NULL,
2433  namebuf, rtsp_st->sdp_port,
2434  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2435  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2436  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2437  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2438 
2439  p = strchr(s->url, '?');
2440  if (p && av_find_info_tag(buf, sizeof(buf), "localaddr", p))
2441  av_strlcatf(url, sizeof(url), "&localaddr=%s", buf);
2442  else if (rt->localaddr && rt->localaddr[0])
2443  av_strlcatf(url, sizeof(url), "&localaddr=%s", rt->localaddr);
2444  append_source_addrs(url, sizeof(url), "sources",
2445  rtsp_st->nb_include_source_addrs,
2446  rtsp_st->include_source_addrs);
2447  append_source_addrs(url, sizeof(url), "block",
2448  rtsp_st->nb_exclude_source_addrs,
2449  rtsp_st->exclude_source_addrs);
2450  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2451  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2452 
2453  av_dict_free(&opts);
2454 
2455  if (err < 0) {
2456  err = AVERROR_INVALIDDATA;
2457  goto fail;
2458  }
2459  }
2460  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2461  goto fail;
2462  }
2463  return 0;
2464 fail:
2466  ff_network_close();
2467  return err;
2468 }
2469 
2470 static int sdp_read_close(AVFormatContext *s)
2471 {
2473  ff_network_close();
2474  return 0;
2475 }
2476 
2477 static const AVClass sdp_demuxer_class = {
2478  .class_name = "SDP demuxer",
2479  .item_name = av_default_item_name,
2480  .option = sdp_options,
2481  .version = LIBAVUTIL_VERSION_INT,
2482 };
2483 
2484 const AVInputFormat ff_sdp_demuxer = {
2485  .name = "sdp",
2486  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2487  .priv_data_size = sizeof(RTSPState),
2488  .read_probe = sdp_probe,
2489  .read_header = sdp_read_header,
2491  .read_close = sdp_read_close,
2492  .priv_class = &sdp_demuxer_class,
2493 };
2494 #endif /* CONFIG_SDP_DEMUXER */
2495 
2496 #if CONFIG_RTP_DEMUXER
2497 static int rtp_probe(const AVProbeData *p)
2498 {
2499  if (av_strstart(p->filename, "rtp:", NULL))
2500  return AVPROBE_SCORE_MAX;
2501  return 0;
2502 }
2503 
2504 static int rtp_read_header(AVFormatContext *s)
2505 {
2506  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2507  char host[500], filters_buf[1000];
2508  int ret, port;
2509  URLContext* in = NULL;
2510  int payload_type;
2511  AVCodecParameters *par = NULL;
2512  struct sockaddr_storage addr;
2513  FFIOContext pb;
2514  socklen_t addrlen = sizeof(addr);
2515  RTSPState *rt = s->priv_data;
2516  const char *p;
2517  AVBPrint sdp;
2518  AVDictionary *opts = NULL;
2519 
2520  if (!ff_network_init())
2521  return AVERROR(EIO);
2522 
2523  opts = map_to_opts(rt);
2525  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2526  av_dict_free(&opts);
2527  if (ret)
2528  goto fail;
2529 
2530  while (1) {
2531  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2532  if (ret == AVERROR(EAGAIN))
2533  continue;
2534  if (ret < 0)
2535  goto fail;
2536  if (ret < 12) {
2537  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2538  continue;
2539  }
2540 
2541  if ((recvbuf[0] & 0xc0) != 0x80) {
2542  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2543  "received\n");
2544  continue;
2545  }
2546 
2547  if (RTP_PT_IS_RTCP(recvbuf[1]))
2548  continue;
2549 
2550  payload_type = recvbuf[1] & 0x7f;
2551  break;
2552  }
2553  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2554  ffurl_closep(&in);
2555 
2556  par = avcodec_parameters_alloc();
2557  if (!par) {
2558  ret = AVERROR(ENOMEM);
2559  goto fail;
2560  }
2561 
2562  if (ff_rtp_get_codec_info(par, payload_type)) {
2563  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2564  "without an SDP file describing it\n",
2565  payload_type);
2567  goto fail;
2568  }
2569  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2570  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2571  "properly you need an SDP file "
2572  "describing it\n");
2573  }
2574 
2575  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2576  NULL, 0, s->url);
2577 
2579  av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2580  addr.ss_family == AF_INET ? 4 : 6, host);
2581 
2582  p = strchr(s->url, '?');
2583  if (p) {
2584  static const char filters[][2][8] = { { "sources", "incl" },
2585  { "block", "excl" } };
2586  int i;
2587  char *q;
2588  for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2589  if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2590  q = filters_buf;
2591  while ((q = strchr(q, ',')) != NULL)
2592  *q = ' ';
2593  av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2594  filters[i][1],
2595  addr.ss_family == AF_INET ? 4 : 6, host,
2596  filters_buf);
2597  }
2598  }
2599  }
2600 
2601  av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2602  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2603  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2604  port, payload_type);
2605  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2606  if (!av_bprint_is_complete(&sdp))
2607  goto fail_nobuf;
2609 
2610  ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2611  s->pb = &pb.pub;
2612 
2613  /* if sdp_read_header() fails then following ff_network_close() cancels out */
2614  /* ff_network_init() at the start of this function. Otherwise it cancels out */
2615  /* ff_network_init() inside sdp_read_header() */
2616  ff_network_close();
2617 
2618  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2619 
2620  ret = sdp_read_header(s);
2621  s->pb = NULL;
2622  av_bprint_finalize(&sdp, NULL);
2623  return ret;
2624 
2625 fail_nobuf:
2626  ret = AVERROR(ENOMEM);
2627  av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2628  av_bprint_finalize(&sdp, NULL);
2629 fail:
2631  ffurl_closep(&in);
2632  ff_network_close();
2633  return ret;
2634 }
2635 
2636 static const AVClass rtp_demuxer_class = {
2637  .class_name = "RTP demuxer",
2638  .item_name = av_default_item_name,
2639  .option = rtp_options,
2640  .version = LIBAVUTIL_VERSION_INT,
2641 };
2642 
2643 const AVInputFormat ff_rtp_demuxer = {
2644  .name = "rtp",
2645  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2646  .priv_data_size = sizeof(RTSPState),
2647  .read_probe = rtp_probe,
2648  .read_header = rtp_read_header,
2650  .read_close = sdp_read_close,
2651  .flags = AVFMT_NOFILE,
2652  .priv_class = &rtp_demuxer_class,
2653 };
2654 #endif /* CONFIG_RTP_DEMUXER */
ff_rtsp_read_reply
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...
RTSPState::last_cmd_time
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:262
AVMEDIA_TYPE_SUBTITLE
@ AVMEDIA_TYPE_SUBTITLE
Definition: avutil.h:204
av_gettime_relative
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:186
AV_BPRINT_SIZE_UNLIMITED
#define AV_BPRINT_SIZE_UNLIMITED
ff_rtsp_close_streams
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:789
RTPDynamicProtocolHandler::init
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:127
name
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default minimum maximum flags name is the option name
Definition: writing_filters.txt:88
LIBAVFORMAT_IDENT
#define LIBAVFORMAT_IDENT
Definition: version.h:45
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
avformat_new_stream
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: options.c:237
av_find_info_tag
int av_find_info_tag(char *arg, int arg_size, const char *tag1, const char *info)
Attempt to find a specific tag in a URL.
Definition: parseutils.c:753
ff_rtp_demuxer
const AVInputFormat ff_rtp_demuxer
RTSPStream::transport_priv
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:445
AVCodecParameters::codec_type
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:57
ff_rtsp_send_cmd_with_content
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.
mpegts.h
RTPDynamicProtocolHandler::parse_sdp_a_line
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:129
av_bprint_finalize
int av_bprint_finalize(AVBPrint *buf, char **ret_str)
Finalize a print buffer.
Definition: bprint.c:235
RTSPStream::rtp_handle
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:444
ff_rtp_codec_id
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:146
ff_rtp_send_rtcp_feedback
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:468
av_bprint_init
void av_bprint_init(AVBPrint *buf, unsigned size_init, unsigned size_max)
Definition: bprint.c:69
RTSP_SERVER_RTP
@ RTSP_SERVER_RTP
Standards-compliant RTP-server.
Definition: rtsp.h:213
AVCodecParameters
This struct describes the properties of an encoded stream.
Definition: codec_par.h:53
RTSPMessageHeader::status_code
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:132
ff_rtsp_send_cmd
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.
RTSPState::control_transport
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:338
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
avpriv_mpegts_parse_packet
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:3404
AVIO_FLAG_READ_WRITE
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:624
RTSP_MODE_PLAIN
@ RTSP_MODE_PLAIN
Normal RTSP.
Definition: rtsp.h:70
parse_fmtp
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:132
rtpdec_formats.h
RTSP_TRANSPORT_RTP
@ RTSP_TRANSPORT_RTP
Standards-compliant RTP.
Definition: rtsp.h:59
RTSPTransportField::source
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:116
AV_TIME_BASE_Q
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:260
RTSPMessageHeader::range_end
int64_t range_end
Definition: rtsp.h:139
sdp_options
static const AVOption sdp_options[]
Definition: rtsp.c:106
RTSPState::get_parameter_supported
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:367
ff_rtsp_averror
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
av_strcasecmp
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:218
RTSP_DEFAULT_AUDIO_SAMPLERATE
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:77
RTSPStream::nb_include_source_addrs
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:460
RTSPTransportField::server_port_min
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:106
RTSPState::auth
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:280
RTSPStream::interleaved_min
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport.
Definition: rtsp.h:452
rtp_options
static const AVOption rtp_options[]
Definition: rtsp.c:117
RTSPState::recvbuf_pos
int recvbuf_pos
Definition: rtsp.h:329
AVOption
AVOption.
Definition: opt.h:251
RTSP_RTP_PORT_MIN
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:78
RTSPTransportField::lower_transport
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:122
RTSPState::rtp_port_min
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:395
AV_OPT_TYPE_DURATION
@ AV_OPT_TYPE_DURATION
Definition: opt.h:239
RTSP_LOWER_TRANSPORT_CUSTOM
@ RTSP_LOWER_TRANSPORT_CUSTOM
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag.
Definition: rtsp.h:47
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:353
avcodec_parameters_free
void avcodec_parameters_free(AVCodecParameters **ppar)
Free an AVCodecParameters instance and everything associated with it and write NULL to the supplied p...
Definition: codec_par.c:63
RTSPTransportField::interleaved_min
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:94
RTSPTransportField::interleaved_max
int interleaved_max
Definition: rtsp.h:94
RTSPStream
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:443
AV_LOG_VERBOSE
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:196
ff_rtsp_send_cmd_async
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.
RTSPState::real_challenge
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:277
mathematics.h
ff_rdt_calc_response_and_checksum
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
AVDictionary
Definition: dict.c:30
AVProbeData::buf_size
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:456
ff_network_close
void ff_network_close(void)
Definition: network.c:116
RTSPMessageHeader::nb_transports
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:135
RTSP_SERVER_REAL
@ RTSP_SERVER_REAL
Realmedia-style server.
Definition: rtsp.h:214
ff_http_auth_create_response
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:240
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:300
ff_rtp_check_and_send_back_rr
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:311
codec_type
enum AVMediaType codec_type
Definition: rtp.c:37
av_strlcatf
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:104
RTSPState::seek_timestamp
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:246
ENC
#define ENC
Definition: rtsp.c:64
os_support.h
FFIOContext
Definition: avio_internal.h:29
sockaddr_storage
Definition: network.h:111
ff_network_init
int ff_network_init(void)
Definition: network.c:58
ff_sdp_parse
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
FF_RTP_FLAG_OPTS
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
map_to_opts
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:127
RTSPState::pkt_size
int pkt_size
Definition: rtsp.h:419
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:30
RTSPStream::feedback
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:478
av_get_random_seed
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:120
av_memdup
void * av_memdup(const void *p, size_t size)
Duplicate a buffer with av_malloc().
Definition: mem.c:312
RTSPState::asf_ctx
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:314
AVPROBE_SCORE_MAX
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:465
avformat_close_input
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: demux.c:368
freeaddrinfo
#define freeaddrinfo
Definition: network.h:218
avpriv_set_pts_info
void avpriv_set_pts_info(AVStream *st, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: avformat.c:697
RTSP_FLAG_SATIP_RAW
#define RTSP_FLAG_SATIP_RAW
Export SAT>IP stream as raw MPEG-TS.
Definition: rtsp.h:431
ffstream
static av_always_inline FFStream * ffstream(AVStream *st)
Definition: internal.h:407
fail
#define fail()
Definition: checkasm.h:130
ff_rtp_get_local_rtp_port
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:539
rtpenc_chain.h
ff_rtp_set_remote_url
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:105
RTSPState::nb_rtsp_streams
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:230
ffurl_connect
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:166
sockaddr_storage::ss_family
uint16_t ss_family
Definition: network.h:116
read_close
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:143
ff_rdt_parse_packet
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:336
get_word
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:165
dynarray_add
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:418
OFFSET
#define OFFSET(x)
Definition: rtsp.c:62
RTSPMessageHeader::content_length
int content_length
length of the data following this header
Definition: rtsp.h:130
av_opt_set
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:487
RTPDynamicProtocolHandler::close
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:134
ff_rtp_parse_set_crypto
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:627
RTSP_TRANSPORT_RDT
@ RTSP_TRANSPORT_RDT
Realmedia Data Transport.
Definition: rtsp.h:60
ff_check_interrupt
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:658
RTSP_STATE_STREAMING
@ RTSP_STATE_STREAMING
initialized and sending/receiving data
Definition: rtsp.h:203
rtsp.h
ff_rtsp_setup_input_streams
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:612
RTSPState::lower_transport_mask
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:351
RTSP_MODE_TUNNEL
@ RTSP_MODE_TUNNEL
RTSP over HTTP (tunneling)
Definition: rtsp.h:71
SPACE_CHARS
#define SPACE_CHARS
Definition: dnn_backend_tf.c:361
RTSPStream::stream_index
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:448
RTSPTransportField::destination
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:115
URLContext::priv_data
void * priv_data
Definition: url.h:40
ff_rdt_parse_close
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
avassert.h
RTSP_LOWER_TRANSPORT_HTTPS
@ RTSP_LOWER_TRANSPORT_HTTPS
HTTPS tunneled.
Definition: rtsp.h:46
RTSPState::rtsp_hd_out
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling.
Definition: rtsp.h:335
AV_LOG_TRACE
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:206
avpriv_mpegts_parse_close
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:3429
pkt
AVPacket * pkt
Definition: movenc.c:59
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen.c:29
AVInputFormat
Definition: avformat.h:656
NTP_OFFSET
#define NTP_OFFSET
Definition: internal.h:466
RTSP_FLAG_LISTEN
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:426
RTSPState::reordering_queue_size
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:410
ffurl_open_whitelist
int ffurl_open_whitelist(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options, const char *whitelist, const char *blacklist, URLContext *parent)
Create an URLContext for accessing to the resource indicated by url, and open it.
Definition: avio.c:306
RTSPState::ts
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:328
AVCodecDescriptor
This struct describes the properties of a single codec described by an AVCodecID.
Definition: codec_desc.h:38
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:256
RTSPState::nb_byes
int nb_byes
Definition: rtsp.h:343
AI_NUMERICHOST
#define AI_NUMERICHOST
Definition: network.h:187
avio_read_to_bprint
int avio_read_to_bprint(AVIOContext *h, struct AVBPrint *pb, size_t max_size)
Read contents of h into print buffer, up to max_size bytes, or up to EOF.
Definition: aviobuf.c:1343
RTSPState::p
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:361
AVInputFormat::name
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:661
RTSPMessageHeader::location
char location[4096]
the "Location:" field.
Definition: rtsp.h:153
RTSPState::control_uri
char control_uri[MAX_URL_SIZE]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests,...
Definition: rtsp.h:324
AVProbeData::buf
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:455
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
s1
#define s1
Definition: regdef.h:38
AVProbeData::filename
const char * filename
Definition: avformat.h:454
RTSPMessageHeader::transports
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:143
RTSPMessageHeader::stream_id
char stream_id[64]
SAT>IP com.ses.streamID header.
Definition: rtsp.h:193
ff_url_join
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:38
filters
#define filters(fmt, type, inverse, clp, inverset, clip, one, clip_fn, packed)
Definition: af_crystalizer.c:55
AV_OPT_TYPE_INT64
@ AV_OPT_TYPE_INT64
Definition: opt.h:226
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AVIO_FLAG_WRITE
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:623
ff_rtsp_undo_setup
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:757
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:201
RTSPState::buffer_size
int buffer_size
Definition: rtsp.h:418
ff_rtsp_open_transport_ctx
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:825
RTSPTransportField::ttl
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:110
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
ff_rtsp_fetch_packet
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...
RTSPStream::dynamic_handler
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:471
av_stristart
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:48
RTSPMessageHeader::seq
int seq
sequence number
Definition: rtsp.h:145
key
const char * key
Definition: hwcontext_opencl.c:174
AVMEDIA_TYPE_DATA
@ AVMEDIA_TYPE_DATA
Opaque data information usually continuous.
Definition: avutil.h:203
RTSP_FLAG_OPTS
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:66
ff_rtp_handler_find_by_id
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with a matching codec ID.
Definition: rtpdec.c:172
handler
static void handler(vbi_event *ev, void *user_data)
Definition: libzvbi-teletextdec.c:508
RTP_REORDER_QUEUE_DEFAULT_SIZE
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:39
ff_http_auth_handle_header
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
FFStream::need_parsing
enum AVStreamParseType need_parsing
Definition: internal.h:380
ff_rtsp_setup_output_streams
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
RTSP_MEDIATYPE_OPTS
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:70
AVFormatContext
Format I/O context.
Definition: avformat.h:1213
internal.h
opts
AVDictionary * opts
Definition: movenc.c:50
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1108
RTP_PT_PRIVATE
#define RTP_PT_PRIVATE
Definition: rtp.h:79
RTSPState::session_id
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:252
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
RTSPMessageHeader::reason
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:183
read_header
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:529
RTSP_STATUS_OK
@ RTSP_STATUS_OK
Definition: rtspcodes.h:33
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
URLContext::protocol_whitelist
const char * protocol_whitelist
Definition: url.h:48
ff_rtsp_next_attr_and_value
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream::time_base
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
Definition: avformat.h:978
NULL
#define NULL
Definition: coverity.c:32
RECVBUF_SIZE
#define RECVBUF_SIZE
Definition: rtsp.c:59
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:64
RTSPState::rtsp_hd
URLContext * rtsp_hd
Definition: rtsp.h:227
read_probe
static int read_probe(const AVProbeData *pd)
Definition: jvdec.c:55
AVFMTCTX_NOHEADER
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1164
ff_rtp_queued_packet_time
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:825
ff_http_init_auth_state
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:188
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:237
AVFormatContext::pb
AVIOContext * pb
I/O context.
Definition: avformat.h:1255
RTSPState::default_lang
char default_lang[4]
Definition: rtsp.h:417
RTSPMessageHeader::real_challenge
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:156
get_word_sep
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:158
parseutils.h
AVProbeData
This structure contains the data a format has to probe a file.
Definition: avformat.h:453
av_channel_layout_default
void av_channel_layout_default(AVChannelLayout *ch_layout, int nb_channels)
Get the default channel layout for a given number of channels.
Definition: channel_layout.c:960
AVStream::metadata
AVDictionary * metadata
Definition: avformat.h:1019
SDP_MAX_SIZE
#define SDP_MAX_SIZE
Definition: rtsp.h:80
AV_CODEC_ID_MPEG2TS
@ AV_CODEC_ID_MPEG2TS
FAKE codec to indicate a raw MPEG-2 TS stream (only used by libavformat)
Definition: codec_id.h:570
DEC
#define DEC
Definition: rtsp.c:63
RTSP_MAX_TRANSPORTS
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:76
av_parse_time
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:589
RTPDemuxContext::rtcp_ts_offset
int64_t rtcp_ts_offset
Definition: rtpdec.h:180
time.h
RTSPState::state
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:238
RTSPState::recvbuf
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:346
RTSP_FLAG_PREFER_TCP
#define RTSP_FLAG_PREFER_TCP
Try RTP via TCP first if possible.
Definition: rtsp.h:430
AVCodecParameters::ch_layout
AVChannelLayout ch_layout
Audio only.
Definition: codec_par.h:212
RTSPStream::sdp_port
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:458
base64.h
AVPROBE_SCORE_EXTENSION
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:463
RTSPStream::exclude_source_addrs
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:463
AVCodecParameters::sample_rate
int sample_rate
Audio only.
Definition: codec_par.h:177
rtpdec.h
ff_log2_tab
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:23
av_bprint_is_complete
static int av_bprint_is_complete(const AVBPrint *buf)
Test if the print buffer is complete (not truncated).
Definition: bprint.h:185
get_sockaddr
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:196
ff_rtp_parse_close
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:957
av_strncasecmp
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:228
interleave
static void interleave(uint8_t *dst, uint8_t *src, int w, int h, int dst_linesize, int src_linesize, enum FilterMode mode, int swap)
Definition: vf_il.c:108
RTP_PT_IS_RTCP
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:112
options
const OptionDef options[]
RTSPSource
Definition: rtsp.h:433
ff_rtp_parse_open
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:573
NI_NUMERICHOST
#define NI_NUMERICHOST
Definition: network.h:195
AVIOContext
Bytestream IO Context.
Definition: avio.h:162
RTSPStream::dynamic_protocol_context
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:474
AVMediaType
AVMediaType
Definition: avutil.h:199
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:116
ff_rtsp_tcp_read_packet
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:783
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:290
RTSPState::rtsp_flags
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:385
ffurl_get_multi_file_handle
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:627
ff_rtsp_options
const AVOption ff_rtsp_options[]
Definition: rtsp.c:83
ff_rtsp_close_connections
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
RTSPState
Private data for the RTSP demuxer.
Definition: rtsp.h:225
RTSPStream::include_source_addrs
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:461
RTSPState::lower_transport
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:269
RTSPMessageHeader::range_start
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:139
RTSPState::recvbuf_len
int recvbuf_len
Definition: rtsp.h:330
RTSPState::last_reply
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:286
size
int size
Definition: twinvq_data.h:10344
RTSPTransportField::transport
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:119
RTPDemuxContext::first_rtcp_ntp_time
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:178
RTSPState::rtsp_streams
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:232
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
AV_RB32
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:96
avpriv_report_missing_feature
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
ff_rtsp_skip_packet
int ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
RTSPState::seq
int seq
RTSP command sequence number.
Definition: rtsp.h:248
COMMON_OPTS
#define COMMON_OPTS()
Definition: rtsp.c:77
RTSPStream::crypto_params
char crypto_params[100]
Definition: rtsp.h:484
AVFMT_NOFILE
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:470
AVMEDIA_TYPE_UNKNOWN
@ AVMEDIA_TYPE_UNKNOWN
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:200
header
static const uint8_t header[24]
Definition: sdr2.c:67
RTSPState::max_p
int max_p
Definition: rtsp.h:362
RTSPState::auth_state
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:283
ff_rtp_get_codec_info
int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:71
ff_rtp_handler_find_by_name
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with the specified name.
Definition: rtpdec.c:158
line
Definition: graph2dot.c:48
ff_rdt_parse_open
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, const RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
READ_PACKET_TIMEOUT_S
#define READ_PACKET_TIMEOUT_S
Definition: rtsp.c:58
ff_rtsp_parse_line
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
av_dict_free
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values.
Definition: dict.c:203
av_strstart
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:37
gai_strerror
#define gai_strerror
Definition: network.h:225
RTSPStream::nb_exclude_source_addrs
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:462
ff_real_parse_sdp_a_line
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:516
RTSPState::timeout
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:257
th
#define th
Definition: regdef.h:75
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:191
DEFAULT_REORDERING_DELAY
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:60
ffurl_alloc
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:293
getaddrinfo
#define getaddrinfo
Definition: network.h:217
avio_closep
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:1285
av_write_trailer
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1250
RTSP_SERVER_SATIP
@ RTSP_SERVER_SATIP
SAT>IP server.
Definition: rtsp.h:216
bprint.h
HTTP_AUTH_NONE
@ HTTP_AUTH_NONE
No authentication specified.
Definition: httpauth.h:29
AV_BASE64_SIZE
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes to a null-terminated string.
Definition: base64.h:66
RTSPState::media_type_mask
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:390
URLContext
Definition: url.h:37
AV_CODEC_ID_NONE
@ AV_CODEC_ID_NONE
Definition: codec_id.h:48
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
RTSPSource::addr
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:434
avio_internal.h
getnameinfo
#define getnameinfo
Definition: network.h:219
ff_rtp_parse_packet
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:944
RTPDemuxContext::ssrc
uint32_t ssrc
Definition: rtpdec.h:152
RTSPMessageHeader::timeout
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:173
ffio_init_context
void ffio_init_context(FFIOContext *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:81
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:31
RTSPState::need_subscription
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:295
RTCP_BYE
@ RTCP_BYE
Definition: rtp.h:102
rtpproto.h
RTPDemuxContext::st
AVStream * st
Definition: rtpdec.h:150
value
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
Definition: writing_filters.txt:86
ff_rtsp_tcp_write_packet
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
av_url_split
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:357
ff_rtsp_parse_streaming_commands
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:483
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
RTSPStream::ssrc
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:481
ff_sdp_demuxer
const AVInputFormat ff_sdp_demuxer
url.h
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:264
RTSP_LOWER_TRANSPORT_TCP
@ RTSP_LOWER_TRANSPORT_TCP
TCP; interleaved in RTSP.
Definition: rtsp.h:40
avcodec_parameters_alloc
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: codec_par.c:53
len
int len
Definition: vorbis_enc_data.h:426
profile
int profile
Definition: mxfenc.c:2005
rtpenc.h
RTSPStream::sdp_ttl
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:464
av_rescale
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:129
RTSP_FLAG_CUSTOM_IO
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:427
RTPDemuxContext
Definition: rtpdec.h:148
RTSPTransportField::client_port_min
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:102
version.h
RTSPState::rtp_port_max
int rtp_port_max
Definition: rtsp.h:395
ffurl_closep
int ffurl_closep(URLContext **hh)
Close the resource accessed by the URLContext h, and free the memory used by it.
Definition: avio.c:438
RTSP_LOWER_TRANSPORT_UDP_MULTICAST
@ RTSP_LOWER_TRANSPORT_UDP_MULTICAST
UDP/multicast.
Definition: rtsp.h:41
ffio_free_dyn_buf
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1544
AVStream::id
int id
Format-specific stream ID.
Definition: avformat.h:962
ret
ret
Definition: filter_design.txt:187
read_packet
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
AVStream
Stream structure.
Definition: avformat.h:948
RTSPState::cur_transport_priv
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:290
AVClass::class_name
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:71
av_strlcat
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:96
RTSPStream::sdp_payload_type
int sdp_payload_type
payload type
Definition: rtsp.h:465
ff_wms_parse_sdp_a_line
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:99
avformat.h
ff_rtp_chain_mux_open
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
av_bprintf
void av_bprintf(AVBPrint *buf, const char *fmt,...)
Definition: bprint.c:94
dict.h
network.h
RTP_MAX_PACKET_LENGTH
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:37
rtsp_parse_range_npt
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:174
RTSPStream::sdp_ip
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:459
HTTPAuthState::stale
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
RTSP_DEFAULT_PORT
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:74
AVStream::index
int index
stream index in AVFormatContext
Definition: avformat.h:956
RTSP_LOWER_TRANSPORT_HTTP
@ RTSP_LOWER_TRANSPORT_HTTP
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:43
RTSPTransportField
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:89
RTSPState::transport
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:265
random_seed.h
MAX_URL_SIZE
#define MAX_URL_SIZE
Definition: internal.h:32
RTPDemuxContext::base_timestamp
uint32_t base_timestamp
Definition: rtpdec.h:155
RTSPStream::control_url
char control_url[MAX_URL_SIZE]
url for this stream (from SDP)
Definition: rtsp.h:454
ffurl_read
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:401
addrinfo::ai_flags
int ai_flags
Definition: network.h:138
RTSPStream::interleaved_max
int interleaved_max
Definition: rtsp.h:452
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
RTSPState::localaddr
char * localaddr
Definition: rtsp.h:420
headers
FFmpeg currently uses a custom build this text attempts to document some of its obscure features and options Makefile the full command issued by make and its output will be shown on the screen DBG Preprocess x86 external assembler files to a dbg asm file in the object which then gets compiled Helps in developing those assembler files DESTDIR Destination directory for the install useful to prepare packages or install FFmpeg in cross environments GEN Set to ‘1’ to generate the missing or mismatched references Makefile builds all the libraries and the executables fate Run the fate test note that you must have installed it fate list List all fate regression test targets install Install headers
Definition: build_system.txt:34
avformat_free_context
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: avformat.c:95
ffurl_write
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:415
RTSP_SERVER_WMS
@ RTSP_SERVER_WMS
Windows Media server.
Definition: rtsp.h:215
avpriv_mpegts_parse_open
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:3382
RTSPMessageHeader
This describes the server response to each RTSP command.
Definition: rtsp.h:128
av_base64_encode
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:145
RTSP_TRANSPORT_RAW
@ RTSP_TRANSPORT_RAW
Raw data (over UDP)
Definition: rtsp.h:61
ff_mpegts_dynamic_handler
const RTPDynamicProtocolHandler ff_mpegts_dynamic_handler
Definition: rtpdec_mpegts.c:92
RTSP_RTP_PORT_MAX
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:79
RTSPState::stimeout
int64_t stimeout
timeout of socket i/o operations.
Definition: rtsp.h:405
av_dict_set_int
int av_dict_set_int(AVDictionary **pm, const char *key, int64_t value, int flags)
Convenience wrapper for av_dict_set that converts the value to a string and stores it.
Definition: dict.c:147
HTTPAuthType
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
AVIO_FLAG_READ
#define AVIO_FLAG_READ
read-only
Definition: avio.h:622
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:280
desc
const char * desc
Definition: libsvtav1.c:83
RTSP_STATE_IDLE
@ RTSP_STATE_IDLE
not initialized
Definition: rtsp.h:202
AVMEDIA_TYPE_VIDEO
@ AVMEDIA_TYPE_VIDEO
Definition: avutil.h:201
HTTPAuthState::auth_type
int auth_type
The currently chosen auth type.
Definition: httpauth.h:59
ffurl_read_complete
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:408
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
rdt.h
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:61
RTSP_LOWER_TRANSPORT_NB
@ RTSP_LOWER_TRANSPORT_NB
Definition: rtsp.h:42
ff_rtp_parse_set_dynamic_protocol
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, const RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:620
AVPacket
This structure stores compressed data.
Definition: packet.h:351
RTSPState::server_type
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:274
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:244
RTSPMessageHeader::content_type
char content_type[64]
Content type header.
Definition: rtsp.h:188
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
av_dict_set
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:70
POLLING_TIME
#define POLLING_TIME
Definition: network.h:249
AV_OPT_TYPE_FLAGS
@ AV_OPT_TYPE_FLAGS
Definition: opt.h:224
convert_header.str
string str
Definition: convert_header.py:20
RTSPState::accept_dynamic_rate
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:380
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
av_strlcpy
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:86
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
RTSPMessageHeader::session_id
char session_id[512]
the "Session:" field.
Definition: rtsp.h:149
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
ff_rtsp_make_setup_request
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.
ff_rtp_enc_name
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:135
AVERROR_EXIT
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:58
avcodec_descriptor_get
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
Definition: codec_desc.c:3545
RTSPMessageHeader::server
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:165
RTSPStream::crypto_suite
char crypto_suite[40]
Definition: rtsp.h:483
get_word_until_chars
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:139
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Definition: opt.h:229
addrinfo
Definition: network.h:137
http.h
ff_rtsp_connect
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
RTSPTransportField::port_min
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data.
Definition: rtsp.h:98
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
snprintf
#define snprintf
Definition: snprintf.h:34
ff_format_set_url
void ff_format_set_url(AVFormatContext *s, char *url)
Set AVFormatContext url field to the provided pointer.
Definition: avformat.c:772
RTSP_FLAG_FILTER_SRC
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port.
Definition: rtsp.h:423
RTSPMessageHeader::notice
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:178
avio_read_partial
int avio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:713
ffurl_get_file_handle
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:620
RTSPS_DEFAULT_PORT
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:75
RTSP_LOWER_TRANSPORT_UDP
@ RTSP_LOWER_TRANSPORT_UDP
UDP/unicast.
Definition: rtsp.h:39
RTPDynamicProtocolHandler
Definition: rtpdec.h:116
AV_RB16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:98
RTSPState::user_agent
char * user_agent
User-Agent string.
Definition: rtsp.h:415
RTSP_FLAG_RTCP_TO_SOURCE
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:428