FFmpeg  4.4.4
rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/bprint.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/parseutils.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/dict.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/time.h"
33 #include "avformat.h"
34 #include "avio_internal.h"
35 
36 #if HAVE_POLL_H
37 #include <poll.h>
38 #endif
39 #include "internal.h"
40 #include "network.h"
41 #include "os_support.h"
42 #include "http.h"
43 #include "rtsp.h"
44 
45 #include "rtpdec.h"
46 #include "rtpproto.h"
47 #include "rdt.h"
48 #include "rtpdec_formats.h"
49 #include "rtpenc_chain.h"
50 #include "url.h"
51 #include "rtpenc.h"
52 #include "mpegts.h"
53 
54 /* Default timeout values for read packet in seconds */
55 #define READ_PACKET_TIMEOUT_S 10
56 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
57 #define DEFAULT_REORDERING_DELAY 100000
58 
59 #define OFFSET(x) offsetof(RTSPState, x)
60 #define DEC AV_OPT_FLAG_DECODING_PARAM
61 #define ENC AV_OPT_FLAG_ENCODING_PARAM
62 
63 #define RTSP_FLAG_OPTS(name, longname) \
64  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
65  { "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" }
66 
67 #define RTSP_MEDIATYPE_OPTS(name, longname) \
68  { 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" }, \
69  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
70  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
71  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
72  { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
73 
74 #define COMMON_OPTS() \
75  { "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 }, \
76  { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
77  { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
78 
79 
81  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
82  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88  { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
89  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
90  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
91  { "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" },
92  { "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" },
93  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96  { "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 },
97 #if FF_API_OLD_RTSP_OPTIONS
98  { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC|AV_OPT_FLAG_DEPRECATED },
99  { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
100 #else
101  { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
102 #endif
103  COMMON_OPTS(),
104  { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
105 #if FF_API_OLD_RTSP_OPTIONS
106  { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC|AV_OPT_FLAG_DEPRECATED },
107 #endif
108  { NULL },
109 };
110 
111 static const AVOption sdp_options[] = {
112  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
113  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
114  { "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" },
115  { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
116  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
117  COMMON_OPTS(),
118  { NULL },
119 };
120 
121 static const AVOption rtp_options[] = {
122  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
123  { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
124  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
125  COMMON_OPTS(),
126  { NULL },
127 };
128 
129 
131 {
133  char buf[256];
134 
135  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
136  av_dict_set(&opts, "buffer_size", buf, 0);
137  snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
138  av_dict_set(&opts, "pkt_size", buf, 0);
139 
140  return opts;
141 }
142 
143 static void get_word_until_chars(char *buf, int buf_size,
144  const char *sep, const char **pp)
145 {
146  const char *p;
147  char *q;
148 
149  p = *pp;
150  p += strspn(p, SPACE_CHARS);
151  q = buf;
152  while (!strchr(sep, *p) && *p != '\0') {
153  if ((q - buf) < buf_size - 1)
154  *q++ = *p;
155  p++;
156  }
157  if (buf_size > 0)
158  *q = '\0';
159  *pp = p;
160 }
161 
162 static void get_word_sep(char *buf, int buf_size, const char *sep,
163  const char **pp)
164 {
165  if (**pp == '/') (*pp)++;
166  get_word_until_chars(buf, buf_size, sep, pp);
167 }
168 
169 static void get_word(char *buf, int buf_size, const char **pp)
170 {
171  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
172 }
173 
174 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
175  * and end time.
176  * Used for seeking in the rtp stream.
177  */
178 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
179 {
180  char buf[256];
181 
182  p += strspn(p, SPACE_CHARS);
183  if (!av_stristart(p, "npt=", &p))
184  return;
185 
186  *start = AV_NOPTS_VALUE;
187  *end = AV_NOPTS_VALUE;
188 
189  get_word_sep(buf, sizeof(buf), "-", &p);
190  if (av_parse_time(start, buf, 1) < 0)
191  return;
192  if (*p == '-') {
193  p++;
194  get_word_sep(buf, sizeof(buf), "-", &p);
195  if (av_parse_time(end, buf, 1) < 0)
196  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
197  }
198 }
199 
201  const char *buf, struct sockaddr_storage *sock)
202 {
203  struct addrinfo hints = { 0 }, *ai = NULL;
204  int ret;
205 
206  hints.ai_flags = AI_NUMERICHOST;
207  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
208  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
209  buf,
210  gai_strerror(ret));
211  return -1;
212  }
213  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
214  freeaddrinfo(ai);
215  return 0;
216 }
217 
218 #if CONFIG_RTPDEC
219 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
220  RTSPStream *rtsp_st, AVStream *st)
221 {
222  AVCodecParameters *par = st ? st->codecpar : NULL;
223  if (!handler)
224  return;
225  if (par)
226  par->codec_id = handler->codec_id;
227  rtsp_st->dynamic_handler = handler;
228  if (st)
229  st->need_parsing = handler->need_parsing;
230  if (handler->priv_data_size) {
231  rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
232  if (!rtsp_st->dynamic_protocol_context)
233  rtsp_st->dynamic_handler = NULL;
234  }
235 }
236 
237 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
238  AVStream *st)
239 {
240  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
241  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
242  rtsp_st->dynamic_protocol_context);
243  if (ret < 0) {
244  if (rtsp_st->dynamic_protocol_context) {
245  if (rtsp_st->dynamic_handler->close)
246  rtsp_st->dynamic_handler->close(
247  rtsp_st->dynamic_protocol_context);
249  }
250  rtsp_st->dynamic_protocol_context = NULL;
251  rtsp_st->dynamic_handler = NULL;
252  }
253  }
254 }
255 
256 static int init_satip_stream(AVFormatContext *s)
257 {
258  RTSPState *rt = s->priv_data;
259  RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream));
260  if (!rtsp_st)
261  return AVERROR(ENOMEM);
263  &rt->nb_rtsp_streams, rtsp_st);
264 
265  rtsp_st->sdp_payload_type = 33; // MP2T
266  av_strlcpy(rtsp_st->control_url,
267  rt->control_uri, sizeof(rtsp_st->control_url));
268 
269  if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) {
271  if (!st)
272  return AVERROR(ENOMEM);
273  st->id = rt->nb_rtsp_streams - 1;
274  rtsp_st->stream_index = st->index;
277  } else {
278  rtsp_st->stream_index = -1;
279  init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
280  finalize_rtp_handler_init(s, rtsp_st, NULL);
281  }
282  return 0;
283 }
284 
285 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
286 static int sdp_parse_rtpmap(AVFormatContext *s,
287  AVStream *st, RTSPStream *rtsp_st,
288  int payload_type, const char *p)
289 {
290  AVCodecParameters *par = st->codecpar;
291  char buf[256];
292  int i;
293  const AVCodecDescriptor *desc;
294  const char *c_name;
295 
296  /* See if we can handle this kind of payload.
297  * The space should normally not be there but some Real streams or
298  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
299  * have a trailing space. */
300  get_word_sep(buf, sizeof(buf), "/ ", &p);
301  if (payload_type < RTP_PT_PRIVATE) {
302  /* We are in a standard case
303  * (from http://www.iana.org/assignments/rtp-parameters). */
304  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
305  }
306 
307  if (par->codec_id == AV_CODEC_ID_NONE) {
310  init_rtp_handler(handler, rtsp_st, st);
311  /* If no dynamic handler was found, check with the list of standard
312  * allocated types, if such a stream for some reason happens to
313  * use a private payload type. This isn't handled in rtpdec.c, since
314  * the format name from the rtpmap line never is passed into rtpdec. */
315  if (!rtsp_st->dynamic_handler)
316  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
317  }
318 
320  if (desc && desc->name)
321  c_name = desc->name;
322  else
323  c_name = "(null)";
324 
325  get_word_sep(buf, sizeof(buf), "/", &p);
326  i = atoi(buf);
327  switch (par->codec_type) {
328  case AVMEDIA_TYPE_AUDIO:
329  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
332  if (i > 0) {
333  par->sample_rate = i;
334  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
335  get_word_sep(buf, sizeof(buf), "/", &p);
336  i = atoi(buf);
337  if (i > 0)
338  par->channels = i;
339  }
340  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
341  par->sample_rate);
342  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
343  par->channels);
344  break;
345  case AVMEDIA_TYPE_VIDEO:
346  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
347  if (i > 0)
348  avpriv_set_pts_info(st, 32, 1, i);
349  break;
350  default:
351  break;
352  }
353  finalize_rtp_handler_init(s, rtsp_st, st);
354  return 0;
355 }
356 
357 /* parse the attribute line from the fmtp a line of an sdp response. This
358  * is broken out as a function because it is used in rtp_h264.c, which is
359  * forthcoming. */
360 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
361  char *value, int value_size)
362 {
363  *p += strspn(*p, SPACE_CHARS);
364  if (**p) {
365  get_word_sep(attr, attr_size, "=", p);
366  if (**p == '=')
367  (*p)++;
368  get_word_sep(value, value_size, ";", p);
369  if (**p == ';')
370  (*p)++;
371  return 1;
372  }
373  return 0;
374 }
375 
376 typedef struct SDPParseState {
377  /* SDP only */
378  struct sockaddr_storage default_ip;
379  int default_ttl;
380  int skip_media; ///< set if an unknown m= line occurs
381  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
382  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
383  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
384  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
385  int seen_rtpmap;
386  int seen_fmtp;
387  char delayed_fmtp[2048];
388 } SDPParseState;
389 
390 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
391  struct RTSPSource ***dest, int *dest_count)
392 {
393  RTSPSource *rtsp_src, *rtsp_src2;
394  int i;
395  for (i = 0; i < count; i++) {
396  rtsp_src = addrs[i];
397  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
398  if (!rtsp_src2)
399  continue;
400  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
401  dynarray_add(dest, dest_count, rtsp_src2);
402  }
403 }
404 
405 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
406  int payload_type, const char *line)
407 {
408  int i;
409 
410  for (i = 0; i < rt->nb_rtsp_streams; i++) {
411  RTSPStream *rtsp_st = rt->rtsp_streams[i];
412  if (rtsp_st->sdp_payload_type == payload_type &&
413  rtsp_st->dynamic_handler &&
414  rtsp_st->dynamic_handler->parse_sdp_a_line) {
415  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
416  rtsp_st->dynamic_protocol_context, line);
417  }
418  }
419 }
420 
421 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
422  int letter, const char *buf)
423 {
424  RTSPState *rt = s->priv_data;
425  char buf1[64], st_type[64];
426  const char *p;
427  enum AVMediaType codec_type;
428  int payload_type;
429  AVStream *st;
430  RTSPStream *rtsp_st;
431  RTSPSource *rtsp_src;
432  struct sockaddr_storage sdp_ip;
433  int ttl;
434 
435  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
436 
437  p = buf;
438  if (s1->skip_media && letter != 'm')
439  return;
440  switch (letter) {
441  case 'c':
442  get_word(buf1, sizeof(buf1), &p);
443  if (strcmp(buf1, "IN") != 0)
444  return;
445  get_word(buf1, sizeof(buf1), &p);
446  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
447  return;
448  get_word_sep(buf1, sizeof(buf1), "/", &p);
449  if (get_sockaddr(s, buf1, &sdp_ip))
450  return;
451  ttl = 16;
452  if (*p == '/') {
453  p++;
454  get_word_sep(buf1, sizeof(buf1), "/", &p);
455  ttl = atoi(buf1);
456  }
457  if (s->nb_streams == 0) {
458  s1->default_ip = sdp_ip;
459  s1->default_ttl = ttl;
460  } else {
461  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
462  rtsp_st->sdp_ip = sdp_ip;
463  rtsp_st->sdp_ttl = ttl;
464  }
465  break;
466  case 's':
467  av_dict_set(&s->metadata, "title", p, 0);
468  break;
469  case 'i':
470  if (s->nb_streams == 0) {
471  av_dict_set(&s->metadata, "comment", p, 0);
472  break;
473  }
474  break;
475  case 'm':
476  /* new stream */
477  s1->skip_media = 0;
478  s1->seen_fmtp = 0;
479  s1->seen_rtpmap = 0;
481  get_word(st_type, sizeof(st_type), &p);
482  if (!strcmp(st_type, "audio")) {
484  } else if (!strcmp(st_type, "video")) {
486  } else if (!strcmp(st_type, "application")) {
488  } else if (!strcmp(st_type, "text")) {
490  }
492  !(rt->media_type_mask & (1 << codec_type)) ||
493  rt->nb_rtsp_streams >= s->max_streams
494  ) {
495  s1->skip_media = 1;
496  return;
497  }
498  rtsp_st = av_mallocz(sizeof(RTSPStream));
499  if (!rtsp_st)
500  return;
501  rtsp_st->stream_index = -1;
502  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
503 
504  rtsp_st->sdp_ip = s1->default_ip;
505  rtsp_st->sdp_ttl = s1->default_ttl;
506 
507  copy_default_source_addrs(s1->default_include_source_addrs,
508  s1->nb_default_include_source_addrs,
509  &rtsp_st->include_source_addrs,
510  &rtsp_st->nb_include_source_addrs);
511  copy_default_source_addrs(s1->default_exclude_source_addrs,
512  s1->nb_default_exclude_source_addrs,
513  &rtsp_st->exclude_source_addrs,
514  &rtsp_st->nb_exclude_source_addrs);
515 
516  get_word(buf1, sizeof(buf1), &p); /* port */
517  rtsp_st->sdp_port = atoi(buf1);
518 
519  get_word(buf1, sizeof(buf1), &p); /* protocol */
520  if (!strcmp(buf1, "udp"))
522  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
523  rtsp_st->feedback = 1;
524 
525  /* XXX: handle list of formats */
526  get_word(buf1, sizeof(buf1), &p); /* format list */
527  rtsp_st->sdp_payload_type = atoi(buf1);
528 
529  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
530  /* no corresponding stream */
531  if (rt->transport == RTSP_TRANSPORT_RAW) {
532  if (CONFIG_RTPDEC && !rt->ts)
534  } else {
538  init_rtp_handler(handler, rtsp_st, NULL);
539  finalize_rtp_handler_init(s, rtsp_st, NULL);
540  }
541  } else if (rt->server_type == RTSP_SERVER_WMS &&
543  /* RTX stream, a stream that carries all the other actual
544  * audio/video streams. Don't expose this to the callers. */
545  } else {
546  st = avformat_new_stream(s, NULL);
547  if (!st)
548  return;
549  st->id = rt->nb_rtsp_streams - 1;
550  rtsp_st->stream_index = st->index;
552  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
554  /* if standard payload type, we can find the codec right now */
556  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
557  st->codecpar->sample_rate > 0)
558  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
559  /* Even static payload types may need a custom depacketizer */
561  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
562  init_rtp_handler(handler, rtsp_st, st);
563  finalize_rtp_handler_init(s, rtsp_st, st);
564  }
565  if (rt->default_lang[0])
566  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
567  }
568  /* put a default control url */
569  av_strlcpy(rtsp_st->control_url, rt->control_uri,
570  sizeof(rtsp_st->control_url));
571  break;
572  case 'a':
573  if (av_strstart(p, "control:", &p)) {
574  if (rt->nb_rtsp_streams == 0) {
575  if (!strncmp(p, "rtsp://", 7))
576  av_strlcpy(rt->control_uri, p,
577  sizeof(rt->control_uri));
578  } else {
579  char proto[32];
580  /* get the control url */
581  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
582 
583  /* XXX: may need to add full url resolution */
584  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
585  NULL, NULL, 0, p);
586  if (proto[0] == '\0') {
587  /* relative control URL */
588  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
589  av_strlcat(rtsp_st->control_url, "/",
590  sizeof(rtsp_st->control_url));
591  av_strlcat(rtsp_st->control_url, p,
592  sizeof(rtsp_st->control_url));
593  } else
594  av_strlcpy(rtsp_st->control_url, p,
595  sizeof(rtsp_st->control_url));
596  }
597  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
598  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
599  get_word(buf1, sizeof(buf1), &p);
600  payload_type = atoi(buf1);
601  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
602  if (rtsp_st->stream_index >= 0) {
603  st = s->streams[rtsp_st->stream_index];
604  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
605  }
606  s1->seen_rtpmap = 1;
607  if (s1->seen_fmtp) {
608  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
609  }
610  } else if (av_strstart(p, "fmtp:", &p) ||
611  av_strstart(p, "framesize:", &p)) {
612  // let dynamic protocol handlers have a stab at the line.
613  get_word(buf1, sizeof(buf1), &p);
614  payload_type = atoi(buf1);
615  if (s1->seen_rtpmap) {
616  parse_fmtp(s, rt, payload_type, buf);
617  } else {
618  s1->seen_fmtp = 1;
619  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
620  }
621  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
622  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
623  get_word(buf1, sizeof(buf1), &p);
624  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
625  } else if (av_strstart(p, "range:", &p)) {
626  int64_t start, end;
627 
628  // this is so that seeking on a streamed file can work.
629  rtsp_parse_range_npt(p, &start, &end);
630  s->start_time = start;
631  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
632  s->duration = (end == AV_NOPTS_VALUE) ?
633  AV_NOPTS_VALUE : end - start;
634  } else if (av_strstart(p, "lang:", &p)) {
635  if (s->nb_streams > 0) {
636  get_word(buf1, sizeof(buf1), &p);
637  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
638  if (rtsp_st->stream_index >= 0) {
639  st = s->streams[rtsp_st->stream_index];
640  av_dict_set(&st->metadata, "language", buf1, 0);
641  }
642  } else
643  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
644  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
645  if (atoi(p) == 1)
647  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
648  s->nb_streams > 0) {
649  st = s->streams[s->nb_streams - 1];
650  st->codecpar->sample_rate = atoi(p);
651  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
652  // RFC 4568
653  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
654  get_word(buf1, sizeof(buf1), &p); // ignore tag
655  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
656  p += strspn(p, SPACE_CHARS);
657  if (av_strstart(p, "inline:", &p))
658  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
659  } else if (av_strstart(p, "source-filter:", &p)) {
660  int exclude = 0;
661  get_word(buf1, sizeof(buf1), &p);
662  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
663  return;
664  exclude = !strcmp(buf1, "excl");
665 
666  get_word(buf1, sizeof(buf1), &p);
667  if (strcmp(buf1, "IN") != 0)
668  return;
669  get_word(buf1, sizeof(buf1), &p);
670  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
671  return;
672  // not checking that the destination address actually matches or is wildcard
673  get_word(buf1, sizeof(buf1), &p);
674 
675  while (*p != '\0') {
676  rtsp_src = av_mallocz(sizeof(*rtsp_src));
677  if (!rtsp_src)
678  return;
679  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
680  if (exclude) {
681  if (s->nb_streams == 0) {
682  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
683  } else {
684  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
685  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
686  }
687  } else {
688  if (s->nb_streams == 0) {
689  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
690  } else {
691  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
692  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
693  }
694  }
695  }
696  } else {
697  if (rt->server_type == RTSP_SERVER_WMS)
699  if (s->nb_streams > 0) {
700  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
701 
702  if (rt->server_type == RTSP_SERVER_REAL)
704 
705  if (rtsp_st->dynamic_handler &&
708  rtsp_st->stream_index,
709  rtsp_st->dynamic_protocol_context, buf);
710  }
711  }
712  break;
713  }
714 }
715 
716 int ff_sdp_parse(AVFormatContext *s, const char *content)
717 {
718  const char *p;
719  int letter, i;
720  char buf[SDP_MAX_SIZE], *q;
721  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
722 
723  p = content;
724  for (;;) {
725  p += strspn(p, SPACE_CHARS);
726  letter = *p;
727  if (letter == '\0')
728  break;
729  p++;
730  if (*p != '=')
731  goto next_line;
732  p++;
733  /* get the content */
734  q = buf;
735  while (*p != '\n' && *p != '\r' && *p != '\0') {
736  if ((q - buf) < sizeof(buf) - 1)
737  *q++ = *p;
738  p++;
739  }
740  *q = '\0';
741  sdp_parse_line(s, s1, letter, buf);
742  next_line:
743  while (*p != '\n' && *p != '\0')
744  p++;
745  if (*p == '\n')
746  p++;
747  }
748 
749  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
750  av_freep(&s1->default_include_source_addrs[i]);
751  av_freep(&s1->default_include_source_addrs);
752  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
753  av_freep(&s1->default_exclude_source_addrs[i]);
754  av_freep(&s1->default_exclude_source_addrs);
755 
756  return 0;
757 }
758 #endif /* CONFIG_RTPDEC */
759 
760 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
761 {
762  RTSPState *rt = s->priv_data;
763  int i;
764 
765  for (i = 0; i < rt->nb_rtsp_streams; i++) {
766  RTSPStream *rtsp_st = rt->rtsp_streams[i];
767  if (!rtsp_st)
768  continue;
769  if (rtsp_st->transport_priv) {
770  if (s->oformat) {
771  AVFormatContext *rtpctx = rtsp_st->transport_priv;
772  av_write_trailer(rtpctx);
774  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
775  ff_rtsp_tcp_write_packet(s, rtsp_st);
776  ffio_free_dyn_buf(&rtpctx->pb);
777  } else {
778  avio_closep(&rtpctx->pb);
779  }
780  avformat_free_context(rtpctx);
781  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
783  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
785  }
786  rtsp_st->transport_priv = NULL;
787  ffurl_closep(&rtsp_st->rtp_handle);
788  }
789 }
790 
791 /* close and free RTSP streams */
793 {
794  RTSPState *rt = s->priv_data;
795  int i, j;
796  RTSPStream *rtsp_st;
797 
798  ff_rtsp_undo_setup(s, 0);
799  for (i = 0; i < rt->nb_rtsp_streams; i++) {
800  rtsp_st = rt->rtsp_streams[i];
801  if (rtsp_st) {
802  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
803  if (rtsp_st->dynamic_handler->close)
804  rtsp_st->dynamic_handler->close(
805  rtsp_st->dynamic_protocol_context);
807  }
808  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
809  av_freep(&rtsp_st->include_source_addrs[j]);
810  av_freep(&rtsp_st->include_source_addrs);
811  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
812  av_freep(&rtsp_st->exclude_source_addrs[j]);
813  av_freep(&rtsp_st->exclude_source_addrs);
814 
815  av_freep(&rtsp_st);
816  }
817  }
818  av_freep(&rt->rtsp_streams);
819  if (rt->asf_ctx) {
821  }
822  if (CONFIG_RTPDEC && rt->ts)
824  av_freep(&rt->p);
825  av_freep(&rt->recvbuf);
826 }
827 
829 {
830  RTSPState *rt = s->priv_data;
831  AVStream *st = NULL;
832  int reordering_queue_size = rt->reordering_queue_size;
833  if (reordering_queue_size < 0) {
834  if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
835  reordering_queue_size = 0;
836  else
837  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
838  }
839 
840  /* open the RTP context */
841  if (rtsp_st->stream_index >= 0)
842  st = s->streams[rtsp_st->stream_index];
843  if (!st)
844  s->ctx_flags |= AVFMTCTX_NOHEADER;
845 
846  if (CONFIG_RTSP_MUXER && s->oformat && st) {
847  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
848  s, st, rtsp_st->rtp_handle,
850  rtsp_st->stream_index);
851  /* Ownership of rtp_handle is passed to the rtp mux context */
852  rtsp_st->rtp_handle = NULL;
853  if (ret < 0)
854  return ret;
855  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
856  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
857  return 0; // Don't need to open any parser here
858  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
859  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
860  rtsp_st->dynamic_protocol_context,
861  rtsp_st->dynamic_handler);
862  else if (CONFIG_RTPDEC)
863  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
864  rtsp_st->sdp_payload_type,
865  reordering_queue_size);
866 
867  if (!rtsp_st->transport_priv) {
868  return AVERROR(ENOMEM);
869  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
870  s->iformat) {
871  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
872  rtpctx->ssrc = rtsp_st->ssrc;
873  if (rtsp_st->dynamic_handler) {
875  rtsp_st->dynamic_protocol_context,
876  rtsp_st->dynamic_handler);
877  }
878  if (rtsp_st->crypto_suite[0])
880  rtsp_st->crypto_suite,
881  rtsp_st->crypto_params);
882  }
883 
884  return 0;
885 }
886 
887 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
888 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
889 {
890  const char *q;
891  char *p;
892  int v;
893 
894  q = *pp;
895  q += strspn(q, SPACE_CHARS);
896  v = strtol(q, &p, 10);
897  if (*p == '-') {
898  p++;
899  *min_ptr = v;
900  v = strtol(p, &p, 10);
901  *max_ptr = v;
902  } else {
903  *min_ptr = v;
904  *max_ptr = v;
905  }
906  *pp = p;
907 }
908 
909 /* XXX: only one transport specification is parsed */
910 static void rtsp_parse_transport(AVFormatContext *s,
911  RTSPMessageHeader *reply, const char *p)
912 {
913  char transport_protocol[16];
914  char profile[16];
915  char lower_transport[16];
916  char parameter[16];
918  char buf[256];
919 
920  reply->nb_transports = 0;
921 
922  for (;;) {
923  p += strspn(p, SPACE_CHARS);
924  if (*p == '\0')
925  break;
926 
927  th = &reply->transports[reply->nb_transports];
928 
929  get_word_sep(transport_protocol, sizeof(transport_protocol),
930  "/", &p);
931  if (!av_strcasecmp (transport_protocol, "rtp")) {
932  get_word_sep(profile, sizeof(profile), "/;,", &p);
933  lower_transport[0] = '\0';
934  /* rtp/avp/<protocol> */
935  if (*p == '/') {
936  get_word_sep(lower_transport, sizeof(lower_transport),
937  ";,", &p);
938  }
939  th->transport = RTSP_TRANSPORT_RTP;
940  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
941  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
942  /* x-pn-tng/<protocol> */
943  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
944  profile[0] = '\0';
945  th->transport = RTSP_TRANSPORT_RDT;
946  } else if (!av_strcasecmp(transport_protocol, "raw")) {
947  get_word_sep(profile, sizeof(profile), "/;,", &p);
948  lower_transport[0] = '\0';
949  /* raw/raw/<protocol> */
950  if (*p == '/') {
951  get_word_sep(lower_transport, sizeof(lower_transport),
952  ";,", &p);
953  }
954  th->transport = RTSP_TRANSPORT_RAW;
955  } else {
956  break;
957  }
958  if (!av_strcasecmp(lower_transport, "TCP"))
959  th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
960  else
961  th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
962 
963  if (*p == ';')
964  p++;
965  /* get each parameter */
966  while (*p != '\0' && *p != ',') {
967  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
968  if (!strcmp(parameter, "port")) {
969  if (*p == '=') {
970  p++;
971  rtsp_parse_range(&th->port_min, &th->port_max, &p);
972  }
973  } else if (!strcmp(parameter, "client_port")) {
974  if (*p == '=') {
975  p++;
976  rtsp_parse_range(&th->client_port_min,
977  &th->client_port_max, &p);
978  }
979  } else if (!strcmp(parameter, "server_port")) {
980  if (*p == '=') {
981  p++;
982  rtsp_parse_range(&th->server_port_min,
983  &th->server_port_max, &p);
984  }
985  } else if (!strcmp(parameter, "interleaved")) {
986  if (*p == '=') {
987  p++;
988  rtsp_parse_range(&th->interleaved_min,
989  &th->interleaved_max, &p);
990  }
991  } else if (!strcmp(parameter, "multicast")) {
992  if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
993  th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
994  } else if (!strcmp(parameter, "ttl")) {
995  if (*p == '=') {
996  char *end;
997  p++;
998  th->ttl = strtol(p, &end, 10);
999  p = end;
1000  }
1001  } else if (!strcmp(parameter, "destination")) {
1002  if (*p == '=') {
1003  p++;
1004  get_word_sep(buf, sizeof(buf), ";,", &p);
1005  get_sockaddr(s, buf, &th->destination);
1006  }
1007  } else if (!strcmp(parameter, "source")) {
1008  if (*p == '=') {
1009  p++;
1010  get_word_sep(buf, sizeof(buf), ";,", &p);
1011  av_strlcpy(th->source, buf, sizeof(th->source));
1012  }
1013  } else if (!strcmp(parameter, "mode")) {
1014  if (*p == '=') {
1015  p++;
1016  get_word_sep(buf, sizeof(buf), ";, ", &p);
1017  if (!strcmp(buf, "record") ||
1018  !strcmp(buf, "receive"))
1019  th->mode_record = 1;
1020  }
1021  }
1022 
1023  while (*p != ';' && *p != '\0' && *p != ',')
1024  p++;
1025  if (*p == ';')
1026  p++;
1027  }
1028  if (*p == ',')
1029  p++;
1030 
1031  reply->nb_transports++;
1032  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1033  break;
1034  }
1035 }
1036 
1037 static void handle_rtp_info(RTSPState *rt, const char *url,
1038  uint32_t seq, uint32_t rtptime)
1039 {
1040  int i;
1041  if (!rtptime || !url[0])
1042  return;
1043  if (rt->transport != RTSP_TRANSPORT_RTP)
1044  return;
1045  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1046  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1047  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1048  if (!rtpctx)
1049  continue;
1050  if (!strcmp(rtsp_st->control_url, url)) {
1051  rtpctx->base_timestamp = rtptime;
1052  break;
1053  }
1054  }
1055 }
1056 
1057 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1058 {
1059  int read = 0;
1060  char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
1061  uint32_t seq = 0, rtptime = 0;
1062 
1063  for (;;) {
1064  p += strspn(p, SPACE_CHARS);
1065  if (!*p)
1066  break;
1067  get_word_sep(key, sizeof(key), "=", &p);
1068  if (*p != '=')
1069  break;
1070  p++;
1071  get_word_sep(value, sizeof(value), ";, ", &p);
1072  read++;
1073  if (!strcmp(key, "url"))
1074  av_strlcpy(url, value, sizeof(url));
1075  else if (!strcmp(key, "seq"))
1076  seq = strtoul(value, NULL, 10);
1077  else if (!strcmp(key, "rtptime"))
1078  rtptime = strtoul(value, NULL, 10);
1079  if (*p == ',') {
1080  handle_rtp_info(rt, url, seq, rtptime);
1081  url[0] = '\0';
1082  seq = rtptime = 0;
1083  read = 0;
1084  }
1085  if (*p)
1086  p++;
1087  }
1088  if (read > 0)
1089  handle_rtp_info(rt, url, seq, rtptime);
1090 }
1091 
1093  RTSPMessageHeader *reply, const char *buf,
1094  RTSPState *rt, const char *method)
1095 {
1096  const char *p;
1097 
1098  /* NOTE: we do case independent match for broken servers */
1099  p = buf;
1100  if (av_stristart(p, "Session:", &p)) {
1101  int t;
1102  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1103  if (av_stristart(p, ";timeout=", &p) &&
1104  (t = strtol(p, NULL, 10)) > 0) {
1105  reply->timeout = t;
1106  }
1107  } else if (av_stristart(p, "Content-Length:", &p)) {
1108  reply->content_length = strtol(p, NULL, 10);
1109  } else if (av_stristart(p, "Transport:", &p)) {
1110  rtsp_parse_transport(s, reply, p);
1111  } else if (av_stristart(p, "CSeq:", &p)) {
1112  reply->seq = strtol(p, NULL, 10);
1113  } else if (av_stristart(p, "Range:", &p)) {
1114  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1115  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1116  p += strspn(p, SPACE_CHARS);
1117  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1118  } else if (av_stristart(p, "Server:", &p)) {
1119  p += strspn(p, SPACE_CHARS);
1120  av_strlcpy(reply->server, p, sizeof(reply->server));
1121  } else if (av_stristart(p, "Notice:", &p) ||
1122  av_stristart(p, "X-Notice:", &p)) {
1123  reply->notice = strtol(p, NULL, 10);
1124  } else if (av_stristart(p, "Location:", &p)) {
1125  p += strspn(p, SPACE_CHARS);
1126  av_strlcpy(reply->location, p , sizeof(reply->location));
1127  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1128  p += strspn(p, SPACE_CHARS);
1129  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1130  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1131  p += strspn(p, SPACE_CHARS);
1132  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1133  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1134  p += strspn(p, SPACE_CHARS);
1135  if (method && !strcmp(method, "DESCRIBE"))
1136  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1137  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1138  p += strspn(p, SPACE_CHARS);
1139  if (method && !strcmp(method, "PLAY"))
1140  rtsp_parse_rtp_info(rt, p);
1141  } else if (av_stristart(p, "Public:", &p) && rt) {
1142  if (strstr(p, "GET_PARAMETER") &&
1143  method && !strcmp(method, "OPTIONS"))
1144  rt->get_parameter_supported = 1;
1145  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1146  p += strspn(p, SPACE_CHARS);
1147  rt->accept_dynamic_rate = atoi(p);
1148  } else if (av_stristart(p, "Content-Type:", &p)) {
1149  p += strspn(p, SPACE_CHARS);
1150  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1151  } else if (av_stristart(p, "com.ses.streamID:", &p)) {
1152  p += strspn(p, SPACE_CHARS);
1153  av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
1154  }
1155 }
1156 
1157 /* skip a RTP/TCP interleaved packet */
1159 {
1160  RTSPState *rt = s->priv_data;
1161  int ret, len, len1;
1162  uint8_t buf[MAX_URL_SIZE];
1163 
1164  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1165  if (ret != 3)
1166  return;
1167  len = AV_RB16(buf + 1);
1168 
1169  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1170 
1171  /* skip payload */
1172  while (len > 0) {
1173  len1 = len;
1174  if (len1 > sizeof(buf))
1175  len1 = sizeof(buf);
1176  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1177  if (ret != len1)
1178  return;
1179  len -= len1;
1180  }
1181 }
1182 
1184  unsigned char **content_ptr,
1185  int return_on_interleaved_data, const char *method)
1186 {
1187  RTSPState *rt = s->priv_data;
1188  char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
1189  unsigned char ch;
1190  const char *p;
1191  int ret, content_length, line_count = 0, request = 0;
1192  unsigned char *content = NULL;
1193 
1194 start:
1195  line_count = 0;
1196  request = 0;
1197  content = NULL;
1198  memset(reply, 0, sizeof(*reply));
1199 
1200  /* parse reply (XXX: use buffers) */
1201  rt->last_reply[0] = '\0';
1202  for (;;) {
1203  q = buf;
1204  for (;;) {
1205  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1206  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1207  if (ret != 1)
1208  return AVERROR_EOF;
1209  if (ch == '\n')
1210  break;
1211  if (ch == '$' && q == buf) {
1212  if (return_on_interleaved_data) {
1213  return 1;
1214  } else
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 (ffurl_read_complete(rt->rtsp_hd, content, content_length) != content_length)
1259  return AVERROR(EIO);
1260  content[content_length] = '\0';
1261  }
1262  if (content_ptr)
1263  *content_ptr = content;
1264  else
1265  av_freep(&content);
1266 
1267  if (request) {
1268  char buf[MAX_URL_SIZE];
1269  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1270  const char* ptr = buf;
1271 
1272  if (!strcmp(reply->reason, "OPTIONS")) {
1273  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1274  if (reply->seq)
1275  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1276  if (reply->session_id[0])
1277  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1278  reply->session_id);
1279  } else {
1280  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1281  }
1282  av_strlcat(buf, "\r\n", sizeof(buf));
1283 
1284  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1285  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1286  ptr = base64buf;
1287  }
1288  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1289 
1291  /* Even if the request from the server had data, it is not the data
1292  * that the caller wants or expects. The memory could also be leaked
1293  * if the actual following reply has content data. */
1294  if (content_ptr)
1295  av_freep(content_ptr);
1296  /* If method is set, this is called from ff_rtsp_send_cmd,
1297  * where a reply to exactly this request is awaited. For
1298  * callers from within packet receiving, we just want to
1299  * return to the caller and go back to receiving packets. */
1300  if (method)
1301  goto start;
1302  return 0;
1303  }
1304 
1305  if (rt->seq != reply->seq) {
1306  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1307  rt->seq, reply->seq);
1308  }
1309 
1310  /* EOS */
1311  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1312  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1313  reply->notice == 2306 /* Continuous Feed Terminated */) {
1314  rt->state = RTSP_STATE_IDLE;
1315  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1316  return AVERROR(EIO); /* data or server error */
1317  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1318  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1319  return AVERROR(EPERM);
1320 
1321  return 0;
1322 }
1323 
1324 /**
1325  * Send a command to the RTSP server without waiting for the reply.
1326  *
1327  * @param s RTSP (de)muxer context
1328  * @param method the method for the request
1329  * @param url the target url for the request
1330  * @param headers extra header lines to include in the request
1331  * @param send_content if non-null, the data to send as request body content
1332  * @param send_content_length the length of the send_content data, or 0 if
1333  * send_content is null
1334  *
1335  * @return zero if success, nonzero otherwise
1336  */
1337 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1338  const char *method, const char *url,
1339  const char *headers,
1340  const unsigned char *send_content,
1341  int send_content_length)
1342 {
1343  RTSPState *rt = s->priv_data;
1344  char buf[MAX_URL_SIZE], *out_buf;
1345  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1346 
1347  if (!rt->rtsp_hd_out)
1348  return AVERROR(ENOTCONN);
1349 
1350  /* Add in RTSP headers */
1351  out_buf = buf;
1352  rt->seq++;
1353  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1354  if (headers)
1355  av_strlcat(buf, headers, sizeof(buf));
1356  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1357  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1358  if (rt->session_id[0] != '\0' && (!headers ||
1359  !strstr(headers, "\nIf-Match:"))) {
1360  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1361  }
1362  if (rt->auth[0]) {
1364  rt->auth, url, method);
1365  if (str)
1366  av_strlcat(buf, str, sizeof(buf));
1367  av_free(str);
1368  }
1369  if (send_content_length > 0 && send_content)
1370  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1371  av_strlcat(buf, "\r\n", sizeof(buf));
1372 
1373  /* base64 encode rtsp if tunneling */
1374  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1375  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1376  out_buf = base64buf;
1377  }
1378 
1379  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1380 
1381  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1382  if (send_content_length > 0 && send_content) {
1383  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1384  avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1385  return AVERROR_PATCHWELCOME;
1386  }
1387  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1388  }
1390 
1391  return 0;
1392 }
1393 
1394 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1395  const char *url, const char *headers)
1396 {
1397  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1398 }
1399 
1400 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1401  const char *headers, RTSPMessageHeader *reply,
1402  unsigned char **content_ptr)
1403 {
1404  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1405  content_ptr, NULL, 0);
1406 }
1407 
1409  const char *method, const char *url,
1410  const char *header,
1411  RTSPMessageHeader *reply,
1412  unsigned char **content_ptr,
1413  const unsigned char *send_content,
1414  int send_content_length)
1415 {
1416  RTSPState *rt = s->priv_data;
1417  HTTPAuthType cur_auth_type;
1418  int ret, attempts = 0;
1419 
1420 retry:
1421  cur_auth_type = rt->auth_state.auth_type;
1422  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1423  send_content,
1424  send_content_length)))
1425  return ret;
1426 
1427  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1428  return ret;
1429  attempts++;
1430 
1431  if (reply->status_code == 401 &&
1432  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1433  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1434  goto retry;
1435 
1436  if (reply->status_code > 400){
1437  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1438  method,
1439  reply->status_code,
1440  reply->reason);
1441  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1442  }
1443 
1444  return 0;
1445 }
1446 
1447 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1448  int lower_transport, const char *real_challenge)
1449 {
1450  RTSPState *rt = s->priv_data;
1451  int rtx = 0, j, i, err, interleave = 0, port_off;
1452  RTSPStream *rtsp_st;
1453  RTSPMessageHeader reply1, *reply = &reply1;
1454  char cmd[MAX_URL_SIZE];
1455  const char *trans_pref;
1456 
1457  if (rt->transport == RTSP_TRANSPORT_RDT)
1458  trans_pref = "x-pn-tng";
1459  else if (rt->transport == RTSP_TRANSPORT_RAW)
1460  trans_pref = "RAW/RAW";
1461  else
1462  trans_pref = "RTP/AVP";
1463 
1464  /* default timeout: 1 minute */
1465  rt->timeout = 60;
1466 
1467  /* Choose a random starting offset within the first half of the
1468  * port range, to allow for a number of ports to try even if the offset
1469  * happens to be at the end of the random range. */
1470  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1471  /* even random offset */
1472  port_off -= port_off & 0x01;
1473 
1474  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1475  char transport[MAX_URL_SIZE];
1476 
1477  /*
1478  * WMS serves all UDP data over a single connection, the RTX, which
1479  * isn't necessarily the first in the SDP but has to be the first
1480  * to be set up, else the second/third SETUP will fail with a 461.
1481  */
1482  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1483  rt->server_type == RTSP_SERVER_WMS) {
1484  if (i == 0) {
1485  /* rtx first */
1486  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1487  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1488  if (len >= 4 &&
1489  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1490  "/rtx"))
1491  break;
1492  }
1493  if (rtx == rt->nb_rtsp_streams)
1494  return -1; /* no RTX found */
1495  rtsp_st = rt->rtsp_streams[rtx];
1496  } else
1497  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1498  } else
1499  rtsp_st = rt->rtsp_streams[i];
1500 
1501  /* RTP/UDP */
1502  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1503  char buf[256];
1504 
1505  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1506  port = reply->transports[0].client_port_min;
1507  goto have_port;
1508  }
1509 
1510  /* first try in specified port range */
1511  while (j <= rt->rtp_port_max) {
1512  AVDictionary *opts = map_to_opts(rt);
1513 
1514  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1515  "?localport=%d", j);
1516  /* we will use two ports per rtp stream (rtp and rtcp) */
1517  j += 2;
1519  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1520 
1521  av_dict_free(&opts);
1522 
1523  if (!err)
1524  goto rtp_opened;
1525  }
1526  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1527  err = AVERROR(EIO);
1528  goto fail;
1529 
1530  rtp_opened:
1531  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1532  have_port:
1533  av_strlcpy(transport, trans_pref, sizeof(transport));
1534  av_strlcat(transport,
1535  rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
1536  sizeof(transport));
1537  if (rt->server_type != RTSP_SERVER_REAL)
1538  av_strlcat(transport, "unicast;", sizeof(transport));
1539  av_strlcatf(transport, sizeof(transport),
1540  "client_port=%d", port);
1541  if (rt->transport == RTSP_TRANSPORT_RTP &&
1542  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1543  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1544  }
1545 
1546  /* RTP/TCP */
1547  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1548  /* For WMS streams, the application streams are only used for
1549  * UDP. When trying to set it up for TCP streams, the server
1550  * will return an error. Therefore, we skip those streams. */
1551  if (rt->server_type == RTSP_SERVER_WMS &&
1552  (rtsp_st->stream_index < 0 ||
1553  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1555  continue;
1556  snprintf(transport, sizeof(transport) - 1,
1557  "%s/TCP;", trans_pref);
1558  if (rt->transport != RTSP_TRANSPORT_RDT)
1559  av_strlcat(transport, "unicast;", sizeof(transport));
1560  av_strlcatf(transport, sizeof(transport),
1561  "interleaved=%d-%d",
1562  interleave, interleave + 1);
1563  interleave += 2;
1564  }
1565 
1566  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1567  snprintf(transport, sizeof(transport) - 1,
1568  "%s/UDP;multicast", trans_pref);
1569  }
1570  if (s->oformat) {
1571  av_strlcat(transport, ";mode=record", sizeof(transport));
1572  } else if (rt->server_type == RTSP_SERVER_REAL ||
1574  av_strlcat(transport, ";mode=play", sizeof(transport));
1575  snprintf(cmd, sizeof(cmd),
1576  "Transport: %s\r\n",
1577  transport);
1578  if (rt->accept_dynamic_rate)
1579  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1580  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1581  char real_res[41], real_csum[9];
1582  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1583  real_challenge);
1584  av_strlcatf(cmd, sizeof(cmd),
1585  "If-Match: %s\r\n"
1586  "RealChallenge2: %s, sd=%s\r\n",
1587  rt->session_id, real_res, real_csum);
1588  }
1589  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1590  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1591  err = 1;
1592  goto fail;
1593  } else if (reply->status_code != RTSP_STATUS_OK ||
1594  reply->nb_transports != 1) {
1596  goto fail;
1597  }
1598 
1599  if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
1600  char proto[128], host[128], path[512], auth[128];
1601  int port;
1602  av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
1603  &port, path, sizeof(path), rt->control_uri);
1604  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
1605  port, "/stream=%s", reply->stream_id);
1606  }
1607 
1608  /* XXX: same protocol for all streams is required */
1609  if (i > 0) {
1610  if (reply->transports[0].lower_transport != rt->lower_transport ||
1611  reply->transports[0].transport != rt->transport) {
1612  err = AVERROR_INVALIDDATA;
1613  goto fail;
1614  }
1615  } else {
1616  rt->lower_transport = reply->transports[0].lower_transport;
1617  rt->transport = reply->transports[0].transport;
1618  }
1619 
1620  /* Fail if the server responded with another lower transport mode
1621  * than what we requested. */
1622  if (reply->transports[0].lower_transport != lower_transport) {
1623  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1624  err = AVERROR_INVALIDDATA;
1625  goto fail;
1626  }
1627 
1628  switch(reply->transports[0].lower_transport) {
1630  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1631  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1632  break;
1633 
1634  case RTSP_LOWER_TRANSPORT_UDP: {
1635  char url[MAX_URL_SIZE], options[30] = "";
1636  const char *peer = host;
1637 
1638  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1639  av_strlcpy(options, "?connect=1", sizeof(options));
1640  /* Use source address if specified */
1641  if (reply->transports[0].source[0])
1642  peer = reply->transports[0].source;
1643  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1644  reply->transports[0].server_port_min, "%s", options);
1645  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1646  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1647  err = AVERROR_INVALIDDATA;
1648  goto fail;
1649  }
1650  break;
1651  }
1653  char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
1654  struct sockaddr_storage addr;
1655  int port, ttl;
1656  AVDictionary *opts = map_to_opts(rt);
1657 
1658  if (reply->transports[0].destination.ss_family) {
1659  addr = reply->transports[0].destination;
1660  port = reply->transports[0].port_min;
1661  ttl = reply->transports[0].ttl;
1662  } else {
1663  addr = rtsp_st->sdp_ip;
1664  port = rtsp_st->sdp_port;
1665  ttl = rtsp_st->sdp_ttl;
1666  }
1667  if (ttl > 0)
1668  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1669  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1670  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1671  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1672  port, "%s", optbuf);
1674  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1675  av_dict_free(&opts);
1676 
1677  if (err < 0) {
1678  err = AVERROR_INVALIDDATA;
1679  goto fail;
1680  }
1681  break;
1682  }
1683  }
1684 
1685  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1686  goto fail;
1687  }
1688 
1689  if (rt->nb_rtsp_streams && reply->timeout > 0)
1690  rt->timeout = reply->timeout;
1691 
1692  if (rt->server_type == RTSP_SERVER_REAL)
1693  rt->need_subscription = 1;
1694 
1695  return 0;
1696 
1697 fail:
1698  ff_rtsp_undo_setup(s, 0);
1699  return err;
1700 }
1701 
1703 {
1704  RTSPState *rt = s->priv_data;
1705  if (rt->rtsp_hd_out != rt->rtsp_hd)
1706  ffurl_closep(&rt->rtsp_hd_out);
1707  rt->rtsp_hd_out = NULL;
1708  ffurl_closep(&rt->rtsp_hd);
1709 }
1710 
1712 {
1713  RTSPState *rt = s->priv_data;
1714  char proto[128], host[1024], path[1024];
1715  char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
1716  const char *lower_rtsp_proto = "tcp";
1717  int port, err, tcp_fd;
1718  RTSPMessageHeader reply1, *reply = &reply1;
1719  int lower_transport_mask = 0;
1720  int default_port = RTSP_DEFAULT_PORT;
1721  int https_tunnel = 0;
1722  char real_challenge[64] = "";
1723  struct sockaddr_storage peer;
1724  socklen_t peer_len = sizeof(peer);
1725 
1726  if (rt->rtp_port_max < rt->rtp_port_min) {
1727  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1728  "than min port %d\n", rt->rtp_port_max,
1729  rt->rtp_port_min);
1730  return AVERROR(EINVAL);
1731  }
1732 
1733  if (!ff_network_init())
1734  return AVERROR(EIO);
1735 
1736  if (s->max_delay < 0) /* Not set by the caller */
1737  s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1738 
1741  (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1742  https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1745  }
1746  /* Only pass through valid flags from here */
1748 
1749 redirect:
1750  memset(&reply1, 0, sizeof(reply1));
1751  /* extract hostname and port */
1752  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1753  host, sizeof(host), &port, path, sizeof(path), s->url);
1754 
1755  if (!strcmp(proto, "rtsps")) {
1756  lower_rtsp_proto = "tls";
1757  default_port = RTSPS_DEFAULT_PORT;
1759  } else if (!strcmp(proto, "satip")) {
1760  av_strlcpy(proto, "rtsp", sizeof(proto));
1762  }
1763 
1764  if (*auth) {
1765  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1766  }
1767  if (port < 0)
1768  port = default_port;
1769 
1770  lower_transport_mask = rt->lower_transport_mask;
1771 
1772  if (!lower_transport_mask)
1773  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1774 
1775  if (s->oformat) {
1776  /* Only UDP or TCP - UDP multicast isn't supported. */
1777  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1778  (1 << RTSP_LOWER_TRANSPORT_TCP);
1779  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1780  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1781  "only UDP and TCP are supported for output.\n");
1782  err = AVERROR(EINVAL);
1783  goto fail;
1784  }
1785  }
1786 
1787  /* Construct the URI used in request; this is similar to s->url,
1788  * but with authentication credentials removed and RTSP specific options
1789  * stripped out. */
1790  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1791  host, port, "%s", path);
1792 
1793  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1794  /* set up initial handshake for tunneling */
1795  char httpname[1024];
1796  char sessioncookie[17];
1797  char headers[1024];
1799 
1800  av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1801 
1802  ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1803  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1805 
1806  /* GET requests */
1807  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1808  &s->interrupt_callback) < 0) {
1809  err = AVERROR(EIO);
1810  goto fail;
1811  }
1812 
1813  /* generate GET headers */
1814  snprintf(headers, sizeof(headers),
1815  "x-sessioncookie: %s\r\n"
1816  "Accept: application/x-rtsp-tunnelled\r\n"
1817  "Pragma: no-cache\r\n"
1818  "Cache-Control: no-cache\r\n",
1819  sessioncookie);
1820  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1821 
1822  if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1823  rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1824  if (!rt->rtsp_hd->protocol_whitelist) {
1825  err = AVERROR(ENOMEM);
1826  goto fail;
1827  }
1828  }
1829 
1830  /* complete the connection */
1831  if (ffurl_connect(rt->rtsp_hd, &options)) {
1833  err = AVERROR(EIO);
1834  goto fail;
1835  }
1836 
1837  /* POST requests */
1838  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1839  &s->interrupt_callback) < 0 ) {
1840  err = AVERROR(EIO);
1841  goto fail;
1842  }
1843 
1844  /* generate POST headers */
1845  snprintf(headers, sizeof(headers),
1846  "x-sessioncookie: %s\r\n"
1847  "Content-Type: application/x-rtsp-tunnelled\r\n"
1848  "Pragma: no-cache\r\n"
1849  "Cache-Control: no-cache\r\n"
1850  "Content-Length: 32767\r\n"
1851  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1852  sessioncookie);
1853  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1854  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1855  av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1856 
1857  /* Initialize the authentication state for the POST session. The HTTP
1858  * protocol implementation doesn't properly handle multi-pass
1859  * authentication for POST requests, since it would require one of
1860  * the following:
1861  * - implementing Expect: 100-continue, which many HTTP servers
1862  * don't support anyway, even less the RTSP servers that do HTTP
1863  * tunneling
1864  * - sending the whole POST data until getting a 401 reply specifying
1865  * what authentication method to use, then resending all that data
1866  * - waiting for potential 401 replies directly after sending the
1867  * POST header (waiting for some unspecified time)
1868  * Therefore, we copy the full auth state, which works for both basic
1869  * and digest. (For digest, we would have to synchronize the nonce
1870  * count variable between the two sessions, if we'd do more requests
1871  * with the original session, though.)
1872  */
1874 
1875  /* complete the connection */
1876  if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1878  err = AVERROR(EIO);
1879  goto fail;
1880  }
1882  } else {
1883  int ret;
1884  /* open the tcp connection */
1885  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1886  host, port,
1887  "?timeout=%d", rt->stimeout);
1888  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1889  &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1890  err = ret;
1891  goto fail;
1892  }
1893  rt->rtsp_hd_out = rt->rtsp_hd;
1894  }
1895  rt->seq = 0;
1896 
1897  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1898  if (tcp_fd < 0) {
1899  err = tcp_fd;
1900  goto fail;
1901  }
1902  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1903  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1904  NULL, 0, NI_NUMERICHOST);
1905  }
1906 
1907  /* request options supported by the server; this also detects server
1908  * type */
1909  if (rt->server_type != RTSP_SERVER_SATIP)
1911  for (;;) {
1912  cmd[0] = 0;
1913  if (rt->server_type == RTSP_SERVER_REAL)
1914  av_strlcat(cmd,
1915  /*
1916  * The following entries are required for proper
1917  * streaming from a Realmedia server. They are
1918  * interdependent in some way although we currently
1919  * don't quite understand how. Values were copied
1920  * from mplayer SVN r23589.
1921  * ClientChallenge is a 16-byte ID in hex
1922  * CompanyID is a 16-byte ID in base64
1923  */
1924  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1925  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1926  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1927  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1928  sizeof(cmd));
1929  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1930  if (reply->status_code != RTSP_STATUS_OK) {
1932  goto fail;
1933  }
1934 
1935  /* detect server type if not standard-compliant RTP */
1936  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1938  continue;
1939  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1941  } else if (rt->server_type == RTSP_SERVER_REAL)
1942  strcpy(real_challenge, reply->real_challenge);
1943  break;
1944  }
1945 
1946 #if CONFIG_RTSP_DEMUXER
1947  if (s->iformat) {
1948  if (rt->server_type == RTSP_SERVER_SATIP)
1949  err = init_satip_stream(s);
1950  else
1951  err = ff_rtsp_setup_input_streams(s, reply);
1952  } else
1953 #endif
1954  if (CONFIG_RTSP_MUXER)
1955  err = ff_rtsp_setup_output_streams(s, host);
1956  else
1957  av_assert0(0);
1958  if (err)
1959  goto fail;
1960 
1961  do {
1962  int lower_transport = ff_log2_tab[lower_transport_mask &
1963  ~(lower_transport_mask - 1)];
1964 
1965  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1966  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1967  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1968 
1969  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1970  rt->server_type == RTSP_SERVER_REAL ?
1971  real_challenge : NULL);
1972  if (err < 0)
1973  goto fail;
1974  lower_transport_mask &= ~(1 << lower_transport);
1975  if (lower_transport_mask == 0 && err == 1) {
1976  err = AVERROR(EPROTONOSUPPORT);
1977  goto fail;
1978  }
1979  } while (err);
1980 
1981  rt->lower_transport_mask = lower_transport_mask;
1982  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1983  rt->state = RTSP_STATE_IDLE;
1984  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1985  return 0;
1986  fail:
1989  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1990  char *new_url = av_strdup(reply->location);
1991  if (!new_url) {
1992  err = AVERROR(ENOMEM);
1993  goto fail2;
1994  }
1995  ff_format_set_url(s, new_url);
1996  rt->session_id[0] = '\0';
1997  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1998  reply->status_code,
1999  s->url);
2000  goto redirect;
2001  }
2002  fail2:
2003  ff_network_close();
2004  return err;
2005 }
2006 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
2007 
2008 #if CONFIG_RTPDEC
2009 static int parse_rtsp_message(AVFormatContext *s)
2010 {
2011  RTSPState *rt = s->priv_data;
2012  int ret;
2013 
2014  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
2015  if (rt->state == RTSP_STATE_STREAMING) {
2017  } else
2018  return AVERROR_EOF;
2019  } else {
2020  RTSPMessageHeader reply;
2021  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
2022  if (ret < 0)
2023  return ret;
2024  /* XXX: parse message */
2025  if (rt->state != RTSP_STATE_STREAMING)
2026  return 0;
2027  }
2028 
2029  return 0;
2030 }
2031 
2032 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
2033  uint8_t *buf, int buf_size, int64_t wait_end)
2034 {
2035  RTSPState *rt = s->priv_data;
2036  RTSPStream *rtsp_st;
2037  int n, i, ret;
2038  struct pollfd *p = rt->p;
2039  int *fds = NULL, fdsnum, fdsidx;
2040  int runs = rt->initial_timeout * 1000LL / POLLING_TIME;
2041 
2042  if (!p) {
2043  p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
2044  if (!p)
2045  return AVERROR(ENOMEM);
2046 
2047  if (rt->rtsp_hd) {
2048  p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2049  p[rt->max_p++].events = POLLIN;
2050  }
2051  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2052  rtsp_st = rt->rtsp_streams[i];
2053  if (rtsp_st->rtp_handle) {
2054  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2055  &fds, &fdsnum)) {
2056  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2057  return ret;
2058  }
2059  if (fdsnum != 2) {
2061  "Number of fds %d not supported\n", fdsnum);
2062  return AVERROR_INVALIDDATA;
2063  }
2064  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2065  p[rt->max_p].fd = fds[fdsidx];
2066  p[rt->max_p++].events = POLLIN;
2067  }
2068  av_freep(&fds);
2069  }
2070  }
2071  }
2072 
2073  for (;;) {
2074  if (ff_check_interrupt(&s->interrupt_callback))
2075  return AVERROR_EXIT;
2076  if (wait_end && wait_end - av_gettime_relative() < 0)
2077  return AVERROR(EAGAIN);
2078  n = poll(p, rt->max_p, POLLING_TIME);
2079  if (n > 0) {
2080  int j = rt->rtsp_hd ? 1 : 0;
2081  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2082  rtsp_st = rt->rtsp_streams[i];
2083  if (rtsp_st->rtp_handle) {
2084  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2085  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2086  if (ret > 0) {
2087  *prtsp_st = rtsp_st;
2088  return ret;
2089  }
2090  }
2091  j+=2;
2092  }
2093  }
2094 #if CONFIG_RTSP_DEMUXER
2095  if (rt->rtsp_hd && p[0].revents & POLLIN) {
2096  if ((ret = parse_rtsp_message(s)) < 0) {
2097  return ret;
2098  }
2099  }
2100 #endif
2101  } else if (n == 0 && rt->initial_timeout > 0 && --runs <= 0) {
2102  return AVERROR(ETIMEDOUT);
2103  } else if (n < 0 && errno != EINTR)
2104  return AVERROR(errno);
2105  }
2106 }
2107 
2108 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2109  const uint8_t *buf, int len)
2110 {
2111  RTSPState *rt = s->priv_data;
2112  int i;
2113  if (len < 0)
2114  return len;
2115  if (rt->nb_rtsp_streams == 1) {
2116  *rtsp_st = rt->rtsp_streams[0];
2117  return len;
2118  }
2119  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2120  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2121  int no_ssrc = 0;
2122  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2123  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2124  if (!rtpctx)
2125  continue;
2126  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2127  *rtsp_st = rt->rtsp_streams[i];
2128  return len;
2129  }
2130  if (!rtpctx->ssrc)
2131  no_ssrc = 1;
2132  }
2133  if (no_ssrc) {
2135  "Unable to pick stream for packet - SSRC not known for "
2136  "all streams\n");
2137  return AVERROR(EAGAIN);
2138  }
2139  } else {
2140  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2141  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2142  *rtsp_st = rt->rtsp_streams[i];
2143  return len;
2144  }
2145  }
2146  }
2147  }
2148  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2149  return AVERROR(EAGAIN);
2150 }
2151 
2152 static int read_packet(AVFormatContext *s,
2153  RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2154  int64_t wait_end)
2155 {
2156  RTSPState *rt = s->priv_data;
2157  int len;
2158 
2159  switch(rt->lower_transport) {
2160  default:
2161 #if CONFIG_RTSP_DEMUXER
2163  len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2164  break;
2165 #endif
2168  len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2169  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2170  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2171  break;
2173  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2174  wait_end && wait_end < av_gettime_relative())
2175  len = AVERROR(EAGAIN);
2176  else
2178  len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2179  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2180  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2181  break;
2182  }
2183 
2184  if (len == 0)
2185  return AVERROR_EOF;
2186 
2187  return len;
2188 }
2189 
2191 {
2192  RTSPState *rt = s->priv_data;
2193  int ret, len;
2194  RTSPStream *rtsp_st, *first_queue_st = NULL;
2195  int64_t wait_end = 0;
2196 
2197  if (rt->nb_byes == rt->nb_rtsp_streams)
2198  return AVERROR_EOF;
2199 
2200  /* get next frames from the same RTP packet */
2201  if (rt->cur_transport_priv) {
2202  if (rt->transport == RTSP_TRANSPORT_RDT) {
2204  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2206  } else if (CONFIG_RTPDEC && rt->ts) {
2207  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2208  if (ret >= 0) {
2209  rt->recvbuf_pos += ret;
2210  ret = rt->recvbuf_pos < rt->recvbuf_len;
2211  }
2212  } else
2213  ret = -1;
2214  if (ret == 0) {
2215  rt->cur_transport_priv = NULL;
2216  return 0;
2217  } else if (ret == 1) {
2218  return 0;
2219  } else
2220  rt->cur_transport_priv = NULL;
2221  }
2222 
2223 redo:
2224  if (rt->transport == RTSP_TRANSPORT_RTP) {
2225  int i;
2226  int64_t first_queue_time = 0;
2227  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2228  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2229  int64_t queue_time;
2230  if (!rtpctx)
2231  continue;
2232  queue_time = ff_rtp_queued_packet_time(rtpctx);
2233  if (queue_time && (queue_time - first_queue_time < 0 ||
2234  !first_queue_time)) {
2235  first_queue_time = queue_time;
2236  first_queue_st = rt->rtsp_streams[i];
2237  }
2238  }
2239  if (first_queue_time) {
2240  wait_end = first_queue_time + s->max_delay;
2241  } else {
2242  wait_end = 0;
2243  first_queue_st = NULL;
2244  }
2245  }
2246 
2247  /* read next RTP packet */
2248  if (!rt->recvbuf) {
2250  if (!rt->recvbuf)
2251  return AVERROR(ENOMEM);
2252  }
2253 
2254  len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2255  if (len == AVERROR(EAGAIN) && first_queue_st &&
2256  rt->transport == RTSP_TRANSPORT_RTP) {
2258  "max delay reached. need to consume packet\n");
2259  rtsp_st = first_queue_st;
2260  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2261  goto end;
2262  }
2263  if (len < 0)
2264  return len;
2265 
2266  if (rt->transport == RTSP_TRANSPORT_RDT) {
2267  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2268  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2269  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2270  if (rtsp_st->feedback) {
2271  AVIOContext *pb = NULL;
2273  pb = s->pb;
2274  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2275  }
2276  if (ret < 0) {
2277  /* Either bad packet, or a RTCP packet. Check if the
2278  * first_rtcp_ntp_time field was initialized. */
2279  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2280  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2281  /* first_rtcp_ntp_time has been initialized for this stream,
2282  * copy the same value to all other uninitialized streams,
2283  * in order to map their timestamp origin to the same ntp time
2284  * as this one. */
2285  int i;
2286  AVStream *st = NULL;
2287  if (rtsp_st->stream_index >= 0)
2288  st = s->streams[rtsp_st->stream_index];
2289  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2290  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2291  AVStream *st2 = NULL;
2292  if (rt->rtsp_streams[i]->stream_index >= 0)
2293  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2294  if (rtpctx2 && st && st2 &&
2295  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2296  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2297  rtpctx2->rtcp_ts_offset = av_rescale_q(
2298  rtpctx->rtcp_ts_offset, st->time_base,
2299  st2->time_base);
2300  }
2301  }
2302  // Make real NTP start time available in AVFormatContext
2303  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2304  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2305  if (rtpctx->st) {
2306  s->start_time_realtime -=
2307  av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
2308  }
2309  }
2310  }
2311  if (ret == -RTCP_BYE) {
2312  rt->nb_byes++;
2313 
2314  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2315  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2316 
2317  if (rt->nb_byes == rt->nb_rtsp_streams)
2318  return AVERROR_EOF;
2319  }
2320  }
2321  } else if (CONFIG_RTPDEC && rt->ts) {
2322  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2323  if (ret >= 0) {
2324  if (ret < len) {
2325  rt->recvbuf_len = len;
2326  rt->recvbuf_pos = ret;
2327  rt->cur_transport_priv = rt->ts;
2328  return 1;
2329  } else {
2330  ret = 0;
2331  }
2332  }
2333  } else {
2334  return AVERROR_INVALIDDATA;
2335  }
2336 end:
2337  if (ret < 0)
2338  goto redo;
2339  if (ret == 1)
2340  /* more packets may follow, so we save the RTP context */
2341  rt->cur_transport_priv = rtsp_st->transport_priv;
2342 
2343  return ret;
2344 }
2345 #endif /* CONFIG_RTPDEC */
2346 
2347 #if CONFIG_SDP_DEMUXER
2348 static int sdp_probe(const AVProbeData *p1)
2349 {
2350  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2351 
2352  /* we look for a line beginning "c=IN IP" */
2353  while (p < p_end && *p != '\0') {
2354  if (sizeof("c=IN IP") - 1 < p_end - p &&
2355  av_strstart(p, "c=IN IP", NULL))
2356  return AVPROBE_SCORE_EXTENSION;
2357 
2358  while (p < p_end - 1 && *p != '\n') p++;
2359  if (++p >= p_end)
2360  break;
2361  if (*p == '\r')
2362  p++;
2363  }
2364  return 0;
2365 }
2366 
2367 static void append_source_addrs(char *buf, int size, const char *name,
2368  int count, struct RTSPSource **addrs)
2369 {
2370  int i;
2371  if (!count)
2372  return;
2373  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2374  for (i = 1; i < count; i++)
2375  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2376 }
2377 
2378 static int sdp_read_header(AVFormatContext *s)
2379 {
2380  RTSPState *rt = s->priv_data;
2381  RTSPStream *rtsp_st;
2382  int size, i, err;
2383  char *content;
2384  char url[MAX_URL_SIZE];
2385 
2386  if (!ff_network_init())
2387  return AVERROR(EIO);
2388 
2389  if (s->max_delay < 0) /* Not set by the caller */
2390  s->max_delay = DEFAULT_REORDERING_DELAY;
2391  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2393 
2394  /* read the whole sdp file */
2395  /* XXX: better loading */
2396  content = av_malloc(SDP_MAX_SIZE);
2397  if (!content) {
2398  ff_network_close();
2399  return AVERROR(ENOMEM);
2400  }
2401  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2402  if (size <= 0) {
2403  av_free(content);
2404  ff_network_close();
2405  return AVERROR_INVALIDDATA;
2406  }
2407  content[size] ='\0';
2408 
2409  err = ff_sdp_parse(s, content);
2410  av_freep(&content);
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 
2421  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2422  sizeof(rtsp_st->sdp_ip),
2423  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2424  if (err) {
2425  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2426  err = AVERROR(EIO);
2427  av_dict_free(&opts);
2428  goto fail;
2429  }
2430  ff_url_join(url, sizeof(url), "rtp", NULL,
2431  namebuf, rtsp_st->sdp_port,
2432  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2433  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2434  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2435  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2436 
2437  append_source_addrs(url, sizeof(url), "sources",
2438  rtsp_st->nb_include_source_addrs,
2439  rtsp_st->include_source_addrs);
2440  append_source_addrs(url, sizeof(url), "block",
2441  rtsp_st->nb_exclude_source_addrs,
2442  rtsp_st->exclude_source_addrs);
2443  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2444  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2445 
2446  av_dict_free(&opts);
2447 
2448  if (err < 0) {
2449  err = AVERROR_INVALIDDATA;
2450  goto fail;
2451  }
2452  }
2453  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2454  goto fail;
2455  }
2456  return 0;
2457 fail:
2459  ff_network_close();
2460  return err;
2461 }
2462 
2463 static int sdp_read_close(AVFormatContext *s)
2464 {
2466  ff_network_close();
2467  return 0;
2468 }
2469 
2470 static const AVClass sdp_demuxer_class = {
2471  .class_name = "SDP demuxer",
2472  .item_name = av_default_item_name,
2473  .option = sdp_options,
2474  .version = LIBAVUTIL_VERSION_INT,
2475 };
2476 
2478  .name = "sdp",
2479  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2480  .priv_data_size = sizeof(RTSPState),
2481  .read_probe = sdp_probe,
2482  .read_header = sdp_read_header,
2484  .read_close = sdp_read_close,
2485  .priv_class = &sdp_demuxer_class,
2486 };
2487 #endif /* CONFIG_SDP_DEMUXER */
2488 
2489 #if CONFIG_RTP_DEMUXER
2490 static int rtp_probe(const AVProbeData *p)
2491 {
2492  if (av_strstart(p->filename, "rtp:", NULL))
2493  return AVPROBE_SCORE_MAX;
2494  return 0;
2495 }
2496 
2497 static int rtp_read_header(AVFormatContext *s)
2498 {
2499  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2500  char host[500], filters_buf[1000];
2501  int ret, port;
2502  URLContext* in = NULL;
2503  int payload_type;
2504  AVCodecParameters *par = NULL;
2505  struct sockaddr_storage addr;
2506  AVIOContext pb;
2507  socklen_t addrlen = sizeof(addr);
2508  RTSPState *rt = s->priv_data;
2509  const char *p;
2510  AVBPrint sdp;
2511  AVDictionary *opts = NULL;
2512 
2513  if (!ff_network_init())
2514  return AVERROR(EIO);
2515 
2516  opts = map_to_opts(rt);
2517  ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2518  &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2519  av_dict_free(&opts);
2520  if (ret)
2521  goto fail;
2522 
2523  while (1) {
2524  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2525  if (ret == AVERROR(EAGAIN))
2526  continue;
2527  if (ret < 0)
2528  goto fail;
2529  if (ret < 12) {
2530  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2531  continue;
2532  }
2533 
2534  if ((recvbuf[0] & 0xc0) != 0x80) {
2535  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2536  "received\n");
2537  continue;
2538  }
2539 
2540  if (RTP_PT_IS_RTCP(recvbuf[1]))
2541  continue;
2542 
2543  payload_type = recvbuf[1] & 0x7f;
2544  break;
2545  }
2546  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2547  ffurl_closep(&in);
2548 
2549  par = avcodec_parameters_alloc();
2550  if (!par) {
2551  ret = AVERROR(ENOMEM);
2552  goto fail;
2553  }
2554 
2555  if (ff_rtp_get_codec_info(par, payload_type)) {
2556  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2557  "without an SDP file describing it\n",
2558  payload_type);
2559  ret = AVERROR_INVALIDDATA;
2560  goto fail;
2561  }
2562  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2563  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2564  "properly you need an SDP file "
2565  "describing it\n");
2566  }
2567 
2568  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2569  NULL, 0, s->url);
2570 
2572  av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2573  addr.ss_family == AF_INET ? 4 : 6, host);
2574 
2575  p = strchr(s->url, '?');
2576  if (p) {
2577  static const char filters[][2][8] = { { "sources", "incl" },
2578  { "block", "excl" } };
2579  int i;
2580  char *q;
2581  for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2582  if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2583  q = filters_buf;
2584  while ((q = strchr(q, ',')) != NULL)
2585  *q = ' ';
2586  av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2587  filters[i][1],
2588  addr.ss_family == AF_INET ? 4 : 6, host,
2589  filters_buf);
2590  }
2591  }
2592  }
2593 
2594  av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2595  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2596  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2597  port, payload_type);
2598  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2599  if (!av_bprint_is_complete(&sdp))
2600  goto fail_nobuf;
2602 
2603  ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2604  s->pb = &pb;
2605 
2606  /* if sdp_read_header() fails then following ff_network_close() cancels out */
2607  /* ff_network_init() at the start of this function. Otherwise it cancels out */
2608  /* ff_network_init() inside sdp_read_header() */
2609  ff_network_close();
2610 
2611  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2612 
2613  ret = sdp_read_header(s);
2614  s->pb = NULL;
2615  av_bprint_finalize(&sdp, NULL);
2616  return ret;
2617 
2618 fail_nobuf:
2619  ret = AVERROR(ENOMEM);
2620  av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2621  av_bprint_finalize(&sdp, NULL);
2622 fail:
2624  ffurl_closep(&in);
2625  ff_network_close();
2626  return ret;
2627 }
2628 
2629 static const AVClass rtp_demuxer_class = {
2630  .class_name = "RTP demuxer",
2631  .item_name = av_default_item_name,
2632  .option = rtp_options,
2633  .version = LIBAVUTIL_VERSION_INT,
2634 };
2635 
2637  .name = "rtp",
2638  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2639  .priv_data_size = sizeof(RTSPState),
2640  .read_probe = rtp_probe,
2641  .read_header = rtp_read_header,
2643  .read_close = sdp_read_close,
2644  .flags = AVFMT_NOFILE,
2645  .priv_class = &rtp_demuxer_class,
2646 };
2647 #endif /* CONFIG_RTP_DEMUXER */
AVInputFormat ff_sdp_demuxer
AVInputFormat ff_rtp_demuxer
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Main libavformat public API header.
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:453
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1177
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:458
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:451
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:296
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:404
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:661
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:309
int ffurl_closep(URLContext **hh)
Close the resource accessed by the URLContext h, and free the memory used by it.
Definition: avio.c:441
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:169
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:418
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:623
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:630
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:411
#define AVIO_FLAG_READ
read-only
Definition: avio.h:674
int avio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:704
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:675
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:676
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:633
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:1195
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1457
int ffio_init_context(AVIOContext *s, unsigned char *buffer, int buffer_size, int write_flag, void *opaque, int(*read_packet)(void *opaque, uint8_t *buf, int buf_size), int(*write_packet)(void *opaque, uint8_t *buf, int buf_size), int64_t(*seek)(void *opaque, int64_t offset, int whence))
Definition: aviobuf.c:88
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
#define AV_RB32
Definition: intreadwrite.h:130
#define AV_RB16
Definition: intreadwrite.h:53
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
void av_bprintf(AVBPrint *buf, const char *fmt,...)
Definition: bprint.c:94
void av_bprint_init(AVBPrint *buf, unsigned size_init, unsigned size_max)
Definition: bprint.c:69
int av_bprint_finalize(AVBPrint *buf, char **ret_str)
Finalize a print buffer.
Definition: bprint.c:235
#define AV_BPRINT_SIZE_UNLIMITED
static int av_bprint_is_complete(const AVBPrint *buf)
Test if the print buffer is complete (not truncated).
Definition: bprint.h:185
#define flags(name, subs,...)
Definition: cbs_av1.c:561
#define s(width, name)
Definition: cbs_vp9.c:257
#define fail()
Definition: checkasm.h:133
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: codec_par.c:51
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:61
#define FFMIN(a, b)
Definition: common.h:105
#define CONFIG_RTPDEC
Definition: config.h:685
#define CONFIG_RTSP_MUXER
Definition: config.h:2618
#define NULL
Definition: coverity.c:32
Public dictionary API.
static av_always_inline void RENAME() interleave(TYPE *dst, TYPE *src0, TYPE *src1, int w2, int add, int shift)
double value
Definition: eval.c:98
const OptionDef options[]
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:545
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_FLAGS
Definition: opt.h:224
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
@ AV_OPT_TYPE_STRING
Definition: opt.h:229
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
Definition: codec_desc.c:3501
@ AV_CODEC_ID_MPEG2TS
FAKE codec to indicate a raw MPEG-2 TS stream (only used by libavformat)
Definition: codec_id.h:569
@ AV_CODEC_ID_NONE
Definition: codec_id.h:47
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4436
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4509
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4481
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1274
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:4799
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:143
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes to a null-terminated string.
Definition: base64.h:66
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:120
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values.
Definition: dict.c:203
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
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
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:220
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:210
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
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
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
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:253
AVMediaType
Definition: avutil.h:199
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AVMEDIA_TYPE_SUBTITLE
Definition: avutil.h:204
@ AVMEDIA_TYPE_VIDEO
Definition: avutil.h:201
@ AVMEDIA_TYPE_DATA
Opaque data information usually continuous.
Definition: avutil.h:203
@ AVMEDIA_TYPE_UNKNOWN
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:200
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes,...
Definition: avstring.c:93
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:215
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:34
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:83
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:45
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:225
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:260
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:465
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:182
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
@ HTTP_AUTH_NONE
No authentication specified.
Definition: httpauth.h:29
const char * key
int i
Definition: input.c:407
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:23
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:355
void ff_format_set_url(AVFormatContext *s, char *url)
Set AVFormatContext url field to the provided pointer.
Definition: utils.c:5862
#define SPACE_CHARS
Definition: internal.h:499
#define NTP_OFFSET
Definition: internal.h:399
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4945
#define MAX_URL_SIZE
Definition: internal.h:30
static int read_probe(const AVProbeData *pd)
Definition: jvdec.c:55
#define LIBAVFORMAT_IDENT
Definition: version.h:46
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
const char * desc
Definition: libsvtav1.c:79
static void handler(vbi_event *ev, void *user_data)
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:3385
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:3360
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:3339
int ff_network_init(void)
Definition: network.c:58
void ff_network_close(void)
Definition: network.c:116
#define AI_NUMERICHOST
Definition: network.h:187
#define POLLING_TIME
Definition: network.h:249
#define gai_strerror
Definition: network.h:225
#define NI_NUMERICHOST
Definition: network.h:195
#define getaddrinfo
Definition: network.h:217
#define getnameinfo
Definition: network.h:219
#define freeaddrinfo
Definition: network.h:218
AVOptions.
#define AV_OPT_FLAG_DEPRECATED
set if option is deprecated, users should refer to AVOption.help text for more information
Definition: opt.h:295
miscellaneous OS support macros and functions.
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:587
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:751
misc parsing utilities
static const struct PPFilter filters[]
Definition: postprocess.c:134
const char * name
Definition: qsvenc.c:46
mfxU16 profile
Definition: qsvenc.c:45
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
Definition: rdt.c:335
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
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
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
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:515
#define s1
Definition: regdef.h:38
#define th
Definition: regdef.h:75
enum AVMediaType codec_type
Definition: rtp.c:37
int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:71
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
Definition: rtp.c:143
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:132
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
@ RTCP_BYE
Definition: rtp.h:100
#define RTP_PT_PRIVATE
Definition: rtp.h:77
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:163
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:149
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:919
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:618
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:302
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, const RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:611
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:564
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:459
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:906
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:792
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:100
const RTPDynamicProtocolHandler ff_mpegts_dynamic_handler
Definition: rtpdec_mpegts.c:92
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:130
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:528
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:103
static const AVOption sdp_options[]
Definition: rtsp.c:111
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:143
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:57
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:828
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:178
#define COMMON_OPTS()
Definition: rtsp.c:74
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:162
#define ENC
Definition: rtsp.c:61
static const AVOption rtp_options[]
Definition: rtsp.c:121
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:169
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:200
const AVOption ff_rtsp_options[]
Definition: rtsp.c:80
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:792
#define RECVBUF_SIZE
Definition: rtsp.c:56
#define OFFSET(x)
Definition: rtsp.c:59
#define READ_PACKET_TIMEOUT_S
Definition: rtsp.c:55
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:63
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:130
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:760
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:67
#define DEC
Definition: rtsp.c:60
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:427
@ RTSP_SERVER_SATIP
SAT>IP server.
Definition: rtsp.h:218
@ RTSP_SERVER_WMS
Windows Media server.
Definition: rtsp.h:217
@ RTSP_SERVER_RTP
Standards-compliant RTP-server.
Definition: rtsp.h:215
@ RTSP_SERVER_REAL
Realmedia-style server.
Definition: rtsp.h:216
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.
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:78
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:603
#define RTSP_FLAG_SATIP_RAW
Export SAT>IP stream as raw MPEG-TS.
Definition: rtsp.h:432
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
int 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.
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:79
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.
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port.
Definition: rtsp.h:424
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:77
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:428
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:76
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:75
@ RTSP_LOWER_TRANSPORT_TCP
TCP; interleaved in RTSP.
Definition: rtsp.h:40
@ RTSP_LOWER_TRANSPORT_HTTP
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:43
@ RTSP_LOWER_TRANSPORT_NB
Definition: rtsp.h:42
@ RTSP_LOWER_TRANSPORT_UDP_MULTICAST
UDP/multicast.
Definition: rtsp.h:41
@ 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
@ RTSP_LOWER_TRANSPORT_UDP
UDP/unicast.
Definition: rtsp.h:39
@ RTSP_LOWER_TRANSPORT_HTTPS
HTTPS tunneled.
Definition: rtsp.h:46
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:474
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:80
@ RTSP_MODE_PLAIN
Normal RTSP.
Definition: rtsp.h:70
@ RTSP_MODE_TUNNEL
RTSP over HTTP (tunneling)
Definition: rtsp.h:71
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
@ RTSP_STATE_STREAMING
initialized and sending/receiving data
Definition: rtsp.h:205
@ RTSP_STATE_IDLE
not initialized
Definition: rtsp.h:204
#define RTSP_FLAG_PREFER_TCP
Try RTP via TCP first if possible.
Definition: rtsp.h:431
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.
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:141
#define SDP_MAX_SIZE
Definition: rtsp.h:82
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:74
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:774
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:81
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
@ RTSP_TRANSPORT_RTP
Standards-compliant RTP.
Definition: rtsp.h:59
@ RTSP_TRANSPORT_RAW
Raw data (over UDP)
Definition: rtsp.h:61
@ RTSP_TRANSPORT_RDT
Realmedia Data Transport.
Definition: rtsp.h:60
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:45
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:429
@ RTSP_STATUS_OK
Definition: rtspcodes.h:33
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
static const uint8_t header[24]
Definition: sdr2.c:67
#define FF_ARRAY_ELEMS(a)
#define snprintf
Definition: snprintf.h:34
Describe the class of an AVClass context structure.
Definition: log.h:67
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
This struct describes the properties of a single codec described by an AVCodecID.
Definition: codec_desc.h:38
This struct describes the properties of an encoded stream.
Definition: codec_par.h:52
int channels
Audio only.
Definition: codec_par.h:166
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
int sample_rate
Audio only.
Definition: codec_par.h:170
Format I/O context.
Definition: avformat.h:1232
AVIOContext * pb
I/O context.
Definition: avformat.h:1274
Bytestream IO Context.
Definition: avio.h:161
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:645
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
This structure contains the data a format has to probe a file.
Definition: avformat.h:441
const char * filename
Definition: avformat.h:442
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:444
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:443
Stream structure.
Definition: avformat.h:873
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1038
AVDictionary * metadata
Definition: avformat.h:937
int id
Format-specific stream ID.
Definition: avformat.h:880
int index
stream index in AVFormatContext
Definition: avformat.h:874
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
Definition: avformat.h:902
enum AVStreamParseType need_parsing
Definition: avformat.h:1081
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
int auth_type
The currently chosen auth type.
Definition: httpauth.h:59
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:177
uint32_t base_timestamp
Definition: rtpdec.h:154
uint32_t ssrc
Definition: rtpdec.h:151
int64_t rtcp_ts_offset
Definition: rtpdec.h:179
AVStream * st
Definition: rtpdec.h:149
int(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
Definition: rtpdec.h:128
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null.
Definition: rtpdec.h:126
This describes the server response to each RTSP command.
Definition: rtsp.h:130
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:185
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:158
char location[4096]
the "Location:" field.
Definition: rtsp.h:155
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:180
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:141
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:175
char session_id[512]
the "Session:" field.
Definition: rtsp.h:151
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:167
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:134
int seq
sequence number
Definition: rtsp.h:147
char content_type[64]
Content type header.
Definition: rtsp.h:190
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:145
int64_t range_end
Definition: rtsp.h:141
char stream_id[64]
SAT>IP com.ses.streamID header.
Definition: rtsp.h:195
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:137
int content_length
length of the data following this header
Definition: rtsp.h:132
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:435
Private data for the RTSP demuxer.
Definition: rtsp.h:227
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:279
int recvbuf_len
Definition: rtsp.h:332
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:232
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:267
int rtp_port_max
Definition: rtsp.h:397
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:316
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:259
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:382
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:285
int initial_timeout
Timeout to wait for incoming connections.
Definition: rtsp.h:402
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling.
Definition: rtsp.h:337
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:353
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:271
int max_p
Definition: rtsp.h:364
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:264
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:297
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:254
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:392
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:330
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:276
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:348
char * user_agent
User-Agent string.
Definition: rtsp.h:417
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:387
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:248
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:326
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:340
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:292
int buffer_size
Definition: rtsp.h:420
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:412
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:282
URLContext * rtsp_hd
Definition: rtsp.h:229
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:397
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:240
int recvbuf_pos
Definition: rtsp.h:331
int seq
RTSP command sequence number.
Definition: rtsp.h:250
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:407
int pkt_size
Definition: rtsp.h:421
int nb_byes
Definition: rtsp.h:345
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:234
char default_lang[4]
Definition: rtsp.h:419
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:363
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:369
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:288
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:444
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:464
char crypto_suite[40]
Definition: rtsp.h:484
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:465
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:472
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:459
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:461
char crypto_params[100]
Definition: rtsp.h:485
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport.
Definition: rtsp.h:453
char control_url[MAX_URL_SIZE]
url for this stream (from SDP)
Definition: rtsp.h:455
int sdp_payload_type
payload type
Definition: rtsp.h:466
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:445
int interleaved_max
Definition: rtsp.h:453
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:449
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:462
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:479
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:475
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:446
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:482
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:463
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:460
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:91
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:108
int interleaved_max
Definition: rtsp.h:96
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:104
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:118
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:121
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:117
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:112
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:124
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data.
Definition: rtsp.h:100
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:96
Definition: url.h:38
void * priv_data
Definition: url.h:41
const char * protocol_whitelist
Definition: url.h:49
int ai_flags
Definition: network.h:138
Definition: graph2dot.c:48
uint16_t ss_family
Definition: network.h:116
#define av_free(p)
#define av_malloc_array(a, b)
#define av_freep(p)
#define av_malloc(s)
#define av_log(a,...)
AVPacket * pkt
Definition: movenc.c:59
AVDictionary * opts
Definition: movenc.c:50
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
int size
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
unbuffered private I/O API
int len