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rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
32 #include "avformat.h"
33 #include "avio_internal.h"
34 
35 #if HAVE_POLL_H
36 #include <poll.h>
37 #endif
38 #include "internal.h"
39 #include "network.h"
40 #include "os_support.h"
41 #include "http.h"
42 #include "rtsp.h"
43 
44 #include "rtpdec.h"
45 #include "rtpproto.h"
46 #include "rdt.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
49 #include "url.h"
50 #include "rtpenc.h"
51 #include "mpegts.h"
52 
53 /* Timeout values for socket poll, in ms,
54  * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
61 
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 
66 #define RTSP_FLAG_OPTS(name, longname) \
67  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68  { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
69 
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71  { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75  { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
76 
77 #define COMMON_OPTS() \
78  { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
79  { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
80 
81 
83  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
84  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92  { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
93  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96  { "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 },
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 },
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  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
116  COMMON_OPTS(),
117  { NULL },
118 };
119 
120 static const AVOption rtp_options[] = {
121  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
122  COMMON_OPTS(),
123  { NULL },
124 };
125 
126 
128 {
130  char buf[256];
131 
132  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
133  av_dict_set(&opts, "buffer_size", buf, 0);
134 
135  return opts;
136 }
137 
138 static void get_word_until_chars(char *buf, int buf_size,
139  const char *sep, const char **pp)
140 {
141  const char *p;
142  char *q;
143 
144  p = *pp;
145  p += strspn(p, SPACE_CHARS);
146  q = buf;
147  while (!strchr(sep, *p) && *p != '\0') {
148  if ((q - buf) < buf_size - 1)
149  *q++ = *p;
150  p++;
151  }
152  if (buf_size > 0)
153  *q = '\0';
154  *pp = p;
155 }
156 
157 static void get_word_sep(char *buf, int buf_size, const char *sep,
158  const char **pp)
159 {
160  if (**pp == '/') (*pp)++;
161  get_word_until_chars(buf, buf_size, sep, pp);
162 }
163 
164 static void get_word(char *buf, int buf_size, const char **pp)
165 {
166  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
167 }
168 
169 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
170  * and end time.
171  * Used for seeking in the rtp stream.
172  */
173 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
174 {
175  char buf[256];
176 
177  p += strspn(p, SPACE_CHARS);
178  if (!av_stristart(p, "npt=", &p))
179  return;
180 
181  *start = AV_NOPTS_VALUE;
182  *end = AV_NOPTS_VALUE;
183 
184  get_word_sep(buf, sizeof(buf), "-", &p);
185  if (av_parse_time(start, buf, 1) < 0)
186  return;
187  if (*p == '-') {
188  p++;
189  get_word_sep(buf, sizeof(buf), "-", &p);
190  if (av_parse_time(end, buf, 1) < 0)
191  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
192  }
193 }
194 
196  const char *buf, struct sockaddr_storage *sock)
197 {
198  struct addrinfo hints = { 0 }, *ai = NULL;
199  int ret;
200 
201  hints.ai_flags = AI_NUMERICHOST;
202  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
203  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
204  buf,
205  gai_strerror(ret));
206  return -1;
207  }
208  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
209  freeaddrinfo(ai);
210  return 0;
211 }
212 
213 #if CONFIG_RTPDEC
214 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
215  RTSPStream *rtsp_st, AVStream *st)
216 {
217  AVCodecParameters *par = st ? st->codecpar : NULL;
218  if (!handler)
219  return;
220  if (par)
221  par->codec_id = handler->codec_id;
222  rtsp_st->dynamic_handler = handler;
223  if (st)
224  st->need_parsing = handler->need_parsing;
225  if (handler->priv_data_size) {
227  if (!rtsp_st->dynamic_protocol_context)
228  rtsp_st->dynamic_handler = NULL;
229  }
230 }
231 
232 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
233  AVStream *st)
234 {
235  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
236  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
237  rtsp_st->dynamic_protocol_context);
238  if (ret < 0) {
239  if (rtsp_st->dynamic_protocol_context) {
240  if (rtsp_st->dynamic_handler->close)
241  rtsp_st->dynamic_handler->close(
242  rtsp_st->dynamic_protocol_context);
244  }
245  rtsp_st->dynamic_protocol_context = NULL;
246  rtsp_st->dynamic_handler = NULL;
247  }
248  }
249 }
250 
251 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
252 static int sdp_parse_rtpmap(AVFormatContext *s,
253  AVStream *st, RTSPStream *rtsp_st,
254  int payload_type, const char *p)
255 {
256  AVCodecParameters *par = st->codecpar;
257  char buf[256];
258  int i;
259  const AVCodecDescriptor *desc;
260  const char *c_name;
261 
262  /* See if we can handle this kind of payload.
263  * The space should normally not be there but some Real streams or
264  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
265  * have a trailing space. */
266  get_word_sep(buf, sizeof(buf), "/ ", &p);
267  if (payload_type < RTP_PT_PRIVATE) {
268  /* We are in a standard case
269  * (from http://www.iana.org/assignments/rtp-parameters). */
270  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
271  }
272 
273  if (par->codec_id == AV_CODEC_ID_NONE) {
274  const RTPDynamicProtocolHandler *handler =
276  init_rtp_handler(handler, rtsp_st, st);
277  /* If no dynamic handler was found, check with the list of standard
278  * allocated types, if such a stream for some reason happens to
279  * use a private payload type. This isn't handled in rtpdec.c, since
280  * the format name from the rtpmap line never is passed into rtpdec. */
281  if (!rtsp_st->dynamic_handler)
282  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
283  }
284 
285  desc = avcodec_descriptor_get(par->codec_id);
286  if (desc && desc->name)
287  c_name = desc->name;
288  else
289  c_name = "(null)";
290 
291  get_word_sep(buf, sizeof(buf), "/", &p);
292  i = atoi(buf);
293  switch (par->codec_type) {
294  case AVMEDIA_TYPE_AUDIO:
295  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
298  if (i > 0) {
299  par->sample_rate = i;
300  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
301  get_word_sep(buf, sizeof(buf), "/", &p);
302  i = atoi(buf);
303  if (i > 0)
304  par->channels = i;
305  }
306  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
307  par->sample_rate);
308  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
309  par->channels);
310  break;
311  case AVMEDIA_TYPE_VIDEO:
312  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
313  if (i > 0)
314  avpriv_set_pts_info(st, 32, 1, i);
315  break;
316  default:
317  break;
318  }
319  finalize_rtp_handler_init(s, rtsp_st, st);
320  return 0;
321 }
322 
323 /* parse the attribute line from the fmtp a line of an sdp response. This
324  * is broken out as a function because it is used in rtp_h264.c, which is
325  * forthcoming. */
326 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
327  char *value, int value_size)
328 {
329  *p += strspn(*p, SPACE_CHARS);
330  if (**p) {
331  get_word_sep(attr, attr_size, "=", p);
332  if (**p == '=')
333  (*p)++;
334  get_word_sep(value, value_size, ";", p);
335  if (**p == ';')
336  (*p)++;
337  return 1;
338  }
339  return 0;
340 }
341 
342 typedef struct SDPParseState {
343  /* SDP only */
344  struct sockaddr_storage default_ip;
345  int default_ttl;
346  int skip_media; ///< set if an unknown m= line occurs
347  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
348  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
349  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
350  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
351  int seen_rtpmap;
352  int seen_fmtp;
353  char delayed_fmtp[2048];
354 } SDPParseState;
355 
356 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
357  struct RTSPSource ***dest, int *dest_count)
358 {
359  RTSPSource *rtsp_src, *rtsp_src2;
360  int i;
361  for (i = 0; i < count; i++) {
362  rtsp_src = addrs[i];
363  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
364  if (!rtsp_src2)
365  continue;
366  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
367  dynarray_add(dest, dest_count, rtsp_src2);
368  }
369 }
370 
371 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
372  int payload_type, const char *line)
373 {
374  int i;
375 
376  for (i = 0; i < rt->nb_rtsp_streams; i++) {
377  RTSPStream *rtsp_st = rt->rtsp_streams[i];
378  if (rtsp_st->sdp_payload_type == payload_type &&
379  rtsp_st->dynamic_handler &&
380  rtsp_st->dynamic_handler->parse_sdp_a_line) {
381  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
382  rtsp_st->dynamic_protocol_context, line);
383  }
384  }
385 }
386 
387 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
388  int letter, const char *buf)
389 {
390  RTSPState *rt = s->priv_data;
391  char buf1[64], st_type[64];
392  const char *p;
393  enum AVMediaType codec_type;
394  int payload_type;
395  AVStream *st;
396  RTSPStream *rtsp_st;
397  RTSPSource *rtsp_src;
398  struct sockaddr_storage sdp_ip;
399  int ttl;
400 
401  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
402 
403  p = buf;
404  if (s1->skip_media && letter != 'm')
405  return;
406  switch (letter) {
407  case 'c':
408  get_word(buf1, sizeof(buf1), &p);
409  if (strcmp(buf1, "IN") != 0)
410  return;
411  get_word(buf1, sizeof(buf1), &p);
412  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
413  return;
414  get_word_sep(buf1, sizeof(buf1), "/", &p);
415  if (get_sockaddr(s, buf1, &sdp_ip))
416  return;
417  ttl = 16;
418  if (*p == '/') {
419  p++;
420  get_word_sep(buf1, sizeof(buf1), "/", &p);
421  ttl = atoi(buf1);
422  }
423  if (s->nb_streams == 0) {
424  s1->default_ip = sdp_ip;
425  s1->default_ttl = ttl;
426  } else {
427  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
428  rtsp_st->sdp_ip = sdp_ip;
429  rtsp_st->sdp_ttl = ttl;
430  }
431  break;
432  case 's':
433  av_dict_set(&s->metadata, "title", p, 0);
434  break;
435  case 'i':
436  if (s->nb_streams == 0) {
437  av_dict_set(&s->metadata, "comment", p, 0);
438  break;
439  }
440  break;
441  case 'm':
442  /* new stream */
443  s1->skip_media = 0;
444  s1->seen_fmtp = 0;
445  s1->seen_rtpmap = 0;
446  codec_type = AVMEDIA_TYPE_UNKNOWN;
447  get_word(st_type, sizeof(st_type), &p);
448  if (!strcmp(st_type, "audio")) {
449  codec_type = AVMEDIA_TYPE_AUDIO;
450  } else if (!strcmp(st_type, "video")) {
451  codec_type = AVMEDIA_TYPE_VIDEO;
452  } else if (!strcmp(st_type, "application")) {
453  codec_type = AVMEDIA_TYPE_DATA;
454  } else if (!strcmp(st_type, "text")) {
455  codec_type = AVMEDIA_TYPE_SUBTITLE;
456  }
457  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
458  s1->skip_media = 1;
459  return;
460  }
461  rtsp_st = av_mallocz(sizeof(RTSPStream));
462  if (!rtsp_st)
463  return;
464  rtsp_st->stream_index = -1;
465  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
466 
467  rtsp_st->sdp_ip = s1->default_ip;
468  rtsp_st->sdp_ttl = s1->default_ttl;
469 
470  copy_default_source_addrs(s1->default_include_source_addrs,
471  s1->nb_default_include_source_addrs,
472  &rtsp_st->include_source_addrs,
473  &rtsp_st->nb_include_source_addrs);
474  copy_default_source_addrs(s1->default_exclude_source_addrs,
475  s1->nb_default_exclude_source_addrs,
476  &rtsp_st->exclude_source_addrs,
477  &rtsp_st->nb_exclude_source_addrs);
478 
479  get_word(buf1, sizeof(buf1), &p); /* port */
480  rtsp_st->sdp_port = atoi(buf1);
481 
482  get_word(buf1, sizeof(buf1), &p); /* protocol */
483  if (!strcmp(buf1, "udp"))
485  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
486  rtsp_st->feedback = 1;
487 
488  /* XXX: handle list of formats */
489  get_word(buf1, sizeof(buf1), &p); /* format list */
490  rtsp_st->sdp_payload_type = atoi(buf1);
491 
492  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
493  /* no corresponding stream */
494  if (rt->transport == RTSP_TRANSPORT_RAW) {
495  if (CONFIG_RTPDEC && !rt->ts)
496  rt->ts = avpriv_mpegts_parse_open(s);
497  } else {
499  handler = ff_rtp_handler_find_by_id(
501  init_rtp_handler(handler, rtsp_st, NULL);
502  finalize_rtp_handler_init(s, rtsp_st, NULL);
503  }
504  } else if (rt->server_type == RTSP_SERVER_WMS &&
505  codec_type == AVMEDIA_TYPE_DATA) {
506  /* RTX stream, a stream that carries all the other actual
507  * audio/video streams. Don't expose this to the callers. */
508  } else {
509  st = avformat_new_stream(s, NULL);
510  if (!st)
511  return;
512  st->id = rt->nb_rtsp_streams - 1;
513  rtsp_st->stream_index = st->index;
515  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
517  /* if standard payload type, we can find the codec right now */
519  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
520  st->codecpar->sample_rate > 0)
521  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
522  /* Even static payload types may need a custom depacketizer */
523  handler = ff_rtp_handler_find_by_id(
524  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
525  init_rtp_handler(handler, rtsp_st, st);
526  finalize_rtp_handler_init(s, rtsp_st, st);
527  }
528  if (rt->default_lang[0])
529  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
530  }
531  /* put a default control url */
532  av_strlcpy(rtsp_st->control_url, rt->control_uri,
533  sizeof(rtsp_st->control_url));
534  break;
535  case 'a':
536  if (av_strstart(p, "control:", &p)) {
537  if (s->nb_streams == 0) {
538  if (!strncmp(p, "rtsp://", 7))
539  av_strlcpy(rt->control_uri, p,
540  sizeof(rt->control_uri));
541  } else {
542  char proto[32];
543  /* get the control url */
544  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
545 
546  /* XXX: may need to add full url resolution */
547  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
548  NULL, NULL, 0, p);
549  if (proto[0] == '\0') {
550  /* relative control URL */
551  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
552  av_strlcat(rtsp_st->control_url, "/",
553  sizeof(rtsp_st->control_url));
554  av_strlcat(rtsp_st->control_url, p,
555  sizeof(rtsp_st->control_url));
556  } else
557  av_strlcpy(rtsp_st->control_url, p,
558  sizeof(rtsp_st->control_url));
559  }
560  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
561  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
562  get_word(buf1, sizeof(buf1), &p);
563  payload_type = atoi(buf1);
564  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
565  if (rtsp_st->stream_index >= 0) {
566  st = s->streams[rtsp_st->stream_index];
567  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
568  }
569  s1->seen_rtpmap = 1;
570  if (s1->seen_fmtp) {
571  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
572  }
573  } else if (av_strstart(p, "fmtp:", &p) ||
574  av_strstart(p, "framesize:", &p)) {
575  // let dynamic protocol handlers have a stab at the line.
576  get_word(buf1, sizeof(buf1), &p);
577  payload_type = atoi(buf1);
578  if (s1->seen_rtpmap) {
579  parse_fmtp(s, rt, payload_type, buf);
580  } else {
581  s1->seen_fmtp = 1;
582  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
583  }
584  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
585  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
586  get_word(buf1, sizeof(buf1), &p);
587  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
588  } else if (av_strstart(p, "range:", &p)) {
589  int64_t start, end;
590 
591  // this is so that seeking on a streamed file can work.
592  rtsp_parse_range_npt(p, &start, &end);
593  s->start_time = start;
594  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
595  s->duration = (end == AV_NOPTS_VALUE) ?
596  AV_NOPTS_VALUE : end - start;
597  } else if (av_strstart(p, "lang:", &p)) {
598  if (s->nb_streams > 0) {
599  get_word(buf1, sizeof(buf1), &p);
600  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
601  if (rtsp_st->stream_index >= 0) {
602  st = s->streams[rtsp_st->stream_index];
603  av_dict_set(&st->metadata, "language", buf1, 0);
604  }
605  } else
606  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
607  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
608  if (atoi(p) == 1)
610  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
611  s->nb_streams > 0) {
612  st = s->streams[s->nb_streams - 1];
613  st->codecpar->sample_rate = atoi(p);
614  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
615  // RFC 4568
616  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
617  get_word(buf1, sizeof(buf1), &p); // ignore tag
618  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
619  p += strspn(p, SPACE_CHARS);
620  if (av_strstart(p, "inline:", &p))
621  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
622  } else if (av_strstart(p, "source-filter:", &p)) {
623  int exclude = 0;
624  get_word(buf1, sizeof(buf1), &p);
625  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
626  return;
627  exclude = !strcmp(buf1, "excl");
628 
629  get_word(buf1, sizeof(buf1), &p);
630  if (strcmp(buf1, "IN") != 0)
631  return;
632  get_word(buf1, sizeof(buf1), &p);
633  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
634  return;
635  // not checking that the destination address actually matches or is wildcard
636  get_word(buf1, sizeof(buf1), &p);
637 
638  while (*p != '\0') {
639  rtsp_src = av_mallocz(sizeof(*rtsp_src));
640  if (!rtsp_src)
641  return;
642  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
643  if (exclude) {
644  if (s->nb_streams == 0) {
645  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
646  } else {
647  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
648  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
649  }
650  } else {
651  if (s->nb_streams == 0) {
652  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
653  } else {
654  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
655  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
656  }
657  }
658  }
659  } else {
660  if (rt->server_type == RTSP_SERVER_WMS)
662  if (s->nb_streams > 0) {
663  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
664 
665  if (rt->server_type == RTSP_SERVER_REAL)
666  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
667 
668  if (rtsp_st->dynamic_handler &&
670  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
671  rtsp_st->stream_index,
672  rtsp_st->dynamic_protocol_context, buf);
673  }
674  }
675  break;
676  }
677 }
678 
679 int ff_sdp_parse(AVFormatContext *s, const char *content)
680 {
681  const char *p;
682  int letter, i;
683  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
684  * contain long SDP lines containing complete ASF Headers (several
685  * kB) or arrays of MDPR (RM stream descriptor) headers plus
686  * "rulebooks" describing their properties. Therefore, the SDP line
687  * buffer is large.
688  *
689  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
690  * in rtpdec_xiph.c. */
691  char buf[16384], *q;
692  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
693 
694  p = content;
695  for (;;) {
696  p += strspn(p, SPACE_CHARS);
697  letter = *p;
698  if (letter == '\0')
699  break;
700  p++;
701  if (*p != '=')
702  goto next_line;
703  p++;
704  /* get the content */
705  q = buf;
706  while (*p != '\n' && *p != '\r' && *p != '\0') {
707  if ((q - buf) < sizeof(buf) - 1)
708  *q++ = *p;
709  p++;
710  }
711  *q = '\0';
712  sdp_parse_line(s, s1, letter, buf);
713  next_line:
714  while (*p != '\n' && *p != '\0')
715  p++;
716  if (*p == '\n')
717  p++;
718  }
719 
720  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
721  av_freep(&s1->default_include_source_addrs[i]);
722  av_freep(&s1->default_include_source_addrs);
723  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
724  av_freep(&s1->default_exclude_source_addrs[i]);
725  av_freep(&s1->default_exclude_source_addrs);
726 
727  return 0;
728 }
729 #endif /* CONFIG_RTPDEC */
730 
731 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
732 {
733  RTSPState *rt = s->priv_data;
734  int i;
735 
736  for (i = 0; i < rt->nb_rtsp_streams; i++) {
737  RTSPStream *rtsp_st = rt->rtsp_streams[i];
738  if (!rtsp_st)
739  continue;
740  if (rtsp_st->transport_priv) {
741  if (s->oformat) {
742  AVFormatContext *rtpctx = rtsp_st->transport_priv;
743  av_write_trailer(rtpctx);
745  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
746  ff_rtsp_tcp_write_packet(s, rtsp_st);
747  ffio_free_dyn_buf(&rtpctx->pb);
748  } else {
749  avio_closep(&rtpctx->pb);
750  }
751  avformat_free_context(rtpctx);
752  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
754  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
756  }
757  rtsp_st->transport_priv = NULL;
758  if (rtsp_st->rtp_handle)
759  ffurl_close(rtsp_st->rtp_handle);
760  rtsp_st->rtp_handle = NULL;
761  }
762 }
763 
764 /* close and free RTSP streams */
766 {
767  RTSPState *rt = s->priv_data;
768  int i, j;
769  RTSPStream *rtsp_st;
770 
771  ff_rtsp_undo_setup(s, 0);
772  for (i = 0; i < rt->nb_rtsp_streams; i++) {
773  rtsp_st = rt->rtsp_streams[i];
774  if (rtsp_st) {
775  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
776  if (rtsp_st->dynamic_handler->close)
777  rtsp_st->dynamic_handler->close(
778  rtsp_st->dynamic_protocol_context);
780  }
781  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
782  av_freep(&rtsp_st->include_source_addrs[j]);
783  av_freep(&rtsp_st->include_source_addrs);
784  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
785  av_freep(&rtsp_st->exclude_source_addrs[j]);
786  av_freep(&rtsp_st->exclude_source_addrs);
787 
788  av_freep(&rtsp_st);
789  }
790  }
791  av_freep(&rt->rtsp_streams);
792  if (rt->asf_ctx) {
794  }
795  if (CONFIG_RTPDEC && rt->ts)
797  av_freep(&rt->p);
798  av_freep(&rt->recvbuf);
799 }
800 
802 {
803  RTSPState *rt = s->priv_data;
804  AVStream *st = NULL;
805  int reordering_queue_size = rt->reordering_queue_size;
806  if (reordering_queue_size < 0) {
808  reordering_queue_size = 0;
809  else
810  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
811  }
812 
813  /* open the RTP context */
814  if (rtsp_st->stream_index >= 0)
815  st = s->streams[rtsp_st->stream_index];
816  if (!st)
818 
819  if (CONFIG_RTSP_MUXER && s->oformat && st) {
820  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
821  s, st, rtsp_st->rtp_handle,
823  rtsp_st->stream_index);
824  /* Ownership of rtp_handle is passed to the rtp mux context */
825  rtsp_st->rtp_handle = NULL;
826  if (ret < 0)
827  return ret;
828  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
829  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
830  return 0; // Don't need to open any parser here
831  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
832  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
833  rtsp_st->dynamic_protocol_context,
834  rtsp_st->dynamic_handler);
835  else if (CONFIG_RTPDEC)
836  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
837  rtsp_st->sdp_payload_type,
838  reordering_queue_size);
839 
840  if (!rtsp_st->transport_priv) {
841  return AVERROR(ENOMEM);
842  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
843  s->iformat) {
844  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
845  rtpctx->ssrc = rtsp_st->ssrc;
846  if (rtsp_st->dynamic_handler) {
848  rtsp_st->dynamic_protocol_context,
849  rtsp_st->dynamic_handler);
850  }
851  if (rtsp_st->crypto_suite[0])
853  rtsp_st->crypto_suite,
854  rtsp_st->crypto_params);
855  }
856 
857  return 0;
858 }
859 
860 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
861 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
862 {
863  const char *q;
864  char *p;
865  int v;
866 
867  q = *pp;
868  q += strspn(q, SPACE_CHARS);
869  v = strtol(q, &p, 10);
870  if (*p == '-') {
871  p++;
872  *min_ptr = v;
873  v = strtol(p, &p, 10);
874  *max_ptr = v;
875  } else {
876  *min_ptr = v;
877  *max_ptr = v;
878  }
879  *pp = p;
880 }
881 
882 /* XXX: only one transport specification is parsed */
883 static void rtsp_parse_transport(AVFormatContext *s,
884  RTSPMessageHeader *reply, const char *p)
885 {
886  char transport_protocol[16];
887  char profile[16];
888  char lower_transport[16];
889  char parameter[16];
891  char buf[256];
892 
893  reply->nb_transports = 0;
894 
895  for (;;) {
896  p += strspn(p, SPACE_CHARS);
897  if (*p == '\0')
898  break;
899 
900  th = &reply->transports[reply->nb_transports];
901 
902  get_word_sep(transport_protocol, sizeof(transport_protocol),
903  "/", &p);
904  if (!av_strcasecmp (transport_protocol, "rtp")) {
905  get_word_sep(profile, sizeof(profile), "/;,", &p);
906  lower_transport[0] = '\0';
907  /* rtp/avp/<protocol> */
908  if (*p == '/') {
909  get_word_sep(lower_transport, sizeof(lower_transport),
910  ";,", &p);
911  }
913  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
914  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
915  /* x-pn-tng/<protocol> */
916  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
917  profile[0] = '\0';
919  } else if (!av_strcasecmp(transport_protocol, "raw")) {
920  get_word_sep(profile, sizeof(profile), "/;,", &p);
921  lower_transport[0] = '\0';
922  /* raw/raw/<protocol> */
923  if (*p == '/') {
924  get_word_sep(lower_transport, sizeof(lower_transport),
925  ";,", &p);
926  }
928  }
929  if (!av_strcasecmp(lower_transport, "TCP"))
931  else
933 
934  if (*p == ';')
935  p++;
936  /* get each parameter */
937  while (*p != '\0' && *p != ',') {
938  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
939  if (!strcmp(parameter, "port")) {
940  if (*p == '=') {
941  p++;
942  rtsp_parse_range(&th->port_min, &th->port_max, &p);
943  }
944  } else if (!strcmp(parameter, "client_port")) {
945  if (*p == '=') {
946  p++;
947  rtsp_parse_range(&th->client_port_min,
948  &th->client_port_max, &p);
949  }
950  } else if (!strcmp(parameter, "server_port")) {
951  if (*p == '=') {
952  p++;
953  rtsp_parse_range(&th->server_port_min,
954  &th->server_port_max, &p);
955  }
956  } else if (!strcmp(parameter, "interleaved")) {
957  if (*p == '=') {
958  p++;
959  rtsp_parse_range(&th->interleaved_min,
960  &th->interleaved_max, &p);
961  }
962  } else if (!strcmp(parameter, "multicast")) {
965  } else if (!strcmp(parameter, "ttl")) {
966  if (*p == '=') {
967  char *end;
968  p++;
969  th->ttl = strtol(p, &end, 10);
970  p = end;
971  }
972  } else if (!strcmp(parameter, "destination")) {
973  if (*p == '=') {
974  p++;
975  get_word_sep(buf, sizeof(buf), ";,", &p);
976  get_sockaddr(s, buf, &th->destination);
977  }
978  } else if (!strcmp(parameter, "source")) {
979  if (*p == '=') {
980  p++;
981  get_word_sep(buf, sizeof(buf), ";,", &p);
982  av_strlcpy(th->source, buf, sizeof(th->source));
983  }
984  } else if (!strcmp(parameter, "mode")) {
985  if (*p == '=') {
986  p++;
987  get_word_sep(buf, sizeof(buf), ";, ", &p);
988  if (!strcmp(buf, "record") ||
989  !strcmp(buf, "receive"))
990  th->mode_record = 1;
991  }
992  }
993 
994  while (*p != ';' && *p != '\0' && *p != ',')
995  p++;
996  if (*p == ';')
997  p++;
998  }
999  if (*p == ',')
1000  p++;
1001 
1002  reply->nb_transports++;
1003  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1004  break;
1005  }
1006 }
1007 
1008 static void handle_rtp_info(RTSPState *rt, const char *url,
1009  uint32_t seq, uint32_t rtptime)
1010 {
1011  int i;
1012  if (!rtptime || !url[0])
1013  return;
1014  if (rt->transport != RTSP_TRANSPORT_RTP)
1015  return;
1016  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1017  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1018  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1019  if (!rtpctx)
1020  continue;
1021  if (!strcmp(rtsp_st->control_url, url)) {
1022  rtpctx->base_timestamp = rtptime;
1023  break;
1024  }
1025  }
1026 }
1027 
1028 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1029 {
1030  int read = 0;
1031  char key[20], value[1024], url[1024] = "";
1032  uint32_t seq = 0, rtptime = 0;
1033 
1034  for (;;) {
1035  p += strspn(p, SPACE_CHARS);
1036  if (!*p)
1037  break;
1038  get_word_sep(key, sizeof(key), "=", &p);
1039  if (*p != '=')
1040  break;
1041  p++;
1042  get_word_sep(value, sizeof(value), ";, ", &p);
1043  read++;
1044  if (!strcmp(key, "url"))
1045  av_strlcpy(url, value, sizeof(url));
1046  else if (!strcmp(key, "seq"))
1047  seq = strtoul(value, NULL, 10);
1048  else if (!strcmp(key, "rtptime"))
1049  rtptime = strtoul(value, NULL, 10);
1050  if (*p == ',') {
1051  handle_rtp_info(rt, url, seq, rtptime);
1052  url[0] = '\0';
1053  seq = rtptime = 0;
1054  read = 0;
1055  }
1056  if (*p)
1057  p++;
1058  }
1059  if (read > 0)
1060  handle_rtp_info(rt, url, seq, rtptime);
1061 }
1062 
1064  RTSPMessageHeader *reply, const char *buf,
1065  RTSPState *rt, const char *method)
1066 {
1067  const char *p;
1068 
1069  /* NOTE: we do case independent match for broken servers */
1070  p = buf;
1071  if (av_stristart(p, "Session:", &p)) {
1072  int t;
1073  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1074  if (av_stristart(p, ";timeout=", &p) &&
1075  (t = strtol(p, NULL, 10)) > 0) {
1076  reply->timeout = t;
1077  }
1078  } else if (av_stristart(p, "Content-Length:", &p)) {
1079  reply->content_length = strtol(p, NULL, 10);
1080  } else if (av_stristart(p, "Transport:", &p)) {
1081  rtsp_parse_transport(s, reply, p);
1082  } else if (av_stristart(p, "CSeq:", &p)) {
1083  reply->seq = strtol(p, NULL, 10);
1084  } else if (av_stristart(p, "Range:", &p)) {
1085  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1086  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1087  p += strspn(p, SPACE_CHARS);
1088  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1089  } else if (av_stristart(p, "Server:", &p)) {
1090  p += strspn(p, SPACE_CHARS);
1091  av_strlcpy(reply->server, p, sizeof(reply->server));
1092  } else if (av_stristart(p, "Notice:", &p) ||
1093  av_stristart(p, "X-Notice:", &p)) {
1094  reply->notice = strtol(p, NULL, 10);
1095  } else if (av_stristart(p, "Location:", &p)) {
1096  p += strspn(p, SPACE_CHARS);
1097  av_strlcpy(reply->location, p , sizeof(reply->location));
1098  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1099  p += strspn(p, SPACE_CHARS);
1100  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1101  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1102  p += strspn(p, SPACE_CHARS);
1103  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1104  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1105  p += strspn(p, SPACE_CHARS);
1106  if (method && !strcmp(method, "DESCRIBE"))
1107  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1108  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1109  p += strspn(p, SPACE_CHARS);
1110  if (method && !strcmp(method, "PLAY"))
1111  rtsp_parse_rtp_info(rt, p);
1112  } else if (av_stristart(p, "Public:", &p) && rt) {
1113  if (strstr(p, "GET_PARAMETER") &&
1114  method && !strcmp(method, "OPTIONS"))
1115  rt->get_parameter_supported = 1;
1116  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1117  p += strspn(p, SPACE_CHARS);
1118  rt->accept_dynamic_rate = atoi(p);
1119  } else if (av_stristart(p, "Content-Type:", &p)) {
1120  p += strspn(p, SPACE_CHARS);
1121  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1122  }
1123 }
1124 
1125 /* skip a RTP/TCP interleaved packet */
1127 {
1128  RTSPState *rt = s->priv_data;
1129  int ret, len, len1;
1130  uint8_t buf[1024];
1131 
1132  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1133  if (ret != 3)
1134  return;
1135  len = AV_RB16(buf + 1);
1136 
1137  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1138 
1139  /* skip payload */
1140  while (len > 0) {
1141  len1 = len;
1142  if (len1 > sizeof(buf))
1143  len1 = sizeof(buf);
1144  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1145  if (ret != len1)
1146  return;
1147  len -= len1;
1148  }
1149 }
1150 
1152  unsigned char **content_ptr,
1153  int return_on_interleaved_data, const char *method)
1154 {
1155  RTSPState *rt = s->priv_data;
1156  char buf[4096], buf1[1024], *q;
1157  unsigned char ch;
1158  const char *p;
1159  int ret, content_length, line_count = 0, request = 0;
1160  unsigned char *content = NULL;
1161 
1162 start:
1163  line_count = 0;
1164  request = 0;
1165  content = NULL;
1166  memset(reply, 0, sizeof(*reply));
1167 
1168  /* parse reply (XXX: use buffers) */
1169  rt->last_reply[0] = '\0';
1170  for (;;) {
1171  q = buf;
1172  for (;;) {
1173  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1174  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1175  if (ret != 1)
1176  return AVERROR_EOF;
1177  if (ch == '\n')
1178  break;
1179  if (ch == '$' && q == buf) {
1180  if (return_on_interleaved_data) {
1181  return 1;
1182  } else
1184  } else if (ch != '\r') {
1185  if ((q - buf) < sizeof(buf) - 1)
1186  *q++ = ch;
1187  }
1188  }
1189  *q = '\0';
1190 
1191  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1192 
1193  /* test if last line */
1194  if (buf[0] == '\0')
1195  break;
1196  p = buf;
1197  if (line_count == 0) {
1198  /* get reply code */
1199  get_word(buf1, sizeof(buf1), &p);
1200  if (!strncmp(buf1, "RTSP/", 5)) {
1201  get_word(buf1, sizeof(buf1), &p);
1202  reply->status_code = atoi(buf1);
1203  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1204  } else {
1205  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1206  get_word(buf1, sizeof(buf1), &p); // object
1207  request = 1;
1208  }
1209  } else {
1210  ff_rtsp_parse_line(s, reply, p, rt, method);
1211  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1212  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1213  }
1214  line_count++;
1215  }
1216 
1217  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1218  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1219 
1220  content_length = reply->content_length;
1221  if (content_length > 0) {
1222  /* leave some room for a trailing '\0' (useful for simple parsing) */
1223  content = av_malloc(content_length + 1);
1224  if (!content)
1225  return AVERROR(ENOMEM);
1226  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1227  content[content_length] = '\0';
1228  }
1229  if (content_ptr)
1230  *content_ptr = content;
1231  else
1232  av_freep(&content);
1233 
1234  if (request) {
1235  char buf[1024];
1236  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1237  const char* ptr = buf;
1238 
1239  if (!strcmp(reply->reason, "OPTIONS")) {
1240  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1241  if (reply->seq)
1242  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1243  if (reply->session_id[0])
1244  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1245  reply->session_id);
1246  } else {
1247  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1248  }
1249  av_strlcat(buf, "\r\n", sizeof(buf));
1250 
1251  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1252  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1253  ptr = base64buf;
1254  }
1255  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1256 
1258  /* Even if the request from the server had data, it is not the data
1259  * that the caller wants or expects. The memory could also be leaked
1260  * if the actual following reply has content data. */
1261  if (content_ptr)
1262  av_freep(content_ptr);
1263  /* If method is set, this is called from ff_rtsp_send_cmd,
1264  * where a reply to exactly this request is awaited. For
1265  * callers from within packet receiving, we just want to
1266  * return to the caller and go back to receiving packets. */
1267  if (method)
1268  goto start;
1269  return 0;
1270  }
1271 
1272  if (rt->seq != reply->seq) {
1273  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1274  rt->seq, reply->seq);
1275  }
1276 
1277  /* EOS */
1278  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1279  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1280  reply->notice == 2306 /* Continuous Feed Terminated */) {
1281  rt->state = RTSP_STATE_IDLE;
1282  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1283  return AVERROR(EIO); /* data or server error */
1284  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1285  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1286  return AVERROR(EPERM);
1287 
1288  return 0;
1289 }
1290 
1291 /**
1292  * Send a command to the RTSP server without waiting for the reply.
1293  *
1294  * @param s RTSP (de)muxer context
1295  * @param method the method for the request
1296  * @param url the target url for the request
1297  * @param headers extra header lines to include in the request
1298  * @param send_content if non-null, the data to send as request body content
1299  * @param send_content_length the length of the send_content data, or 0 if
1300  * send_content is null
1301  *
1302  * @return zero if success, nonzero otherwise
1303  */
1304 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1305  const char *method, const char *url,
1306  const char *headers,
1307  const unsigned char *send_content,
1308  int send_content_length)
1309 {
1310  RTSPState *rt = s->priv_data;
1311  char buf[4096], *out_buf;
1312  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1313 
1314  /* Add in RTSP headers */
1315  out_buf = buf;
1316  rt->seq++;
1317  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1318  if (headers)
1319  av_strlcat(buf, headers, sizeof(buf));
1320  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1321  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1322  if (rt->session_id[0] != '\0' && (!headers ||
1323  !strstr(headers, "\nIf-Match:"))) {
1324  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1325  }
1326  if (rt->auth[0]) {
1327  char *str = ff_http_auth_create_response(&rt->auth_state,
1328  rt->auth, url, method);
1329  if (str)
1330  av_strlcat(buf, str, sizeof(buf));
1331  av_free(str);
1332  }
1333  if (send_content_length > 0 && send_content)
1334  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1335  av_strlcat(buf, "\r\n", sizeof(buf));
1336 
1337  /* base64 encode rtsp if tunneling */
1338  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1339  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1340  out_buf = base64buf;
1341  }
1342 
1343  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1344 
1345  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1346  if (send_content_length > 0 && send_content) {
1347  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1348  avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1349  return AVERROR_PATCHWELCOME;
1350  }
1351  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1352  }
1354 
1355  return 0;
1356 }
1357 
1358 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1359  const char *url, const char *headers)
1360 {
1361  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1362 }
1363 
1364 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1365  const char *headers, RTSPMessageHeader *reply,
1366  unsigned char **content_ptr)
1367 {
1368  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1369  content_ptr, NULL, 0);
1370 }
1371 
1373  const char *method, const char *url,
1374  const char *header,
1375  RTSPMessageHeader *reply,
1376  unsigned char **content_ptr,
1377  const unsigned char *send_content,
1378  int send_content_length)
1379 {
1380  RTSPState *rt = s->priv_data;
1381  HTTPAuthType cur_auth_type;
1382  int ret, attempts = 0;
1383 
1384 retry:
1385  cur_auth_type = rt->auth_state.auth_type;
1386  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1387  send_content,
1388  send_content_length)))
1389  return ret;
1390 
1391  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1392  return ret;
1393  attempts++;
1394 
1395  if (reply->status_code == 401 &&
1396  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1397  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1398  goto retry;
1399 
1400  if (reply->status_code > 400){
1401  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1402  method,
1403  reply->status_code,
1404  reply->reason);
1405  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1406  }
1407 
1408  return 0;
1409 }
1410 
1411 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1412  int lower_transport, const char *real_challenge)
1413 {
1414  RTSPState *rt = s->priv_data;
1415  int rtx = 0, j, i, err, interleave = 0, port_off;
1416  RTSPStream *rtsp_st;
1417  RTSPMessageHeader reply1, *reply = &reply1;
1418  char cmd[2048];
1419  const char *trans_pref;
1420 
1421  if (rt->transport == RTSP_TRANSPORT_RDT)
1422  trans_pref = "x-pn-tng";
1423  else if (rt->transport == RTSP_TRANSPORT_RAW)
1424  trans_pref = "RAW/RAW";
1425  else
1426  trans_pref = "RTP/AVP";
1427 
1428  /* default timeout: 1 minute */
1429  rt->timeout = 60;
1430 
1431  /* Choose a random starting offset within the first half of the
1432  * port range, to allow for a number of ports to try even if the offset
1433  * happens to be at the end of the random range. */
1434  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1435  /* even random offset */
1436  port_off -= port_off & 0x01;
1437 
1438  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1439  char transport[2048];
1440 
1441  /*
1442  * WMS serves all UDP data over a single connection, the RTX, which
1443  * isn't necessarily the first in the SDP but has to be the first
1444  * to be set up, else the second/third SETUP will fail with a 461.
1445  */
1446  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1447  rt->server_type == RTSP_SERVER_WMS) {
1448  if (i == 0) {
1449  /* rtx first */
1450  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1451  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1452  if (len >= 4 &&
1453  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1454  "/rtx"))
1455  break;
1456  }
1457  if (rtx == rt->nb_rtsp_streams)
1458  return -1; /* no RTX found */
1459  rtsp_st = rt->rtsp_streams[rtx];
1460  } else
1461  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1462  } else
1463  rtsp_st = rt->rtsp_streams[i];
1464 
1465  /* RTP/UDP */
1466  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1467  char buf[256];
1468 
1469  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1470  port = reply->transports[0].client_port_min;
1471  goto have_port;
1472  }
1473 
1474  /* first try in specified port range */
1475  while (j <= rt->rtp_port_max) {
1476  AVDictionary *opts = map_to_opts(rt);
1477 
1478  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1479  "?localport=%d", j);
1480  /* we will use two ports per rtp stream (rtp and rtcp) */
1481  j += 2;
1484 
1485  av_dict_free(&opts);
1486 
1487  if (!err)
1488  goto rtp_opened;
1489  }
1490  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1491  err = AVERROR(EIO);
1492  goto fail;
1493 
1494  rtp_opened:
1495  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1496  have_port:
1497  snprintf(transport, sizeof(transport) - 1,
1498  "%s/UDP;", trans_pref);
1499  if (rt->server_type != RTSP_SERVER_REAL)
1500  av_strlcat(transport, "unicast;", sizeof(transport));
1501  av_strlcatf(transport, sizeof(transport),
1502  "client_port=%d", port);
1503  if (rt->transport == RTSP_TRANSPORT_RTP &&
1504  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1505  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1506  }
1507 
1508  /* RTP/TCP */
1509  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1510  /* For WMS streams, the application streams are only used for
1511  * UDP. When trying to set it up for TCP streams, the server
1512  * will return an error. Therefore, we skip those streams. */
1513  if (rt->server_type == RTSP_SERVER_WMS &&
1514  (rtsp_st->stream_index < 0 ||
1515  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1517  continue;
1518  snprintf(transport, sizeof(transport) - 1,
1519  "%s/TCP;", trans_pref);
1520  if (rt->transport != RTSP_TRANSPORT_RDT)
1521  av_strlcat(transport, "unicast;", sizeof(transport));
1522  av_strlcatf(transport, sizeof(transport),
1523  "interleaved=%d-%d",
1524  interleave, interleave + 1);
1525  interleave += 2;
1526  }
1527 
1528  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1529  snprintf(transport, sizeof(transport) - 1,
1530  "%s/UDP;multicast", trans_pref);
1531  }
1532  if (s->oformat) {
1533  av_strlcat(transport, ";mode=record", sizeof(transport));
1534  } else if (rt->server_type == RTSP_SERVER_REAL ||
1536  av_strlcat(transport, ";mode=play", sizeof(transport));
1537  snprintf(cmd, sizeof(cmd),
1538  "Transport: %s\r\n",
1539  transport);
1540  if (rt->accept_dynamic_rate)
1541  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1542  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1543  char real_res[41], real_csum[9];
1544  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1545  real_challenge);
1546  av_strlcatf(cmd, sizeof(cmd),
1547  "If-Match: %s\r\n"
1548  "RealChallenge2: %s, sd=%s\r\n",
1549  rt->session_id, real_res, real_csum);
1550  }
1551  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1552  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1553  err = 1;
1554  goto fail;
1555  } else if (reply->status_code != RTSP_STATUS_OK ||
1556  reply->nb_transports != 1) {
1558  goto fail;
1559  }
1560 
1561  /* XXX: same protocol for all streams is required */
1562  if (i > 0) {
1563  if (reply->transports[0].lower_transport != rt->lower_transport ||
1564  reply->transports[0].transport != rt->transport) {
1565  err = AVERROR_INVALIDDATA;
1566  goto fail;
1567  }
1568  } else {
1569  rt->lower_transport = reply->transports[0].lower_transport;
1570  rt->transport = reply->transports[0].transport;
1571  }
1572 
1573  /* Fail if the server responded with another lower transport mode
1574  * than what we requested. */
1575  if (reply->transports[0].lower_transport != lower_transport) {
1576  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1577  err = AVERROR_INVALIDDATA;
1578  goto fail;
1579  }
1580 
1581  switch(reply->transports[0].lower_transport) {
1583  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1584  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1585  break;
1586 
1587  case RTSP_LOWER_TRANSPORT_UDP: {
1588  char url[1024], options[30] = "";
1589  const char *peer = host;
1590 
1591  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1592  av_strlcpy(options, "?connect=1", sizeof(options));
1593  /* Use source address if specified */
1594  if (reply->transports[0].source[0])
1595  peer = reply->transports[0].source;
1596  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1597  reply->transports[0].server_port_min, "%s", options);
1598  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1599  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1600  err = AVERROR_INVALIDDATA;
1601  goto fail;
1602  }
1603  break;
1604  }
1606  char url[1024], namebuf[50], optbuf[20] = "";
1607  struct sockaddr_storage addr;
1608  int port, ttl;
1609 
1610  if (reply->transports[0].destination.ss_family) {
1611  addr = reply->transports[0].destination;
1612  port = reply->transports[0].port_min;
1613  ttl = reply->transports[0].ttl;
1614  } else {
1615  addr = rtsp_st->sdp_ip;
1616  port = rtsp_st->sdp_port;
1617  ttl = rtsp_st->sdp_ttl;
1618  }
1619  if (ttl > 0)
1620  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1621  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1622  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1623  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1624  port, "%s", optbuf);
1627  err = AVERROR_INVALIDDATA;
1628  goto fail;
1629  }
1630  break;
1631  }
1632  }
1633 
1634  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1635  goto fail;
1636  }
1637 
1638  if (rt->nb_rtsp_streams && reply->timeout > 0)
1639  rt->timeout = reply->timeout;
1640 
1641  if (rt->server_type == RTSP_SERVER_REAL)
1642  rt->need_subscription = 1;
1643 
1644  return 0;
1645 
1646 fail:
1647  ff_rtsp_undo_setup(s, 0);
1648  return err;
1649 }
1650 
1652 {
1653  RTSPState *rt = s->priv_data;
1654  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1655  ffurl_close(rt->rtsp_hd);
1656  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1657 }
1658 
1660 {
1661  RTSPState *rt = s->priv_data;
1662  char proto[128], host[1024], path[1024];
1663  char tcpname[1024], cmd[2048], auth[128];
1664  const char *lower_rtsp_proto = "tcp";
1665  int port, err, tcp_fd;
1666  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1667  int lower_transport_mask = 0;
1668  int default_port = RTSP_DEFAULT_PORT;
1669  char real_challenge[64] = "";
1670  struct sockaddr_storage peer;
1671  socklen_t peer_len = sizeof(peer);
1672 
1673  if (rt->rtp_port_max < rt->rtp_port_min) {
1674  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1675  "than min port %d\n", rt->rtp_port_max,
1676  rt->rtp_port_min);
1677  return AVERROR(EINVAL);
1678  }
1679 
1680  if (!ff_network_init())
1681  return AVERROR(EIO);
1682 
1683  if (s->max_delay < 0) /* Not set by the caller */
1685 
1690  }
1691  /* Only pass through valid flags from here */
1693 
1694 redirect:
1695  /* extract hostname and port */
1696  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1697  host, sizeof(host), &port, path, sizeof(path), s->url);
1698 
1699  if (!strcmp(proto, "rtsps")) {
1700  lower_rtsp_proto = "tls";
1701  default_port = RTSPS_DEFAULT_PORT;
1703  }
1704 
1705  if (*auth) {
1706  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1707  }
1708  if (port < 0)
1709  port = default_port;
1710 
1711  lower_transport_mask = rt->lower_transport_mask;
1712 
1713  if (!lower_transport_mask)
1714  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1715 
1716  if (s->oformat) {
1717  /* Only UDP or TCP - UDP multicast isn't supported. */
1718  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1719  (1 << RTSP_LOWER_TRANSPORT_TCP);
1720  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1721  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1722  "only UDP and TCP are supported for output.\n");
1723  err = AVERROR(EINVAL);
1724  goto fail;
1725  }
1726  }
1727 
1728  /* Construct the URI used in request; this is similar to s->url,
1729  * but with authentication credentials removed and RTSP specific options
1730  * stripped out. */
1731  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1732  host, port, "%s", path);
1733 
1734  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1735  /* set up initial handshake for tunneling */
1736  char httpname[1024];
1737  char sessioncookie[17];
1738  char headers[1024];
1739 
1740  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1741  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1743 
1744  /* GET requests */
1745  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1746  &s->interrupt_callback) < 0) {
1747  err = AVERROR(EIO);
1748  goto fail;
1749  }
1750 
1751  /* generate GET headers */
1752  snprintf(headers, sizeof(headers),
1753  "x-sessioncookie: %s\r\n"
1754  "Accept: application/x-rtsp-tunnelled\r\n"
1755  "Pragma: no-cache\r\n"
1756  "Cache-Control: no-cache\r\n",
1757  sessioncookie);
1758  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1759 
1760  if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1762  if (!rt->rtsp_hd->protocol_whitelist) {
1763  err = AVERROR(ENOMEM);
1764  goto fail;
1765  }
1766  }
1767 
1768  /* complete the connection */
1769  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1770  err = AVERROR(EIO);
1771  goto fail;
1772  }
1773 
1774  /* POST requests */
1775  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1776  &s->interrupt_callback) < 0 ) {
1777  err = AVERROR(EIO);
1778  goto fail;
1779  }
1780 
1781  /* generate POST headers */
1782  snprintf(headers, sizeof(headers),
1783  "x-sessioncookie: %s\r\n"
1784  "Content-Type: application/x-rtsp-tunnelled\r\n"
1785  "Pragma: no-cache\r\n"
1786  "Cache-Control: no-cache\r\n"
1787  "Content-Length: 32767\r\n"
1788  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1789  sessioncookie);
1790  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1791  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1792 
1793  /* Initialize the authentication state for the POST session. The HTTP
1794  * protocol implementation doesn't properly handle multi-pass
1795  * authentication for POST requests, since it would require one of
1796  * the following:
1797  * - implementing Expect: 100-continue, which many HTTP servers
1798  * don't support anyway, even less the RTSP servers that do HTTP
1799  * tunneling
1800  * - sending the whole POST data until getting a 401 reply specifying
1801  * what authentication method to use, then resending all that data
1802  * - waiting for potential 401 replies directly after sending the
1803  * POST header (waiting for some unspecified time)
1804  * Therefore, we copy the full auth state, which works for both basic
1805  * and digest. (For digest, we would have to synchronize the nonce
1806  * count variable between the two sessions, if we'd do more requests
1807  * with the original session, though.)
1808  */
1810 
1811  /* complete the connection */
1812  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1813  err = AVERROR(EIO);
1814  goto fail;
1815  }
1816  } else {
1817  int ret;
1818  /* open the tcp connection */
1819  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1820  host, port,
1821  "?timeout=%d", rt->stimeout);
1822  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1824  err = ret;
1825  goto fail;
1826  }
1827  rt->rtsp_hd_out = rt->rtsp_hd;
1828  }
1829  rt->seq = 0;
1830 
1831  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1832  if (tcp_fd < 0) {
1833  err = tcp_fd;
1834  goto fail;
1835  }
1836  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1837  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1838  NULL, 0, NI_NUMERICHOST);
1839  }
1840 
1841  /* request options supported by the server; this also detects server
1842  * type */
1843  for (rt->server_type = RTSP_SERVER_RTP;;) {
1844  cmd[0] = 0;
1845  if (rt->server_type == RTSP_SERVER_REAL)
1846  av_strlcat(cmd,
1847  /*
1848  * The following entries are required for proper
1849  * streaming from a Realmedia server. They are
1850  * interdependent in some way although we currently
1851  * don't quite understand how. Values were copied
1852  * from mplayer SVN r23589.
1853  * ClientChallenge is a 16-byte ID in hex
1854  * CompanyID is a 16-byte ID in base64
1855  */
1856  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1857  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1858  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1859  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1860  sizeof(cmd));
1861  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1862  if (reply->status_code != RTSP_STATUS_OK) {
1864  goto fail;
1865  }
1866 
1867  /* detect server type if not standard-compliant RTP */
1868  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1870  continue;
1871  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1873  } else if (rt->server_type == RTSP_SERVER_REAL)
1874  strcpy(real_challenge, reply->real_challenge);
1875  break;
1876  }
1877 
1878  if (CONFIG_RTSP_DEMUXER && s->iformat)
1879  err = ff_rtsp_setup_input_streams(s, reply);
1880  else if (CONFIG_RTSP_MUXER)
1881  err = ff_rtsp_setup_output_streams(s, host);
1882  else
1883  av_assert0(0);
1884  if (err)
1885  goto fail;
1886 
1887  do {
1888  int lower_transport = ff_log2_tab[lower_transport_mask &
1889  ~(lower_transport_mask - 1)];
1890 
1891  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1892  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1893  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1894 
1895  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1896  rt->server_type == RTSP_SERVER_REAL ?
1897  real_challenge : NULL);
1898  if (err < 0)
1899  goto fail;
1900  lower_transport_mask &= ~(1 << lower_transport);
1901  if (lower_transport_mask == 0 && err == 1) {
1902  err = AVERROR(EPROTONOSUPPORT);
1903  goto fail;
1904  }
1905  } while (err);
1906 
1907  rt->lower_transport_mask = lower_transport_mask;
1908  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1909  rt->state = RTSP_STATE_IDLE;
1910  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1911  return 0;
1912  fail:
1915  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1916  char *new_url = av_strdup(reply->location);
1917  if (!new_url) {
1918  err = AVERROR(ENOMEM);
1919  goto fail2;
1920  }
1921  ff_format_set_url(s, new_url);
1922  rt->session_id[0] = '\0';
1923  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1924  reply->status_code,
1925  s->url);
1926  goto redirect;
1927  }
1928  fail2:
1929  ff_network_close();
1930  return err;
1931 }
1932 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1933 
1934 #if CONFIG_RTPDEC
1935 static int parse_rtsp_message(AVFormatContext *s)
1936 {
1937  RTSPState *rt = s->priv_data;
1938  int ret;
1939 
1940  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1941  if (rt->state == RTSP_STATE_STREAMING) {
1943  return AVERROR_EOF;
1944  else
1946  "Unable to answer to TEARDOWN\n");
1947  } else
1948  return 0;
1949  } else {
1950  RTSPMessageHeader reply;
1951  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1952  if (ret < 0)
1953  return ret;
1954  /* XXX: parse message */
1955  if (rt->state != RTSP_STATE_STREAMING)
1956  return 0;
1957  }
1958 
1959  return 0;
1960 }
1961 
1962 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1963  uint8_t *buf, int buf_size, int64_t wait_end)
1964 {
1965  RTSPState *rt = s->priv_data;
1966  RTSPStream *rtsp_st;
1967  int n, i, ret, timeout_cnt = 0;
1968  struct pollfd *p = rt->p;
1969  int *fds = NULL, fdsnum, fdsidx;
1970 
1971  if (!p) {
1972  p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
1973  if (!p)
1974  return AVERROR(ENOMEM);
1975 
1976  if (rt->rtsp_hd) {
1977  p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1978  p[rt->max_p++].events = POLLIN;
1979  }
1980  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1981  rtsp_st = rt->rtsp_streams[i];
1982  if (rtsp_st->rtp_handle) {
1983  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1984  &fds, &fdsnum)) {
1985  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1986  return ret;
1987  }
1988  if (fdsnum != 2) {
1989  av_log(s, AV_LOG_ERROR,
1990  "Number of fds %d not supported\n", fdsnum);
1991  return AVERROR_INVALIDDATA;
1992  }
1993  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1994  p[rt->max_p].fd = fds[fdsidx];
1995  p[rt->max_p++].events = POLLIN;
1996  }
1997  av_freep(&fds);
1998  }
1999  }
2000  }
2001 
2002  for (;;) {
2004  return AVERROR_EXIT;
2005  if (wait_end && wait_end - av_gettime_relative() < 0)
2006  return AVERROR(EAGAIN);
2007  n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2008  if (n > 0) {
2009  int j = rt->rtsp_hd ? 1 : 0;
2010  timeout_cnt = 0;
2011  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2012  rtsp_st = rt->rtsp_streams[i];
2013  if (rtsp_st->rtp_handle) {
2014  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2015  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2016  if (ret > 0) {
2017  *prtsp_st = rtsp_st;
2018  return ret;
2019  }
2020  }
2021  j+=2;
2022  }
2023  }
2024 #if CONFIG_RTSP_DEMUXER
2025  if (rt->rtsp_hd && p[0].revents & POLLIN) {
2026  if ((ret = parse_rtsp_message(s)) < 0) {
2027  return ret;
2028  }
2029  }
2030 #endif
2031  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2032  return AVERROR(ETIMEDOUT);
2033  } else if (n < 0 && errno != EINTR)
2034  return AVERROR(errno);
2035  }
2036 }
2037 
2038 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2039  const uint8_t *buf, int len)
2040 {
2041  RTSPState *rt = s->priv_data;
2042  int i;
2043  if (len < 0)
2044  return len;
2045  if (rt->nb_rtsp_streams == 1) {
2046  *rtsp_st = rt->rtsp_streams[0];
2047  return len;
2048  }
2049  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2050  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2051  int no_ssrc = 0;
2052  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2053  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2054  if (!rtpctx)
2055  continue;
2056  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2057  *rtsp_st = rt->rtsp_streams[i];
2058  return len;
2059  }
2060  if (!rtpctx->ssrc)
2061  no_ssrc = 1;
2062  }
2063  if (no_ssrc) {
2065  "Unable to pick stream for packet - SSRC not known for "
2066  "all streams\n");
2067  return AVERROR(EAGAIN);
2068  }
2069  } else {
2070  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2071  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2072  *rtsp_st = rt->rtsp_streams[i];
2073  return len;
2074  }
2075  }
2076  }
2077  }
2078  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2079  return AVERROR(EAGAIN);
2080 }
2081 
2082 static int read_packet(AVFormatContext *s,
2083  RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2084  int64_t wait_end)
2085 {
2086  RTSPState *rt = s->priv_data;
2087  int len;
2088 
2089  switch(rt->lower_transport) {
2090  default:
2091 #if CONFIG_RTSP_DEMUXER
2093  len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2094  break;
2095 #endif
2098  len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2099  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2100  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2101  break;
2103  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2104  wait_end && wait_end < av_gettime_relative())
2105  len = AVERROR(EAGAIN);
2106  else
2107  len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2108  len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2109  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2110  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2111  break;
2112  }
2113 
2114  if (len == 0)
2115  return AVERROR_EOF;
2116 
2117  return len;
2118 }
2119 
2121 {
2122  RTSPState *rt = s->priv_data;
2123  int ret, len;
2124  RTSPStream *rtsp_st, *first_queue_st = NULL;
2125  int64_t wait_end = 0;
2126 
2127  if (rt->nb_byes == rt->nb_rtsp_streams)
2128  return AVERROR_EOF;
2129 
2130  /* get next frames from the same RTP packet */
2131  if (rt->cur_transport_priv) {
2132  if (rt->transport == RTSP_TRANSPORT_RDT) {
2133  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2134  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2135  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2136  } else if (CONFIG_RTPDEC && rt->ts) {
2137  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2138  if (ret >= 0) {
2139  rt->recvbuf_pos += ret;
2140  ret = rt->recvbuf_pos < rt->recvbuf_len;
2141  }
2142  } else
2143  ret = -1;
2144  if (ret == 0) {
2145  rt->cur_transport_priv = NULL;
2146  return 0;
2147  } else if (ret == 1) {
2148  return 0;
2149  } else
2150  rt->cur_transport_priv = NULL;
2151  }
2152 
2153 redo:
2154  if (rt->transport == RTSP_TRANSPORT_RTP) {
2155  int i;
2156  int64_t first_queue_time = 0;
2157  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2158  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2159  int64_t queue_time;
2160  if (!rtpctx)
2161  continue;
2162  queue_time = ff_rtp_queued_packet_time(rtpctx);
2163  if (queue_time && (queue_time - first_queue_time < 0 ||
2164  !first_queue_time)) {
2165  first_queue_time = queue_time;
2166  first_queue_st = rt->rtsp_streams[i];
2167  }
2168  }
2169  if (first_queue_time) {
2170  wait_end = first_queue_time + s->max_delay;
2171  } else {
2172  wait_end = 0;
2173  first_queue_st = NULL;
2174  }
2175  }
2176 
2177  /* read next RTP packet */
2178  if (!rt->recvbuf) {
2180  if (!rt->recvbuf)
2181  return AVERROR(ENOMEM);
2182  }
2183 
2184  len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2185  if (len == AVERROR(EAGAIN) && first_queue_st &&
2186  rt->transport == RTSP_TRANSPORT_RTP) {
2188  "max delay reached. need to consume packet\n");
2189  rtsp_st = first_queue_st;
2190  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2191  goto end;
2192  }
2193  if (len < 0)
2194  return len;
2195 
2196  if (rt->transport == RTSP_TRANSPORT_RDT) {
2197  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2198  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2199  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2200  if (rtsp_st->feedback) {
2201  AVIOContext *pb = NULL;
2203  pb = s->pb;
2204  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2205  }
2206  if (ret < 0) {
2207  /* Either bad packet, or a RTCP packet. Check if the
2208  * first_rtcp_ntp_time field was initialized. */
2209  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2210  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2211  /* first_rtcp_ntp_time has been initialized for this stream,
2212  * copy the same value to all other uninitialized streams,
2213  * in order to map their timestamp origin to the same ntp time
2214  * as this one. */
2215  int i;
2216  AVStream *st = NULL;
2217  if (rtsp_st->stream_index >= 0)
2218  st = s->streams[rtsp_st->stream_index];
2219  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2220  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2221  AVStream *st2 = NULL;
2222  if (rt->rtsp_streams[i]->stream_index >= 0)
2223  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2224  if (rtpctx2 && st && st2 &&
2225  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2226  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2227  rtpctx2->rtcp_ts_offset = av_rescale_q(
2228  rtpctx->rtcp_ts_offset, st->time_base,
2229  st2->time_base);
2230  }
2231  }
2232  // Make real NTP start time available in AVFormatContext
2233  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2234  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2235  if (rtpctx->st) {
2236  s->start_time_realtime -=
2237  av_rescale (rtpctx->rtcp_ts_offset,
2238  (uint64_t) rtpctx->st->time_base.num * 1000000,
2239  rtpctx->st->time_base.den);
2240  }
2241  }
2242  }
2243  if (ret == -RTCP_BYE) {
2244  rt->nb_byes++;
2245 
2246  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2247  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2248 
2249  if (rt->nb_byes == rt->nb_rtsp_streams)
2250  return AVERROR_EOF;
2251  }
2252  }
2253  } else if (CONFIG_RTPDEC && rt->ts) {
2254  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2255  if (ret >= 0) {
2256  if (ret < len) {
2257  rt->recvbuf_len = len;
2258  rt->recvbuf_pos = ret;
2259  rt->cur_transport_priv = rt->ts;
2260  return 1;
2261  } else {
2262  ret = 0;
2263  }
2264  }
2265  } else {
2266  return AVERROR_INVALIDDATA;
2267  }
2268 end:
2269  if (ret < 0)
2270  goto redo;
2271  if (ret == 1)
2272  /* more packets may follow, so we save the RTP context */
2273  rt->cur_transport_priv = rtsp_st->transport_priv;
2274 
2275  return ret;
2276 }
2277 #endif /* CONFIG_RTPDEC */
2278 
2279 #if CONFIG_SDP_DEMUXER
2280 static int sdp_probe(AVProbeData *p1)
2281 {
2282  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2283 
2284  /* we look for a line beginning "c=IN IP" */
2285  while (p < p_end && *p != '\0') {
2286  if (sizeof("c=IN IP") - 1 < p_end - p &&
2287  av_strstart(p, "c=IN IP", NULL))
2288  return AVPROBE_SCORE_EXTENSION;
2289 
2290  while (p < p_end - 1 && *p != '\n') p++;
2291  if (++p >= p_end)
2292  break;
2293  if (*p == '\r')
2294  p++;
2295  }
2296  return 0;
2297 }
2298 
2299 static void append_source_addrs(char *buf, int size, const char *name,
2300  int count, struct RTSPSource **addrs)
2301 {
2302  int i;
2303  if (!count)
2304  return;
2305  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2306  for (i = 1; i < count; i++)
2307  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2308 }
2309 
2310 static int sdp_read_header(AVFormatContext *s)
2311 {
2312  RTSPState *rt = s->priv_data;
2313  RTSPStream *rtsp_st;
2314  int size, i, err;
2315  char *content;
2316  char url[1024];
2317 
2318  if (!ff_network_init())
2319  return AVERROR(EIO);
2320 
2321  if (s->max_delay < 0) /* Not set by the caller */
2323  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2325 
2326  /* read the whole sdp file */
2327  /* XXX: better loading */
2328  content = av_malloc(SDP_MAX_SIZE);
2329  if (!content)
2330  return AVERROR(ENOMEM);
2331  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2332  if (size <= 0) {
2333  av_free(content);
2334  return AVERROR_INVALIDDATA;
2335  }
2336  content[size] ='\0';
2337 
2338  err = ff_sdp_parse(s, content);
2339  av_freep(&content);
2340  if (err) goto fail;
2341 
2342  /* open each RTP stream */
2343  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2344  char namebuf[50];
2345  rtsp_st = rt->rtsp_streams[i];
2346 
2347  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2348  AVDictionary *opts = map_to_opts(rt);
2349 
2350  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2351  sizeof(rtsp_st->sdp_ip),
2352  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2353  if (err) {
2354  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2355  err = AVERROR(EIO);
2356  av_dict_free(&opts);
2357  goto fail;
2358  }
2359  ff_url_join(url, sizeof(url), "rtp", NULL,
2360  namebuf, rtsp_st->sdp_port,
2361  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2362  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2363  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2364  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2365 
2366  append_source_addrs(url, sizeof(url), "sources",
2367  rtsp_st->nb_include_source_addrs,
2368  rtsp_st->include_source_addrs);
2369  append_source_addrs(url, sizeof(url), "block",
2370  rtsp_st->nb_exclude_source_addrs,
2371  rtsp_st->exclude_source_addrs);
2372  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2374 
2375  av_dict_free(&opts);
2376 
2377  if (err < 0) {
2378  err = AVERROR_INVALIDDATA;
2379  goto fail;
2380  }
2381  }
2382  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2383  goto fail;
2384  }
2385  return 0;
2386 fail:
2388  ff_network_close();
2389  return err;
2390 }
2391 
2392 static int sdp_read_close(AVFormatContext *s)
2393 {
2395  ff_network_close();
2396  return 0;
2397 }
2398 
2399 static const AVClass sdp_demuxer_class = {
2400  .class_name = "SDP demuxer",
2401  .item_name = av_default_item_name,
2402  .option = sdp_options,
2403  .version = LIBAVUTIL_VERSION_INT,
2404 };
2405 
2407  .name = "sdp",
2408  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2409  .priv_data_size = sizeof(RTSPState),
2410  .read_probe = sdp_probe,
2411  .read_header = sdp_read_header,
2413  .read_close = sdp_read_close,
2414  .priv_class = &sdp_demuxer_class,
2415 };
2416 #endif /* CONFIG_SDP_DEMUXER */
2417 
2418 #if CONFIG_RTP_DEMUXER
2419 static int rtp_probe(AVProbeData *p)
2420 {
2421  if (av_strstart(p->filename, "rtp:", NULL))
2422  return AVPROBE_SCORE_MAX;
2423  return 0;
2424 }
2425 
2426 static int rtp_read_header(AVFormatContext *s)
2427 {
2428  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2429  char host[500], sdp[500];
2430  int ret, port;
2431  URLContext* in = NULL;
2432  int payload_type;
2433  AVCodecParameters *par = NULL;
2434  struct sockaddr_storage addr;
2435  AVIOContext pb;
2436  socklen_t addrlen = sizeof(addr);
2437  RTSPState *rt = s->priv_data;
2438 
2439  if (!ff_network_init())
2440  return AVERROR(EIO);
2441 
2442  ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2444  if (ret)
2445  goto fail;
2446 
2447  while (1) {
2448  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2449  if (ret == AVERROR(EAGAIN))
2450  continue;
2451  if (ret < 0)
2452  goto fail;
2453  if (ret < 12) {
2454  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2455  continue;
2456  }
2457 
2458  if ((recvbuf[0] & 0xc0) != 0x80) {
2459  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2460  "received\n");
2461  continue;
2462  }
2463 
2464  if (RTP_PT_IS_RTCP(recvbuf[1]))
2465  continue;
2466 
2467  payload_type = recvbuf[1] & 0x7f;
2468  break;
2469  }
2470  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2471  ffurl_close(in);
2472  in = NULL;
2473 
2474  par = avcodec_parameters_alloc();
2475  if (!par) {
2476  ret = AVERROR(ENOMEM);
2477  goto fail;
2478  }
2479 
2480  if (ff_rtp_get_codec_info(par, payload_type)) {
2481  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2482  "without an SDP file describing it\n",
2483  payload_type);
2484  goto fail;
2485  }
2486  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2487  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2488  "properly you need an SDP file "
2489  "describing it\n");
2490  }
2491 
2492  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2493  NULL, 0, s->url);
2494 
2495  snprintf(sdp, sizeof(sdp),
2496  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2497  addr.ss_family == AF_INET ? 4 : 6, host,
2498  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2499  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2500  port, payload_type);
2501  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2503 
2504  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2505  s->pb = &pb;
2506 
2507  /* sdp_read_header initializes this again */
2508  ff_network_close();
2509 
2510  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2511 
2512  ret = sdp_read_header(s);
2513  s->pb = NULL;
2514  return ret;
2515 
2516 fail:
2518  if (in)
2519  ffurl_close(in);
2520  ff_network_close();
2521  return ret;
2522 }
2523 
2524 static const AVClass rtp_demuxer_class = {
2525  .class_name = "RTP demuxer",
2526  .item_name = av_default_item_name,
2527  .option = rtp_options,
2528  .version = LIBAVUTIL_VERSION_INT,
2529 };
2530 
2532  .name = "rtp",
2533  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2534  .priv_data_size = sizeof(RTSPState),
2535  .read_probe = rtp_probe,
2536  .read_header = rtp_read_header,
2538  .read_close = sdp_read_close,
2539  .flags = AVFMT_NOFILE,
2540  .priv_class = &rtp_demuxer_class,
2541 };
2542 #endif /* CONFIG_RTP_DEMUXER */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:273
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:93
void av_url_split(char *proto, int proto_size, char *authorization, int authorization_size, char *hostname, int hostname_size, int *port_ptr, char *path, int path_size, const char *url)
Split a URL string into components.
Definition: utils.c:4707
char crypto_suite[40]
Definition: rtsp.h:476
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:388
#define NULL
Definition: coverity.c:32
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:100
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:581
Bytestream IO Context.
Definition: avio.h:161
Realmedia Data Transport.
Definition: rtsp.h:58
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:627
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
Definition: avformat.h:1602
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:307
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1627
AVOption.
Definition: opt.h:246
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:115
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:187
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
const char * filename
Definition: avformat.h:449
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
Definition: rtsp.c:173
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:317
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4827
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 ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:421
const char * desc
Definition: nvenc.c:65
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
Windows Media server.
Definition: rtsp.h:209
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:354
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:801
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:3101
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:166
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:131
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:3884
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
Definition: rdt.c:335
int num
Numerator.
Definition: rational.h:59
int index
stream index in AVFormatContext
Definition: avformat.h:875
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
#define AVIO_FLAG_READ
read-only
Definition: avio.h:654
char * user_agent
User-Agent string.
Definition: rtsp.h:408
char location[4096]
the "Location:" field.
Definition: rtsp.h:152
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:655
int mode_record
transport set to record data
Definition: rtsp.h:112
enum AVMediaType codec_type
Definition: rtp.c:37
int avio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:717
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:223
void ff_network_close(void)
Definition: network.c:115
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:144
initialized and sending/receiving data
Definition: rtsp.h:197
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:270
const char * key
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:421
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:79
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
#define freeaddrinfo
Definition: network.h:215
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:453
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1400
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:419
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:245
int auth_type
The currently chosen auth type.
Definition: httpauth.h:59
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:239
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:132
#define AI_NUMERICHOST
Definition: network.h:184
This struct describes the properties of an encoded stream.
Definition: avcodec.h:3876
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:121
This describes the server response to each RTSP command.
Definition: rtsp.h:127
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
Definition: rtpdec.c:538
#define RECVBUF_SIZE
Definition: rtsp.c:59
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:142
Format I/O context.
Definition: avformat.h:1351
#define RTP_PT_PRIVATE
Definition: rtp.h:77
#define COMMON_OPTS()
Definition: rtsp.c:77
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
Definition: rtp.c:143
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:207
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:403
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
#define RTSP_FLAG_PREFER_TCP
Try RTP via TCP first if possible.
Definition: rtsp.h:424
int recvbuf_len
Definition: rtsp.h:323
static void interleave(uint8_t *dst, uint8_t *src, int w, int h, int dst_linesize, int src_linesize, enum FilterMode mode, int swap)
Definition: vf_il.c:117
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:180
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Public dictionary API.
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:45
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:360
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:148
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:74
#define av_malloc(s)
Opaque data information usually continuous.
Definition: avutil.h:203
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:109
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
Definition: rtpdec.h:126
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:195
int ff_network_init(void)
Definition: network.c:57
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1295
AVOptions.
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: utils.c:1982
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:471
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
uint16_t ss_family
Definition: network.h:113
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int id
Format-specific stream ID.
Definition: avformat.h:881
enum AVStreamParseType need_parsing
Definition: avformat.h:1092
#define POLL_TIMEOUT_MS
Definition: rtsp.c:55
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:60
static void handler(vbi_event *ev, void *user_data)
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4456
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1419
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:373
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:328
int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:71
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:436
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
char * protocol_whitelist
',' separated list of allowed protocols.
Definition: avformat.h:1909
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:131
#define AVERROR_EOF
End of file.
Definition: error.h:55
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:179
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
ptrdiff_t size
Definition: opengl_enc.c:101
static const uint8_t header[24]
Definition: sdr2.c:67
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:465
#define flags(name, subs,...)
Definition: cbs_h2645.c:263
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
Normal RTSP.
Definition: rtsp.h:68
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
#define av_log(a,...)
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:134
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:648
AVInputFormat ff_rtp_demuxer
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1370
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:177
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:77
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:456
Private data for the RTSP demuxer.
Definition: rtsp.h:218
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:255
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
Definition: avio.c:290
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVDictionary * metadata
Metadata that applies to the whole file.
Definition: avformat.h:1589
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:633
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
const char * protocol_whitelist
Definition: url.h:49
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
char * url
input or output URL.
Definition: avformat.h:1447
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const AVOption ff_rtsp_options[]
Definition: rtsp.c:82
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:203
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:3880
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:182
Definition: graph2dot.c:48
URLContext * rtsp_hd
Definition: rtsp.h:220
simple assert() macros that are a bit more flexible than ISO C assert().
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:331
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:454
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, const RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:574
GLsizei count
Definition: opengl_enc.c:109
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
Definition: base64.c:138
void avcodec_parameters_free(AVCodecParameters **par)
Free an AVCodecParameters instance and everything associated with it and write NULL to the supplied p...
Definition: utils.c:1992
int64_t rtcp_ts_offset
Definition: rtpdec.h:182
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:83
#define fail()
Definition: checkasm.h:117
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:225
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:164
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
Definition: codec_desc.c:3135
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:441
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:451
int seq
RTSP command sequence number.
Definition: rtsp.h:241
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:450
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:339
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1407
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:420
AVDictionary * opts
Definition: movenc.c:50
#define NI_NUMERICHOST
Definition: network.h:192
#define th
Definition: regdef.h:75
#define LIBAVFORMAT_IDENT
Definition: version.h:46
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:307
int recvbuf_pos
Definition: rtsp.h:322
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:198
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:223
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes to a null-terminated string.
Definition: base64.h:66
#define FFMIN(a, b)
Definition: common.h:96
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:283
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:55
int content_length
length of the data following this header
Definition: rtsp.h:129
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:172
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:75
enum AVStreamParseType need_parsing
Definition: rtpdec.h:119
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:88
RTSP over HTTP (tunneling)
Definition: rtsp.h:69
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:138
#define s(width, name)
Definition: cbs_vp9.c:257
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
void ff_format_set_url(AVFormatContext *s, char *url)
Set AVFormatContext url field to the provided pointer.
Definition: utils.c:5775
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:164
int n
Definition: avisynth_c.h:684
AVDictionary * metadata
Definition: avformat.h:938
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:464
char crypto_params[100]
Definition: rtsp.h:477
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:200
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
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
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:626
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define ENC
Definition: rtsp.c:64
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:451
Raw data (over UDP)
Definition: rtsp.h:59
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:321
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:23
int sdp_payload_type
payload type
Definition: rtsp.h:458
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:455
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1450
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:530
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:470
Stream structure.
Definition: avformat.h:874
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
Definition: url.c:36
int nb_byes
Definition: rtsp.h:336
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:262
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:427
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:452
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:251
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:731
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:664
int rtp_port_max
Definition: rtsp.h:388
#define NTP_OFFSET
Definition: internal.h:236
Definition: rtp.h:100
AVIOContext * pb
I/O context.
Definition: avformat.h:1393
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:383
AVInputFormat ff_sdp_demuxer
int server_port_max
Definition: rtsp.h:105
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:66
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:474
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:414
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:258
void * buf
Definition: avisynth_c.h:690
Definition: url.h:38
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:73
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:656
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 rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:378
int client_port_max
Definition: rtsp.h:101
Describe the class of an AVClass context structure.
Definition: log.h:67
#define SDP_MAX_SIZE
Definition: rtsp.c:58
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:515
#define SPACE_CHARS
Definition: internal.h:338
void * priv_data
Definition: url.h:41
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:467
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:279
#define gai_strerror
Definition: network.h:222
not initialized
Definition: rtsp.h:196
int64_t range_end
Definition: rtsp.h:138
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:118
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:3120
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:155
AVMediaType
Definition: avutil.h:199
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:70
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:755
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
Definition: rtspdec.c:750
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:765
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:299
#define s1
Definition: regdef.h:38
const char * name
Name of the codec described by this descriptor.
Definition: avcodec.h:707
#define snprintf
Definition: snprintf.h:34
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:458
int max_p
Definition: rtsp.h:355
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4390
int buffer_size
Definition: rtsp.h:411
This structure contains the data a format has to probe a file.
Definition: avformat.h:448
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:76
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
Definition: avstring.c:93
int interleaved_max
Definition: rtsp.h:93
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
mfxU16 profile
Definition: qsvenc.c:44
This struct describes the properties of a single codec described by an AVCodecID. ...
Definition: avcodec.h:699
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:267
int ffurl_close(URLContext *h)
Definition: avio.c:467
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:138
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1456
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:231
int sample_rate
Audio only.
Definition: avcodec.h:3994
#define DEC
Definition: rtsp.c:63
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:460
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:34
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:214
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:111
const OptionDef options[]
Definition: ffmpeg_opt.c:3294
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
uint32_t ssrc
Definition: rtpdec.h:153
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:127
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:465
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:81
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:288
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:414
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
Definition: rdt.c:94
int den
Denominator.
Definition: rational.h:60
char default_lang[4]
Definition: rtsp.h:410
struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:1363
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4428
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
uint32_t base_timestamp
Definition: rtpdec.h:156
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:398
#define getnameinfo
Definition: network.h:216
#define av_free(p)
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:157
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:276
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:78
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:447
void * priv_data
Format private data.
Definition: avformat.h:1379
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:593
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:882
int channels
Audio only.
Definition: avcodec.h:3990
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:457
#define MAX_TIMEOUTS
Definition: rtsp.c:57
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1247
char * protocol_blacklist
',' separated list of disallowed protocols.
Definition: avformat.h:1944
int ai_flags
Definition: network.h:135
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1466
Realmedia-style server.
Definition: rtsp.h:208
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:344
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:690
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:647
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:120
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1021
#define av_malloc_array(a, b)
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:903
int interleaved_max
Definition: rtsp.h:445
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:869
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:114
int ff_rtp_set_remote_url(URLContext *h, const char *uri)
If no filename is given to av_open_input_file because you want to get the local port first...
Definition: rtpproto.c:100
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:3145
AVStream * st
Definition: rtpdec.h:151
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:160
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:445
This structure stores compressed data.
Definition: avcodec.h:1407
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:1215
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:105
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:449
static const AVOption rtp_options[]
Definition: rtsp.c:120
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:407
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:437
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
#define OFFSET(x)
Definition: rtsp.c:62
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:146
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:97
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:438
No authentication specified.
Definition: httpauth.h:29
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:101
const char * name
Definition: opengl_enc.c:103