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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  { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97  { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
98  COMMON_OPTS(),
99  { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
100  { NULL },
101 };
102 
103 static const AVOption sdp_options[] = {
104  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106  { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
107  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
108  COMMON_OPTS(),
109  { NULL },
110 };
111 
112 static const AVOption rtp_options[] = {
113  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
114  COMMON_OPTS(),
115  { NULL },
116 };
117 
118 
120 {
122  char buf[256];
123 
124  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125  av_dict_set(&opts, "buffer_size", buf, 0);
126 
127  return opts;
128 }
129 
130 static void get_word_until_chars(char *buf, int buf_size,
131  const char *sep, const char **pp)
132 {
133  const char *p;
134  char *q;
135 
136  p = *pp;
137  p += strspn(p, SPACE_CHARS);
138  q = buf;
139  while (!strchr(sep, *p) && *p != '\0') {
140  if ((q - buf) < buf_size - 1)
141  *q++ = *p;
142  p++;
143  }
144  if (buf_size > 0)
145  *q = '\0';
146  *pp = p;
147 }
148 
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
150  const char **pp)
151 {
152  if (**pp == '/') (*pp)++;
153  get_word_until_chars(buf, buf_size, sep, pp);
154 }
155 
156 static void get_word(char *buf, int buf_size, const char **pp)
157 {
158  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
159 }
160 
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
162  * and end time.
163  * Used for seeking in the rtp stream.
164  */
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
166 {
167  char buf[256];
168 
169  p += strspn(p, SPACE_CHARS);
170  if (!av_stristart(p, "npt=", &p))
171  return;
172 
173  *start = AV_NOPTS_VALUE;
174  *end = AV_NOPTS_VALUE;
175 
176  get_word_sep(buf, sizeof(buf), "-", &p);
177  if (av_parse_time(start, buf, 1) < 0)
178  return;
179  if (*p == '-') {
180  p++;
181  get_word_sep(buf, sizeof(buf), "-", &p);
182  if (av_parse_time(end, buf, 1) < 0)
183  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
184  }
185 }
186 
188  const char *buf, struct sockaddr_storage *sock)
189 {
190  struct addrinfo hints = { 0 }, *ai = NULL;
191  int ret;
192 
193  hints.ai_flags = AI_NUMERICHOST;
194  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
195  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
196  buf,
197  gai_strerror(ret));
198  return -1;
199  }
200  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
201  freeaddrinfo(ai);
202  return 0;
203 }
204 
205 #if CONFIG_RTPDEC
206 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
207  RTSPStream *rtsp_st, AVStream *st)
208 {
209  AVCodecParameters *par = st ? st->codecpar : NULL;
210  if (!handler)
211  return;
212  if (par)
213  par->codec_id = handler->codec_id;
214  rtsp_st->dynamic_handler = handler;
215  if (st)
216  st->need_parsing = handler->need_parsing;
217  if (handler->priv_data_size) {
219  if (!rtsp_st->dynamic_protocol_context)
220  rtsp_st->dynamic_handler = NULL;
221  }
222 }
223 
224 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
225  AVStream *st)
226 {
227  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
228  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
229  rtsp_st->dynamic_protocol_context);
230  if (ret < 0) {
231  if (rtsp_st->dynamic_protocol_context) {
232  if (rtsp_st->dynamic_handler->close)
233  rtsp_st->dynamic_handler->close(
234  rtsp_st->dynamic_protocol_context);
236  }
237  rtsp_st->dynamic_protocol_context = NULL;
238  rtsp_st->dynamic_handler = NULL;
239  }
240  }
241 }
242 
243 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
244 static int sdp_parse_rtpmap(AVFormatContext *s,
245  AVStream *st, RTSPStream *rtsp_st,
246  int payload_type, const char *p)
247 {
248  AVCodecParameters *par = st->codecpar;
249  char buf[256];
250  int i;
251  const AVCodecDescriptor *desc;
252  const char *c_name;
253 
254  /* See if we can handle this kind of payload.
255  * The space should normally not be there but some Real streams or
256  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
257  * have a trailing space. */
258  get_word_sep(buf, sizeof(buf), "/ ", &p);
259  if (payload_type < RTP_PT_PRIVATE) {
260  /* We are in a standard case
261  * (from http://www.iana.org/assignments/rtp-parameters). */
262  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
263  }
264 
265  if (par->codec_id == AV_CODEC_ID_NONE) {
266  RTPDynamicProtocolHandler *handler =
268  init_rtp_handler(handler, rtsp_st, st);
269  /* If no dynamic handler was found, check with the list of standard
270  * allocated types, if such a stream for some reason happens to
271  * use a private payload type. This isn't handled in rtpdec.c, since
272  * the format name from the rtpmap line never is passed into rtpdec. */
273  if (!rtsp_st->dynamic_handler)
274  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
275  }
276 
277  desc = avcodec_descriptor_get(par->codec_id);
278  if (desc && desc->name)
279  c_name = desc->name;
280  else
281  c_name = "(null)";
282 
283  get_word_sep(buf, sizeof(buf), "/", &p);
284  i = atoi(buf);
285  switch (par->codec_type) {
286  case AVMEDIA_TYPE_AUDIO:
287  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
290  if (i > 0) {
291  par->sample_rate = i;
292  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
293  get_word_sep(buf, sizeof(buf), "/", &p);
294  i = atoi(buf);
295  if (i > 0)
296  par->channels = i;
297  }
298  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
299  par->sample_rate);
300  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
301  par->channels);
302  break;
303  case AVMEDIA_TYPE_VIDEO:
304  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
305  if (i > 0)
306  avpriv_set_pts_info(st, 32, 1, i);
307  break;
308  default:
309  break;
310  }
311  finalize_rtp_handler_init(s, rtsp_st, st);
312  return 0;
313 }
314 
315 /* parse the attribute line from the fmtp a line of an sdp response. This
316  * is broken out as a function because it is used in rtp_h264.c, which is
317  * forthcoming. */
318 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
319  char *value, int value_size)
320 {
321  *p += strspn(*p, SPACE_CHARS);
322  if (**p) {
323  get_word_sep(attr, attr_size, "=", p);
324  if (**p == '=')
325  (*p)++;
326  get_word_sep(value, value_size, ";", p);
327  if (**p == ';')
328  (*p)++;
329  return 1;
330  }
331  return 0;
332 }
333 
334 typedef struct SDPParseState {
335  /* SDP only */
336  struct sockaddr_storage default_ip;
337  int default_ttl;
338  int skip_media; ///< set if an unknown m= line occurs
339  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
340  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
341  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
342  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
343  int seen_rtpmap;
344  int seen_fmtp;
345  char delayed_fmtp[2048];
346 } SDPParseState;
347 
348 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
349  struct RTSPSource ***dest, int *dest_count)
350 {
351  RTSPSource *rtsp_src, *rtsp_src2;
352  int i;
353  for (i = 0; i < count; i++) {
354  rtsp_src = addrs[i];
355  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
356  if (!rtsp_src2)
357  continue;
358  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
359  dynarray_add(dest, dest_count, rtsp_src2);
360  }
361 }
362 
363 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
364  int payload_type, const char *line)
365 {
366  int i;
367 
368  for (i = 0; i < rt->nb_rtsp_streams; i++) {
369  RTSPStream *rtsp_st = rt->rtsp_streams[i];
370  if (rtsp_st->sdp_payload_type == payload_type &&
371  rtsp_st->dynamic_handler &&
372  rtsp_st->dynamic_handler->parse_sdp_a_line) {
373  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
374  rtsp_st->dynamic_protocol_context, line);
375  }
376  }
377 }
378 
379 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
380  int letter, const char *buf)
381 {
382  RTSPState *rt = s->priv_data;
383  char buf1[64], st_type[64];
384  const char *p;
385  enum AVMediaType codec_type;
386  int payload_type;
387  AVStream *st;
388  RTSPStream *rtsp_st;
389  RTSPSource *rtsp_src;
390  struct sockaddr_storage sdp_ip;
391  int ttl;
392 
393  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
394 
395  p = buf;
396  if (s1->skip_media && letter != 'm')
397  return;
398  switch (letter) {
399  case 'c':
400  get_word(buf1, sizeof(buf1), &p);
401  if (strcmp(buf1, "IN") != 0)
402  return;
403  get_word(buf1, sizeof(buf1), &p);
404  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
405  return;
406  get_word_sep(buf1, sizeof(buf1), "/", &p);
407  if (get_sockaddr(s, buf1, &sdp_ip))
408  return;
409  ttl = 16;
410  if (*p == '/') {
411  p++;
412  get_word_sep(buf1, sizeof(buf1), "/", &p);
413  ttl = atoi(buf1);
414  }
415  if (s->nb_streams == 0) {
416  s1->default_ip = sdp_ip;
417  s1->default_ttl = ttl;
418  } else {
419  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
420  rtsp_st->sdp_ip = sdp_ip;
421  rtsp_st->sdp_ttl = ttl;
422  }
423  break;
424  case 's':
425  av_dict_set(&s->metadata, "title", p, 0);
426  break;
427  case 'i':
428  if (s->nb_streams == 0) {
429  av_dict_set(&s->metadata, "comment", p, 0);
430  break;
431  }
432  break;
433  case 'm':
434  /* new stream */
435  s1->skip_media = 0;
436  s1->seen_fmtp = 0;
437  s1->seen_rtpmap = 0;
438  codec_type = AVMEDIA_TYPE_UNKNOWN;
439  get_word(st_type, sizeof(st_type), &p);
440  if (!strcmp(st_type, "audio")) {
441  codec_type = AVMEDIA_TYPE_AUDIO;
442  } else if (!strcmp(st_type, "video")) {
443  codec_type = AVMEDIA_TYPE_VIDEO;
444  } else if (!strcmp(st_type, "application")) {
445  codec_type = AVMEDIA_TYPE_DATA;
446  } else if (!strcmp(st_type, "text")) {
447  codec_type = AVMEDIA_TYPE_SUBTITLE;
448  }
449  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
450  s1->skip_media = 1;
451  return;
452  }
453  rtsp_st = av_mallocz(sizeof(RTSPStream));
454  if (!rtsp_st)
455  return;
456  rtsp_st->stream_index = -1;
457  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
458 
459  rtsp_st->sdp_ip = s1->default_ip;
460  rtsp_st->sdp_ttl = s1->default_ttl;
461 
462  copy_default_source_addrs(s1->default_include_source_addrs,
463  s1->nb_default_include_source_addrs,
464  &rtsp_st->include_source_addrs,
465  &rtsp_st->nb_include_source_addrs);
466  copy_default_source_addrs(s1->default_exclude_source_addrs,
467  s1->nb_default_exclude_source_addrs,
468  &rtsp_st->exclude_source_addrs,
469  &rtsp_st->nb_exclude_source_addrs);
470 
471  get_word(buf1, sizeof(buf1), &p); /* port */
472  rtsp_st->sdp_port = atoi(buf1);
473 
474  get_word(buf1, sizeof(buf1), &p); /* protocol */
475  if (!strcmp(buf1, "udp"))
477  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
478  rtsp_st->feedback = 1;
479 
480  /* XXX: handle list of formats */
481  get_word(buf1, sizeof(buf1), &p); /* format list */
482  rtsp_st->sdp_payload_type = atoi(buf1);
483 
484  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
485  /* no corresponding stream */
486  if (rt->transport == RTSP_TRANSPORT_RAW) {
487  if (CONFIG_RTPDEC && !rt->ts)
488  rt->ts = avpriv_mpegts_parse_open(s);
489  } else {
491  handler = ff_rtp_handler_find_by_id(
493  init_rtp_handler(handler, rtsp_st, NULL);
494  finalize_rtp_handler_init(s, rtsp_st, NULL);
495  }
496  } else if (rt->server_type == RTSP_SERVER_WMS &&
497  codec_type == AVMEDIA_TYPE_DATA) {
498  /* RTX stream, a stream that carries all the other actual
499  * audio/video streams. Don't expose this to the callers. */
500  } else {
501  st = avformat_new_stream(s, NULL);
502  if (!st)
503  return;
504  st->id = rt->nb_rtsp_streams - 1;
505  rtsp_st->stream_index = st->index;
507  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
509  /* if standard payload type, we can find the codec right now */
511  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
512  st->codecpar->sample_rate > 0)
513  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
514  /* Even static payload types may need a custom depacketizer */
515  handler = ff_rtp_handler_find_by_id(
516  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
517  init_rtp_handler(handler, rtsp_st, st);
518  finalize_rtp_handler_init(s, rtsp_st, st);
519  }
520  if (rt->default_lang[0])
521  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
522  }
523  /* put a default control url */
524  av_strlcpy(rtsp_st->control_url, rt->control_uri,
525  sizeof(rtsp_st->control_url));
526  break;
527  case 'a':
528  if (av_strstart(p, "control:", &p)) {
529  if (s->nb_streams == 0) {
530  if (!strncmp(p, "rtsp://", 7))
531  av_strlcpy(rt->control_uri, p,
532  sizeof(rt->control_uri));
533  } else {
534  char proto[32];
535  /* get the control url */
536  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
537 
538  /* XXX: may need to add full url resolution */
539  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
540  NULL, NULL, 0, p);
541  if (proto[0] == '\0') {
542  /* relative control URL */
543  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
544  av_strlcat(rtsp_st->control_url, "/",
545  sizeof(rtsp_st->control_url));
546  av_strlcat(rtsp_st->control_url, p,
547  sizeof(rtsp_st->control_url));
548  } else
549  av_strlcpy(rtsp_st->control_url, p,
550  sizeof(rtsp_st->control_url));
551  }
552  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
553  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
554  get_word(buf1, sizeof(buf1), &p);
555  payload_type = atoi(buf1);
556  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
557  if (rtsp_st->stream_index >= 0) {
558  st = s->streams[rtsp_st->stream_index];
559  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
560  }
561  s1->seen_rtpmap = 1;
562  if (s1->seen_fmtp) {
563  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
564  }
565  } else if (av_strstart(p, "fmtp:", &p) ||
566  av_strstart(p, "framesize:", &p)) {
567  // let dynamic protocol handlers have a stab at the line.
568  get_word(buf1, sizeof(buf1), &p);
569  payload_type = atoi(buf1);
570  if (s1->seen_rtpmap) {
571  parse_fmtp(s, rt, payload_type, buf);
572  } else {
573  s1->seen_fmtp = 1;
574  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
575  }
576  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
577  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
578  get_word(buf1, sizeof(buf1), &p);
579  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
580  } else if (av_strstart(p, "range:", &p)) {
581  int64_t start, end;
582 
583  // this is so that seeking on a streamed file can work.
584  rtsp_parse_range_npt(p, &start, &end);
585  s->start_time = start;
586  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
587  s->duration = (end == AV_NOPTS_VALUE) ?
588  AV_NOPTS_VALUE : end - start;
589  } else if (av_strstart(p, "lang:", &p)) {
590  if (s->nb_streams > 0) {
591  get_word(buf1, sizeof(buf1), &p);
592  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593  if (rtsp_st->stream_index >= 0) {
594  st = s->streams[rtsp_st->stream_index];
595  av_dict_set(&st->metadata, "language", buf1, 0);
596  }
597  } else
598  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
599  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
600  if (atoi(p) == 1)
602  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
603  s->nb_streams > 0) {
604  st = s->streams[s->nb_streams - 1];
605  st->codecpar->sample_rate = atoi(p);
606  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
607  // RFC 4568
608  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
609  get_word(buf1, sizeof(buf1), &p); // ignore tag
610  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
611  p += strspn(p, SPACE_CHARS);
612  if (av_strstart(p, "inline:", &p))
613  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
614  } else if (av_strstart(p, "source-filter:", &p)) {
615  int exclude = 0;
616  get_word(buf1, sizeof(buf1), &p);
617  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
618  return;
619  exclude = !strcmp(buf1, "excl");
620 
621  get_word(buf1, sizeof(buf1), &p);
622  if (strcmp(buf1, "IN") != 0)
623  return;
624  get_word(buf1, sizeof(buf1), &p);
625  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
626  return;
627  // not checking that the destination address actually matches or is wildcard
628  get_word(buf1, sizeof(buf1), &p);
629 
630  while (*p != '\0') {
631  rtsp_src = av_mallocz(sizeof(*rtsp_src));
632  if (!rtsp_src)
633  return;
634  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
635  if (exclude) {
636  if (s->nb_streams == 0) {
637  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
638  } else {
639  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
640  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
641  }
642  } else {
643  if (s->nb_streams == 0) {
644  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
645  } else {
646  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
647  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
648  }
649  }
650  }
651  } else {
652  if (rt->server_type == RTSP_SERVER_WMS)
654  if (s->nb_streams > 0) {
655  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
656 
657  if (rt->server_type == RTSP_SERVER_REAL)
658  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
659 
660  if (rtsp_st->dynamic_handler &&
662  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
663  rtsp_st->stream_index,
664  rtsp_st->dynamic_protocol_context, buf);
665  }
666  }
667  break;
668  }
669 }
670 
671 int ff_sdp_parse(AVFormatContext *s, const char *content)
672 {
673  const char *p;
674  int letter, i;
675  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
676  * contain long SDP lines containing complete ASF Headers (several
677  * kB) or arrays of MDPR (RM stream descriptor) headers plus
678  * "rulebooks" describing their properties. Therefore, the SDP line
679  * buffer is large.
680  *
681  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
682  * in rtpdec_xiph.c. */
683  char buf[16384], *q;
684  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
685 
686  p = content;
687  for (;;) {
688  p += strspn(p, SPACE_CHARS);
689  letter = *p;
690  if (letter == '\0')
691  break;
692  p++;
693  if (*p != '=')
694  goto next_line;
695  p++;
696  /* get the content */
697  q = buf;
698  while (*p != '\n' && *p != '\r' && *p != '\0') {
699  if ((q - buf) < sizeof(buf) - 1)
700  *q++ = *p;
701  p++;
702  }
703  *q = '\0';
704  sdp_parse_line(s, s1, letter, buf);
705  next_line:
706  while (*p != '\n' && *p != '\0')
707  p++;
708  if (*p == '\n')
709  p++;
710  }
711 
712  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
713  av_freep(&s1->default_include_source_addrs[i]);
714  av_freep(&s1->default_include_source_addrs);
715  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
716  av_freep(&s1->default_exclude_source_addrs[i]);
717  av_freep(&s1->default_exclude_source_addrs);
718 
719  return 0;
720 }
721 #endif /* CONFIG_RTPDEC */
722 
723 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
724 {
725  RTSPState *rt = s->priv_data;
726  int i;
727 
728  for (i = 0; i < rt->nb_rtsp_streams; i++) {
729  RTSPStream *rtsp_st = rt->rtsp_streams[i];
730  if (!rtsp_st)
731  continue;
732  if (rtsp_st->transport_priv) {
733  if (s->oformat) {
734  AVFormatContext *rtpctx = rtsp_st->transport_priv;
735  av_write_trailer(rtpctx);
737  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
738  ff_rtsp_tcp_write_packet(s, rtsp_st);
739  ffio_free_dyn_buf(&rtpctx->pb);
740  } else {
741  avio_closep(&rtpctx->pb);
742  }
743  avformat_free_context(rtpctx);
744  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
746  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
748  }
749  rtsp_st->transport_priv = NULL;
750  if (rtsp_st->rtp_handle)
751  ffurl_close(rtsp_st->rtp_handle);
752  rtsp_st->rtp_handle = NULL;
753  }
754 }
755 
756 /* close and free RTSP streams */
758 {
759  RTSPState *rt = s->priv_data;
760  int i, j;
761  RTSPStream *rtsp_st;
762 
763  ff_rtsp_undo_setup(s, 0);
764  for (i = 0; i < rt->nb_rtsp_streams; i++) {
765  rtsp_st = rt->rtsp_streams[i];
766  if (rtsp_st) {
767  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
768  if (rtsp_st->dynamic_handler->close)
769  rtsp_st->dynamic_handler->close(
770  rtsp_st->dynamic_protocol_context);
772  }
773  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
774  av_freep(&rtsp_st->include_source_addrs[j]);
775  av_freep(&rtsp_st->include_source_addrs);
776  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
777  av_freep(&rtsp_st->exclude_source_addrs[j]);
778  av_freep(&rtsp_st->exclude_source_addrs);
779 
780  av_freep(&rtsp_st);
781  }
782  }
783  av_freep(&rt->rtsp_streams);
784  if (rt->asf_ctx) {
786  }
787  if (CONFIG_RTPDEC && rt->ts)
789  av_freep(&rt->p);
790  av_freep(&rt->recvbuf);
791 }
792 
794 {
795  RTSPState *rt = s->priv_data;
796  AVStream *st = NULL;
797  int reordering_queue_size = rt->reordering_queue_size;
798  if (reordering_queue_size < 0) {
800  reordering_queue_size = 0;
801  else
802  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
803  }
804 
805  /* open the RTP context */
806  if (rtsp_st->stream_index >= 0)
807  st = s->streams[rtsp_st->stream_index];
808  if (!st)
810 
811  if (CONFIG_RTSP_MUXER && s->oformat && st) {
812  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
813  s, st, rtsp_st->rtp_handle,
815  rtsp_st->stream_index);
816  /* Ownership of rtp_handle is passed to the rtp mux context */
817  rtsp_st->rtp_handle = NULL;
818  if (ret < 0)
819  return ret;
820  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
821  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
822  return 0; // Don't need to open any parser here
823  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
824  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
825  rtsp_st->dynamic_protocol_context,
826  rtsp_st->dynamic_handler);
827  else if (CONFIG_RTPDEC)
828  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
829  rtsp_st->sdp_payload_type,
830  reordering_queue_size);
831 
832  if (!rtsp_st->transport_priv) {
833  return AVERROR(ENOMEM);
834  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
835  s->iformat) {
836  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
837  rtpctx->ssrc = rtsp_st->ssrc;
838  if (rtsp_st->dynamic_handler) {
840  rtsp_st->dynamic_protocol_context,
841  rtsp_st->dynamic_handler);
842  }
843  if (rtsp_st->crypto_suite[0])
845  rtsp_st->crypto_suite,
846  rtsp_st->crypto_params);
847  }
848 
849  return 0;
850 }
851 
852 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
853 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
854 {
855  const char *q;
856  char *p;
857  int v;
858 
859  q = *pp;
860  q += strspn(q, SPACE_CHARS);
861  v = strtol(q, &p, 10);
862  if (*p == '-') {
863  p++;
864  *min_ptr = v;
865  v = strtol(p, &p, 10);
866  *max_ptr = v;
867  } else {
868  *min_ptr = v;
869  *max_ptr = v;
870  }
871  *pp = p;
872 }
873 
874 /* XXX: only one transport specification is parsed */
875 static void rtsp_parse_transport(AVFormatContext *s,
876  RTSPMessageHeader *reply, const char *p)
877 {
878  char transport_protocol[16];
879  char profile[16];
880  char lower_transport[16];
881  char parameter[16];
883  char buf[256];
884 
885  reply->nb_transports = 0;
886 
887  for (;;) {
888  p += strspn(p, SPACE_CHARS);
889  if (*p == '\0')
890  break;
891 
892  th = &reply->transports[reply->nb_transports];
893 
894  get_word_sep(transport_protocol, sizeof(transport_protocol),
895  "/", &p);
896  if (!av_strcasecmp (transport_protocol, "rtp")) {
897  get_word_sep(profile, sizeof(profile), "/;,", &p);
898  lower_transport[0] = '\0';
899  /* rtp/avp/<protocol> */
900  if (*p == '/') {
901  get_word_sep(lower_transport, sizeof(lower_transport),
902  ";,", &p);
903  }
905  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
906  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
907  /* x-pn-tng/<protocol> */
908  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
909  profile[0] = '\0';
911  } else if (!av_strcasecmp(transport_protocol, "raw")) {
912  get_word_sep(profile, sizeof(profile), "/;,", &p);
913  lower_transport[0] = '\0';
914  /* raw/raw/<protocol> */
915  if (*p == '/') {
916  get_word_sep(lower_transport, sizeof(lower_transport),
917  ";,", &p);
918  }
920  }
921  if (!av_strcasecmp(lower_transport, "TCP"))
923  else
925 
926  if (*p == ';')
927  p++;
928  /* get each parameter */
929  while (*p != '\0' && *p != ',') {
930  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
931  if (!strcmp(parameter, "port")) {
932  if (*p == '=') {
933  p++;
934  rtsp_parse_range(&th->port_min, &th->port_max, &p);
935  }
936  } else if (!strcmp(parameter, "client_port")) {
937  if (*p == '=') {
938  p++;
939  rtsp_parse_range(&th->client_port_min,
940  &th->client_port_max, &p);
941  }
942  } else if (!strcmp(parameter, "server_port")) {
943  if (*p == '=') {
944  p++;
945  rtsp_parse_range(&th->server_port_min,
946  &th->server_port_max, &p);
947  }
948  } else if (!strcmp(parameter, "interleaved")) {
949  if (*p == '=') {
950  p++;
951  rtsp_parse_range(&th->interleaved_min,
952  &th->interleaved_max, &p);
953  }
954  } else if (!strcmp(parameter, "multicast")) {
957  } else if (!strcmp(parameter, "ttl")) {
958  if (*p == '=') {
959  char *end;
960  p++;
961  th->ttl = strtol(p, &end, 10);
962  p = end;
963  }
964  } else if (!strcmp(parameter, "destination")) {
965  if (*p == '=') {
966  p++;
967  get_word_sep(buf, sizeof(buf), ";,", &p);
968  get_sockaddr(s, buf, &th->destination);
969  }
970  } else if (!strcmp(parameter, "source")) {
971  if (*p == '=') {
972  p++;
973  get_word_sep(buf, sizeof(buf), ";,", &p);
974  av_strlcpy(th->source, buf, sizeof(th->source));
975  }
976  } else if (!strcmp(parameter, "mode")) {
977  if (*p == '=') {
978  p++;
979  get_word_sep(buf, sizeof(buf), ";, ", &p);
980  if (!strcmp(buf, "record") ||
981  !strcmp(buf, "receive"))
982  th->mode_record = 1;
983  }
984  }
985 
986  while (*p != ';' && *p != '\0' && *p != ',')
987  p++;
988  if (*p == ';')
989  p++;
990  }
991  if (*p == ',')
992  p++;
993 
994  reply->nb_transports++;
995  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
996  break;
997  }
998 }
999 
1000 static void handle_rtp_info(RTSPState *rt, const char *url,
1001  uint32_t seq, uint32_t rtptime)
1002 {
1003  int i;
1004  if (!rtptime || !url[0])
1005  return;
1006  if (rt->transport != RTSP_TRANSPORT_RTP)
1007  return;
1008  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1009  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1010  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1011  if (!rtpctx)
1012  continue;
1013  if (!strcmp(rtsp_st->control_url, url)) {
1014  rtpctx->base_timestamp = rtptime;
1015  break;
1016  }
1017  }
1018 }
1019 
1020 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1021 {
1022  int read = 0;
1023  char key[20], value[1024], url[1024] = "";
1024  uint32_t seq = 0, rtptime = 0;
1025 
1026  for (;;) {
1027  p += strspn(p, SPACE_CHARS);
1028  if (!*p)
1029  break;
1030  get_word_sep(key, sizeof(key), "=", &p);
1031  if (*p != '=')
1032  break;
1033  p++;
1034  get_word_sep(value, sizeof(value), ";, ", &p);
1035  read++;
1036  if (!strcmp(key, "url"))
1037  av_strlcpy(url, value, sizeof(url));
1038  else if (!strcmp(key, "seq"))
1039  seq = strtoul(value, NULL, 10);
1040  else if (!strcmp(key, "rtptime"))
1041  rtptime = strtoul(value, NULL, 10);
1042  if (*p == ',') {
1043  handle_rtp_info(rt, url, seq, rtptime);
1044  url[0] = '\0';
1045  seq = rtptime = 0;
1046  read = 0;
1047  }
1048  if (*p)
1049  p++;
1050  }
1051  if (read > 0)
1052  handle_rtp_info(rt, url, seq, rtptime);
1053 }
1054 
1056  RTSPMessageHeader *reply, const char *buf,
1057  RTSPState *rt, const char *method)
1058 {
1059  const char *p;
1060 
1061  /* NOTE: we do case independent match for broken servers */
1062  p = buf;
1063  if (av_stristart(p, "Session:", &p)) {
1064  int t;
1065  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1066  if (av_stristart(p, ";timeout=", &p) &&
1067  (t = strtol(p, NULL, 10)) > 0) {
1068  reply->timeout = t;
1069  }
1070  } else if (av_stristart(p, "Content-Length:", &p)) {
1071  reply->content_length = strtol(p, NULL, 10);
1072  } else if (av_stristart(p, "Transport:", &p)) {
1073  rtsp_parse_transport(s, reply, p);
1074  } else if (av_stristart(p, "CSeq:", &p)) {
1075  reply->seq = strtol(p, NULL, 10);
1076  } else if (av_stristart(p, "Range:", &p)) {
1077  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1078  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1079  p += strspn(p, SPACE_CHARS);
1080  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1081  } else if (av_stristart(p, "Server:", &p)) {
1082  p += strspn(p, SPACE_CHARS);
1083  av_strlcpy(reply->server, p, sizeof(reply->server));
1084  } else if (av_stristart(p, "Notice:", &p) ||
1085  av_stristart(p, "X-Notice:", &p)) {
1086  reply->notice = strtol(p, NULL, 10);
1087  } else if (av_stristart(p, "Location:", &p)) {
1088  p += strspn(p, SPACE_CHARS);
1089  av_strlcpy(reply->location, p , sizeof(reply->location));
1090  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1091  p += strspn(p, SPACE_CHARS);
1092  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1093  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1094  p += strspn(p, SPACE_CHARS);
1095  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1096  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1097  p += strspn(p, SPACE_CHARS);
1098  if (method && !strcmp(method, "DESCRIBE"))
1099  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1100  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1101  p += strspn(p, SPACE_CHARS);
1102  if (method && !strcmp(method, "PLAY"))
1103  rtsp_parse_rtp_info(rt, p);
1104  } else if (av_stristart(p, "Public:", &p) && rt) {
1105  if (strstr(p, "GET_PARAMETER") &&
1106  method && !strcmp(method, "OPTIONS"))
1107  rt->get_parameter_supported = 1;
1108  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1109  p += strspn(p, SPACE_CHARS);
1110  rt->accept_dynamic_rate = atoi(p);
1111  } else if (av_stristart(p, "Content-Type:", &p)) {
1112  p += strspn(p, SPACE_CHARS);
1113  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1114  }
1115 }
1116 
1117 /* skip a RTP/TCP interleaved packet */
1119 {
1120  RTSPState *rt = s->priv_data;
1121  int ret, len, len1;
1122  uint8_t buf[1024];
1123 
1124  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1125  if (ret != 3)
1126  return;
1127  len = AV_RB16(buf + 1);
1128 
1129  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1130 
1131  /* skip payload */
1132  while (len > 0) {
1133  len1 = len;
1134  if (len1 > sizeof(buf))
1135  len1 = sizeof(buf);
1136  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1137  if (ret != len1)
1138  return;
1139  len -= len1;
1140  }
1141 }
1142 
1144  unsigned char **content_ptr,
1145  int return_on_interleaved_data, const char *method)
1146 {
1147  RTSPState *rt = s->priv_data;
1148  char buf[4096], buf1[1024], *q;
1149  unsigned char ch;
1150  const char *p;
1151  int ret, content_length, line_count = 0, request = 0;
1152  unsigned char *content = NULL;
1153 
1154 start:
1155  line_count = 0;
1156  request = 0;
1157  content = NULL;
1158  memset(reply, 0, sizeof(*reply));
1159 
1160  /* parse reply (XXX: use buffers) */
1161  rt->last_reply[0] = '\0';
1162  for (;;) {
1163  q = buf;
1164  for (;;) {
1165  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1166  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1167  if (ret != 1)
1168  return AVERROR_EOF;
1169  if (ch == '\n')
1170  break;
1171  if (ch == '$' && q == buf) {
1172  if (return_on_interleaved_data) {
1173  return 1;
1174  } else
1176  } else if (ch != '\r') {
1177  if ((q - buf) < sizeof(buf) - 1)
1178  *q++ = ch;
1179  }
1180  }
1181  *q = '\0';
1182 
1183  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1184 
1185  /* test if last line */
1186  if (buf[0] == '\0')
1187  break;
1188  p = buf;
1189  if (line_count == 0) {
1190  /* get reply code */
1191  get_word(buf1, sizeof(buf1), &p);
1192  if (!strncmp(buf1, "RTSP/", 5)) {
1193  get_word(buf1, sizeof(buf1), &p);
1194  reply->status_code = atoi(buf1);
1195  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1196  } else {
1197  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1198  get_word(buf1, sizeof(buf1), &p); // object
1199  request = 1;
1200  }
1201  } else {
1202  ff_rtsp_parse_line(s, reply, p, rt, method);
1203  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1204  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1205  }
1206  line_count++;
1207  }
1208 
1209  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1210  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1211 
1212  content_length = reply->content_length;
1213  if (content_length > 0) {
1214  /* leave some room for a trailing '\0' (useful for simple parsing) */
1215  content = av_malloc(content_length + 1);
1216  if (!content)
1217  return AVERROR(ENOMEM);
1218  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1219  content[content_length] = '\0';
1220  }
1221  if (content_ptr)
1222  *content_ptr = content;
1223  else
1224  av_freep(&content);
1225 
1226  if (request) {
1227  char buf[1024];
1228  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1229  const char* ptr = buf;
1230 
1231  if (!strcmp(reply->reason, "OPTIONS")) {
1232  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1233  if (reply->seq)
1234  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1235  if (reply->session_id[0])
1236  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1237  reply->session_id);
1238  } else {
1239  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1240  }
1241  av_strlcat(buf, "\r\n", sizeof(buf));
1242 
1243  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1244  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1245  ptr = base64buf;
1246  }
1247  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1248 
1250  /* Even if the request from the server had data, it is not the data
1251  * that the caller wants or expects. The memory could also be leaked
1252  * if the actual following reply has content data. */
1253  if (content_ptr)
1254  av_freep(content_ptr);
1255  /* If method is set, this is called from ff_rtsp_send_cmd,
1256  * where a reply to exactly this request is awaited. For
1257  * callers from within packet receiving, we just want to
1258  * return to the caller and go back to receiving packets. */
1259  if (method)
1260  goto start;
1261  return 0;
1262  }
1263 
1264  if (rt->seq != reply->seq) {
1265  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1266  rt->seq, reply->seq);
1267  }
1268 
1269  /* EOS */
1270  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1271  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1272  reply->notice == 2306 /* Continuous Feed Terminated */) {
1273  rt->state = RTSP_STATE_IDLE;
1274  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1275  return AVERROR(EIO); /* data or server error */
1276  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1277  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1278  return AVERROR(EPERM);
1279 
1280  return 0;
1281 }
1282 
1283 /**
1284  * Send a command to the RTSP server without waiting for the reply.
1285  *
1286  * @param s RTSP (de)muxer context
1287  * @param method the method for the request
1288  * @param url the target url for the request
1289  * @param headers extra header lines to include in the request
1290  * @param send_content if non-null, the data to send as request body content
1291  * @param send_content_length the length of the send_content data, or 0 if
1292  * send_content is null
1293  *
1294  * @return zero if success, nonzero otherwise
1295  */
1296 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1297  const char *method, const char *url,
1298  const char *headers,
1299  const unsigned char *send_content,
1300  int send_content_length)
1301 {
1302  RTSPState *rt = s->priv_data;
1303  char buf[4096], *out_buf;
1304  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1305 
1306  /* Add in RTSP headers */
1307  out_buf = buf;
1308  rt->seq++;
1309  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1310  if (headers)
1311  av_strlcat(buf, headers, sizeof(buf));
1312  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1313  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1314  if (rt->session_id[0] != '\0' && (!headers ||
1315  !strstr(headers, "\nIf-Match:"))) {
1316  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1317  }
1318  if (rt->auth[0]) {
1319  char *str = ff_http_auth_create_response(&rt->auth_state,
1320  rt->auth, url, method);
1321  if (str)
1322  av_strlcat(buf, str, sizeof(buf));
1323  av_free(str);
1324  }
1325  if (send_content_length > 0 && send_content)
1326  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1327  av_strlcat(buf, "\r\n", sizeof(buf));
1328 
1329  /* base64 encode rtsp if tunneling */
1330  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1331  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1332  out_buf = base64buf;
1333  }
1334 
1335  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1336 
1337  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1338  if (send_content_length > 0 && send_content) {
1339  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1340  avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1341  return AVERROR_PATCHWELCOME;
1342  }
1343  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1344  }
1346 
1347  return 0;
1348 }
1349 
1350 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1351  const char *url, const char *headers)
1352 {
1353  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1354 }
1355 
1356 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1357  const char *headers, RTSPMessageHeader *reply,
1358  unsigned char **content_ptr)
1359 {
1360  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1361  content_ptr, NULL, 0);
1362 }
1363 
1365  const char *method, const char *url,
1366  const char *header,
1367  RTSPMessageHeader *reply,
1368  unsigned char **content_ptr,
1369  const unsigned char *send_content,
1370  int send_content_length)
1371 {
1372  RTSPState *rt = s->priv_data;
1373  HTTPAuthType cur_auth_type;
1374  int ret, attempts = 0;
1375 
1376 retry:
1377  cur_auth_type = rt->auth_state.auth_type;
1378  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1379  send_content,
1380  send_content_length)))
1381  return ret;
1382 
1383  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1384  return ret;
1385  attempts++;
1386 
1387  if (reply->status_code == 401 &&
1388  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1389  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1390  goto retry;
1391 
1392  if (reply->status_code > 400){
1393  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1394  method,
1395  reply->status_code,
1396  reply->reason);
1397  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1398  }
1399 
1400  return 0;
1401 }
1402 
1403 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1404  int lower_transport, const char *real_challenge)
1405 {
1406  RTSPState *rt = s->priv_data;
1407  int rtx = 0, j, i, err, interleave = 0, port_off;
1408  RTSPStream *rtsp_st;
1409  RTSPMessageHeader reply1, *reply = &reply1;
1410  char cmd[2048];
1411  const char *trans_pref;
1412 
1413  if (rt->transport == RTSP_TRANSPORT_RDT)
1414  trans_pref = "x-pn-tng";
1415  else if (rt->transport == RTSP_TRANSPORT_RAW)
1416  trans_pref = "RAW/RAW";
1417  else
1418  trans_pref = "RTP/AVP";
1419 
1420  /* default timeout: 1 minute */
1421  rt->timeout = 60;
1422 
1423  /* Choose a random starting offset within the first half of the
1424  * port range, to allow for a number of ports to try even if the offset
1425  * happens to be at the end of the random range. */
1426  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1427  /* even random offset */
1428  port_off -= port_off & 0x01;
1429 
1430  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1431  char transport[2048];
1432 
1433  /*
1434  * WMS serves all UDP data over a single connection, the RTX, which
1435  * isn't necessarily the first in the SDP but has to be the first
1436  * to be set up, else the second/third SETUP will fail with a 461.
1437  */
1438  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1439  rt->server_type == RTSP_SERVER_WMS) {
1440  if (i == 0) {
1441  /* rtx first */
1442  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1443  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1444  if (len >= 4 &&
1445  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1446  "/rtx"))
1447  break;
1448  }
1449  if (rtx == rt->nb_rtsp_streams)
1450  return -1; /* no RTX found */
1451  rtsp_st = rt->rtsp_streams[rtx];
1452  } else
1453  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1454  } else
1455  rtsp_st = rt->rtsp_streams[i];
1456 
1457  /* RTP/UDP */
1458  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1459  char buf[256];
1460 
1461  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1462  port = reply->transports[0].client_port_min;
1463  goto have_port;
1464  }
1465 
1466  /* first try in specified port range */
1467  while (j <= rt->rtp_port_max) {
1468  AVDictionary *opts = map_to_opts(rt);
1469 
1470  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1471  "?localport=%d", j);
1472  /* we will use two ports per rtp stream (rtp and rtcp) */
1473  j += 2;
1476 
1477  av_dict_free(&opts);
1478 
1479  if (!err)
1480  goto rtp_opened;
1481  }
1482  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1483  err = AVERROR(EIO);
1484  goto fail;
1485 
1486  rtp_opened:
1487  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1488  have_port:
1489  snprintf(transport, sizeof(transport) - 1,
1490  "%s/UDP;", trans_pref);
1491  if (rt->server_type != RTSP_SERVER_REAL)
1492  av_strlcat(transport, "unicast;", sizeof(transport));
1493  av_strlcatf(transport, sizeof(transport),
1494  "client_port=%d", port);
1495  if (rt->transport == RTSP_TRANSPORT_RTP &&
1496  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1497  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1498  }
1499 
1500  /* RTP/TCP */
1501  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1502  /* For WMS streams, the application streams are only used for
1503  * UDP. When trying to set it up for TCP streams, the server
1504  * will return an error. Therefore, we skip those streams. */
1505  if (rt->server_type == RTSP_SERVER_WMS &&
1506  (rtsp_st->stream_index < 0 ||
1507  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1509  continue;
1510  snprintf(transport, sizeof(transport) - 1,
1511  "%s/TCP;", trans_pref);
1512  if (rt->transport != RTSP_TRANSPORT_RDT)
1513  av_strlcat(transport, "unicast;", sizeof(transport));
1514  av_strlcatf(transport, sizeof(transport),
1515  "interleaved=%d-%d",
1516  interleave, interleave + 1);
1517  interleave += 2;
1518  }
1519 
1520  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1521  snprintf(transport, sizeof(transport) - 1,
1522  "%s/UDP;multicast", trans_pref);
1523  }
1524  if (s->oformat) {
1525  av_strlcat(transport, ";mode=record", sizeof(transport));
1526  } else if (rt->server_type == RTSP_SERVER_REAL ||
1528  av_strlcat(transport, ";mode=play", sizeof(transport));
1529  snprintf(cmd, sizeof(cmd),
1530  "Transport: %s\r\n",
1531  transport);
1532  if (rt->accept_dynamic_rate)
1533  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1534  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1535  char real_res[41], real_csum[9];
1536  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1537  real_challenge);
1538  av_strlcatf(cmd, sizeof(cmd),
1539  "If-Match: %s\r\n"
1540  "RealChallenge2: %s, sd=%s\r\n",
1541  rt->session_id, real_res, real_csum);
1542  }
1543  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1544  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1545  err = 1;
1546  goto fail;
1547  } else if (reply->status_code != RTSP_STATUS_OK ||
1548  reply->nb_transports != 1) {
1550  goto fail;
1551  }
1552 
1553  /* XXX: same protocol for all streams is required */
1554  if (i > 0) {
1555  if (reply->transports[0].lower_transport != rt->lower_transport ||
1556  reply->transports[0].transport != rt->transport) {
1557  err = AVERROR_INVALIDDATA;
1558  goto fail;
1559  }
1560  } else {
1561  rt->lower_transport = reply->transports[0].lower_transport;
1562  rt->transport = reply->transports[0].transport;
1563  }
1564 
1565  /* Fail if the server responded with another lower transport mode
1566  * than what we requested. */
1567  if (reply->transports[0].lower_transport != lower_transport) {
1568  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1569  err = AVERROR_INVALIDDATA;
1570  goto fail;
1571  }
1572 
1573  switch(reply->transports[0].lower_transport) {
1575  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1576  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1577  break;
1578 
1579  case RTSP_LOWER_TRANSPORT_UDP: {
1580  char url[1024], options[30] = "";
1581  const char *peer = host;
1582 
1583  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1584  av_strlcpy(options, "?connect=1", sizeof(options));
1585  /* Use source address if specified */
1586  if (reply->transports[0].source[0])
1587  peer = reply->transports[0].source;
1588  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1589  reply->transports[0].server_port_min, "%s", options);
1590  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1591  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1592  err = AVERROR_INVALIDDATA;
1593  goto fail;
1594  }
1595  break;
1596  }
1598  char url[1024], namebuf[50], optbuf[20] = "";
1599  struct sockaddr_storage addr;
1600  int port, ttl;
1601 
1602  if (reply->transports[0].destination.ss_family) {
1603  addr = reply->transports[0].destination;
1604  port = reply->transports[0].port_min;
1605  ttl = reply->transports[0].ttl;
1606  } else {
1607  addr = rtsp_st->sdp_ip;
1608  port = rtsp_st->sdp_port;
1609  ttl = rtsp_st->sdp_ttl;
1610  }
1611  if (ttl > 0)
1612  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1613  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1614  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1615  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1616  port, "%s", optbuf);
1619  err = AVERROR_INVALIDDATA;
1620  goto fail;
1621  }
1622  break;
1623  }
1624  }
1625 
1626  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1627  goto fail;
1628  }
1629 
1630  if (rt->nb_rtsp_streams && reply->timeout > 0)
1631  rt->timeout = reply->timeout;
1632 
1633  if (rt->server_type == RTSP_SERVER_REAL)
1634  rt->need_subscription = 1;
1635 
1636  return 0;
1637 
1638 fail:
1639  ff_rtsp_undo_setup(s, 0);
1640  return err;
1641 }
1642 
1644 {
1645  RTSPState *rt = s->priv_data;
1646  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1647  ffurl_close(rt->rtsp_hd);
1648  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1649 }
1650 
1652 {
1653  RTSPState *rt = s->priv_data;
1654  char proto[128], host[1024], path[1024];
1655  char tcpname[1024], cmd[2048], auth[128];
1656  const char *lower_rtsp_proto = "tcp";
1657  int port, err, tcp_fd;
1658  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1659  int lower_transport_mask = 0;
1660  int default_port = RTSP_DEFAULT_PORT;
1661  char real_challenge[64] = "";
1662  struct sockaddr_storage peer;
1663  socklen_t peer_len = sizeof(peer);
1664 
1665  if (rt->rtp_port_max < rt->rtp_port_min) {
1666  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1667  "than min port %d\n", rt->rtp_port_max,
1668  rt->rtp_port_min);
1669  return AVERROR(EINVAL);
1670  }
1671 
1672  if (!ff_network_init())
1673  return AVERROR(EIO);
1674 
1675  if (s->max_delay < 0) /* Not set by the caller */
1677 
1682  }
1683  /* Only pass through valid flags from here */
1685 
1686 redirect:
1687  /* extract hostname and port */
1688  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1689  host, sizeof(host), &port, path, sizeof(path), s->filename);
1690 
1691  if (!strcmp(proto, "rtsps")) {
1692  lower_rtsp_proto = "tls";
1693  default_port = RTSPS_DEFAULT_PORT;
1695  }
1696 
1697  if (*auth) {
1698  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1699  }
1700  if (port < 0)
1701  port = default_port;
1702 
1703  lower_transport_mask = rt->lower_transport_mask;
1704 
1705  if (!lower_transport_mask)
1706  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1707 
1708  if (s->oformat) {
1709  /* Only UDP or TCP - UDP multicast isn't supported. */
1710  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1711  (1 << RTSP_LOWER_TRANSPORT_TCP);
1712  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1713  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1714  "only UDP and TCP are supported for output.\n");
1715  err = AVERROR(EINVAL);
1716  goto fail;
1717  }
1718  }
1719 
1720  /* Construct the URI used in request; this is similar to s->filename,
1721  * but with authentication credentials removed and RTSP specific options
1722  * stripped out. */
1723  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1724  host, port, "%s", path);
1725 
1726  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1727  /* set up initial handshake for tunneling */
1728  char httpname[1024];
1729  char sessioncookie[17];
1730  char headers[1024];
1731 
1732  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1733  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1735 
1736  /* GET requests */
1737  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1738  &s->interrupt_callback) < 0) {
1739  err = AVERROR(EIO);
1740  goto fail;
1741  }
1742 
1743  /* generate GET headers */
1744  snprintf(headers, sizeof(headers),
1745  "x-sessioncookie: %s\r\n"
1746  "Accept: application/x-rtsp-tunnelled\r\n"
1747  "Pragma: no-cache\r\n"
1748  "Cache-Control: no-cache\r\n",
1749  sessioncookie);
1750  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1751 
1752  if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1754  if (!rt->rtsp_hd->protocol_whitelist) {
1755  err = AVERROR(ENOMEM);
1756  goto fail;
1757  }
1758  }
1759 
1760  /* complete the connection */
1761  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1762  err = AVERROR(EIO);
1763  goto fail;
1764  }
1765 
1766  /* POST requests */
1767  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1768  &s->interrupt_callback) < 0 ) {
1769  err = AVERROR(EIO);
1770  goto fail;
1771  }
1772 
1773  /* generate POST headers */
1774  snprintf(headers, sizeof(headers),
1775  "x-sessioncookie: %s\r\n"
1776  "Content-Type: application/x-rtsp-tunnelled\r\n"
1777  "Pragma: no-cache\r\n"
1778  "Cache-Control: no-cache\r\n"
1779  "Content-Length: 32767\r\n"
1780  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1781  sessioncookie);
1782  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1783  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1784 
1785  /* Initialize the authentication state for the POST session. The HTTP
1786  * protocol implementation doesn't properly handle multi-pass
1787  * authentication for POST requests, since it would require one of
1788  * the following:
1789  * - implementing Expect: 100-continue, which many HTTP servers
1790  * don't support anyway, even less the RTSP servers that do HTTP
1791  * tunneling
1792  * - sending the whole POST data until getting a 401 reply specifying
1793  * what authentication method to use, then resending all that data
1794  * - waiting for potential 401 replies directly after sending the
1795  * POST header (waiting for some unspecified time)
1796  * Therefore, we copy the full auth state, which works for both basic
1797  * and digest. (For digest, we would have to synchronize the nonce
1798  * count variable between the two sessions, if we'd do more requests
1799  * with the original session, though.)
1800  */
1802 
1803  /* complete the connection */
1804  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1805  err = AVERROR(EIO);
1806  goto fail;
1807  }
1808  } else {
1809  int ret;
1810  /* open the tcp connection */
1811  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1812  host, port,
1813  "?timeout=%d", rt->stimeout);
1814  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1816  err = ret;
1817  goto fail;
1818  }
1819  rt->rtsp_hd_out = rt->rtsp_hd;
1820  }
1821  rt->seq = 0;
1822 
1823  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1824  if (tcp_fd < 0) {
1825  err = tcp_fd;
1826  goto fail;
1827  }
1828  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1829  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1830  NULL, 0, NI_NUMERICHOST);
1831  }
1832 
1833  /* request options supported by the server; this also detects server
1834  * type */
1835  for (rt->server_type = RTSP_SERVER_RTP;;) {
1836  cmd[0] = 0;
1837  if (rt->server_type == RTSP_SERVER_REAL)
1838  av_strlcat(cmd,
1839  /*
1840  * The following entries are required for proper
1841  * streaming from a Realmedia server. They are
1842  * interdependent in some way although we currently
1843  * don't quite understand how. Values were copied
1844  * from mplayer SVN r23589.
1845  * ClientChallenge is a 16-byte ID in hex
1846  * CompanyID is a 16-byte ID in base64
1847  */
1848  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1849  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1850  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1851  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1852  sizeof(cmd));
1853  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1854  if (reply->status_code != RTSP_STATUS_OK) {
1856  goto fail;
1857  }
1858 
1859  /* detect server type if not standard-compliant RTP */
1860  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1862  continue;
1863  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1865  } else if (rt->server_type == RTSP_SERVER_REAL)
1866  strcpy(real_challenge, reply->real_challenge);
1867  break;
1868  }
1869 
1870  if (CONFIG_RTSP_DEMUXER && s->iformat)
1871  err = ff_rtsp_setup_input_streams(s, reply);
1872  else if (CONFIG_RTSP_MUXER)
1873  err = ff_rtsp_setup_output_streams(s, host);
1874  else
1875  av_assert0(0);
1876  if (err)
1877  goto fail;
1878 
1879  do {
1880  int lower_transport = ff_log2_tab[lower_transport_mask &
1881  ~(lower_transport_mask - 1)];
1882 
1883  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1884  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1885  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1886 
1887  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1888  rt->server_type == RTSP_SERVER_REAL ?
1889  real_challenge : NULL);
1890  if (err < 0)
1891  goto fail;
1892  lower_transport_mask &= ~(1 << lower_transport);
1893  if (lower_transport_mask == 0 && err == 1) {
1894  err = AVERROR(EPROTONOSUPPORT);
1895  goto fail;
1896  }
1897  } while (err);
1898 
1899  rt->lower_transport_mask = lower_transport_mask;
1900  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1901  rt->state = RTSP_STATE_IDLE;
1902  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1903  return 0;
1904  fail:
1907  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1908  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1909  rt->session_id[0] = '\0';
1910  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1911  reply->status_code,
1912  s->filename);
1913  goto redirect;
1914  }
1915  ff_network_close();
1916  return err;
1917 }
1918 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1919 
1920 #if CONFIG_RTPDEC
1921 static int parse_rtsp_message(AVFormatContext *s)
1922 {
1923  RTSPState *rt = s->priv_data;
1924  int ret;
1925 
1926  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1927  if (rt->state == RTSP_STATE_STREAMING) {
1929  return AVERROR_EOF;
1930  else
1932  "Unable to answer to TEARDOWN\n");
1933  } else
1934  return 0;
1935  } else {
1936  RTSPMessageHeader reply;
1937  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1938  if (ret < 0)
1939  return ret;
1940  /* XXX: parse message */
1941  if (rt->state != RTSP_STATE_STREAMING)
1942  return 0;
1943  }
1944 
1945  return 0;
1946 }
1947 
1948 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1949  uint8_t *buf, int buf_size, int64_t wait_end)
1950 {
1951  RTSPState *rt = s->priv_data;
1952  RTSPStream *rtsp_st;
1953  int n, i, ret, timeout_cnt = 0;
1954  struct pollfd *p = rt->p;
1955  int *fds = NULL, fdsnum, fdsidx;
1956 
1957  if (!p) {
1958  p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
1959  if (!p)
1960  return AVERROR(ENOMEM);
1961 
1962  if (rt->rtsp_hd) {
1963  p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1964  p[rt->max_p++].events = POLLIN;
1965  }
1966  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1967  rtsp_st = rt->rtsp_streams[i];
1968  if (rtsp_st->rtp_handle) {
1969  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1970  &fds, &fdsnum)) {
1971  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1972  return ret;
1973  }
1974  if (fdsnum != 2) {
1975  av_log(s, AV_LOG_ERROR,
1976  "Number of fds %d not supported\n", fdsnum);
1977  return AVERROR_INVALIDDATA;
1978  }
1979  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1980  p[rt->max_p].fd = fds[fdsidx];
1981  p[rt->max_p++].events = POLLIN;
1982  }
1983  av_freep(&fds);
1984  }
1985  }
1986  }
1987 
1988  for (;;) {
1990  return AVERROR_EXIT;
1991  if (wait_end && wait_end - av_gettime_relative() < 0)
1992  return AVERROR(EAGAIN);
1993  n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
1994  if (n > 0) {
1995  int j = rt->rtsp_hd ? 1 : 0;
1996  timeout_cnt = 0;
1997  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1998  rtsp_st = rt->rtsp_streams[i];
1999  if (rtsp_st->rtp_handle) {
2000  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2001  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2002  if (ret > 0) {
2003  *prtsp_st = rtsp_st;
2004  return ret;
2005  }
2006  }
2007  j+=2;
2008  }
2009  }
2010 #if CONFIG_RTSP_DEMUXER
2011  if (rt->rtsp_hd && p[0].revents & POLLIN) {
2012  return parse_rtsp_message(s);
2013  }
2014 #endif
2015  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2016  return AVERROR(ETIMEDOUT);
2017  } else if (n < 0 && errno != EINTR)
2018  return AVERROR(errno);
2019  }
2020 }
2021 
2022 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2023  const uint8_t *buf, int len)
2024 {
2025  RTSPState *rt = s->priv_data;
2026  int i;
2027  if (len < 0)
2028  return len;
2029  if (rt->nb_rtsp_streams == 1) {
2030  *rtsp_st = rt->rtsp_streams[0];
2031  return len;
2032  }
2033  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2034  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2035  int no_ssrc = 0;
2036  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2037  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2038  if (!rtpctx)
2039  continue;
2040  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2041  *rtsp_st = rt->rtsp_streams[i];
2042  return len;
2043  }
2044  if (!rtpctx->ssrc)
2045  no_ssrc = 1;
2046  }
2047  if (no_ssrc) {
2049  "Unable to pick stream for packet - SSRC not known for "
2050  "all streams\n");
2051  return AVERROR(EAGAIN);
2052  }
2053  } else {
2054  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2055  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2056  *rtsp_st = rt->rtsp_streams[i];
2057  return len;
2058  }
2059  }
2060  }
2061  }
2062  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2063  return AVERROR(EAGAIN);
2064 }
2065 
2066 static int read_packet(AVFormatContext *s,
2067  RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2068  int64_t wait_end)
2069 {
2070  RTSPState *rt = s->priv_data;
2071  int len;
2072 
2073  switch(rt->lower_transport) {
2074  default:
2075 #if CONFIG_RTSP_DEMUXER
2077  len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2078  break;
2079 #endif
2082  len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2083  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2084  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2085  break;
2087  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2088  wait_end && wait_end < av_gettime_relative())
2089  len = AVERROR(EAGAIN);
2090  else
2091  len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2092  len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2093  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2094  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2095  break;
2096  }
2097 
2098  if (len == 0)
2099  return AVERROR_EOF;
2100 
2101  return len;
2102 }
2103 
2105 {
2106  RTSPState *rt = s->priv_data;
2107  int ret, len;
2108  RTSPStream *rtsp_st, *first_queue_st = NULL;
2109  int64_t wait_end = 0;
2110 
2111  if (rt->nb_byes == rt->nb_rtsp_streams)
2112  return AVERROR_EOF;
2113 
2114  /* get next frames from the same RTP packet */
2115  if (rt->cur_transport_priv) {
2116  if (rt->transport == RTSP_TRANSPORT_RDT) {
2117  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2118  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2119  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2120  } else if (CONFIG_RTPDEC && rt->ts) {
2121  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2122  if (ret >= 0) {
2123  rt->recvbuf_pos += ret;
2124  ret = rt->recvbuf_pos < rt->recvbuf_len;
2125  }
2126  } else
2127  ret = -1;
2128  if (ret == 0) {
2129  rt->cur_transport_priv = NULL;
2130  return 0;
2131  } else if (ret == 1) {
2132  return 0;
2133  } else
2134  rt->cur_transport_priv = NULL;
2135  }
2136 
2137 redo:
2138  if (rt->transport == RTSP_TRANSPORT_RTP) {
2139  int i;
2140  int64_t first_queue_time = 0;
2141  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2142  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2143  int64_t queue_time;
2144  if (!rtpctx)
2145  continue;
2146  queue_time = ff_rtp_queued_packet_time(rtpctx);
2147  if (queue_time && (queue_time - first_queue_time < 0 ||
2148  !first_queue_time)) {
2149  first_queue_time = queue_time;
2150  first_queue_st = rt->rtsp_streams[i];
2151  }
2152  }
2153  if (first_queue_time) {
2154  wait_end = first_queue_time + s->max_delay;
2155  } else {
2156  wait_end = 0;
2157  first_queue_st = NULL;
2158  }
2159  }
2160 
2161  /* read next RTP packet */
2162  if (!rt->recvbuf) {
2164  if (!rt->recvbuf)
2165  return AVERROR(ENOMEM);
2166  }
2167 
2168  len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2169  if (len == AVERROR(EAGAIN) && first_queue_st &&
2170  rt->transport == RTSP_TRANSPORT_RTP) {
2172  "max delay reached. need to consume packet\n");
2173  rtsp_st = first_queue_st;
2174  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2175  goto end;
2176  }
2177  if (len < 0)
2178  return len;
2179 
2180  if (rt->transport == RTSP_TRANSPORT_RDT) {
2181  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2182  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2183  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2184  if (rtsp_st->feedback) {
2185  AVIOContext *pb = NULL;
2187  pb = s->pb;
2188  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2189  }
2190  if (ret < 0) {
2191  /* Either bad packet, or a RTCP packet. Check if the
2192  * first_rtcp_ntp_time field was initialized. */
2193  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2194  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2195  /* first_rtcp_ntp_time has been initialized for this stream,
2196  * copy the same value to all other uninitialized streams,
2197  * in order to map their timestamp origin to the same ntp time
2198  * as this one. */
2199  int i;
2200  AVStream *st = NULL;
2201  if (rtsp_st->stream_index >= 0)
2202  st = s->streams[rtsp_st->stream_index];
2203  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2204  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2205  AVStream *st2 = NULL;
2206  if (rt->rtsp_streams[i]->stream_index >= 0)
2207  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2208  if (rtpctx2 && st && st2 &&
2209  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2210  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2211  rtpctx2->rtcp_ts_offset = av_rescale_q(
2212  rtpctx->rtcp_ts_offset, st->time_base,
2213  st2->time_base);
2214  }
2215  }
2216  // Make real NTP start time available in AVFormatContext
2217  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2218  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2219  if (rtpctx->st) {
2220  s->start_time_realtime -=
2221  av_rescale (rtpctx->rtcp_ts_offset,
2222  (uint64_t) rtpctx->st->time_base.num * 1000000,
2223  rtpctx->st->time_base.den);
2224  }
2225  }
2226  }
2227  if (ret == -RTCP_BYE) {
2228  rt->nb_byes++;
2229 
2230  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2231  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2232 
2233  if (rt->nb_byes == rt->nb_rtsp_streams)
2234  return AVERROR_EOF;
2235  }
2236  }
2237  } else if (CONFIG_RTPDEC && rt->ts) {
2238  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2239  if (ret >= 0) {
2240  if (ret < len) {
2241  rt->recvbuf_len = len;
2242  rt->recvbuf_pos = ret;
2243  rt->cur_transport_priv = rt->ts;
2244  return 1;
2245  } else {
2246  ret = 0;
2247  }
2248  }
2249  } else {
2250  return AVERROR_INVALIDDATA;
2251  }
2252 end:
2253  if (ret < 0)
2254  goto redo;
2255  if (ret == 1)
2256  /* more packets may follow, so we save the RTP context */
2257  rt->cur_transport_priv = rtsp_st->transport_priv;
2258 
2259  return ret;
2260 }
2261 #endif /* CONFIG_RTPDEC */
2262 
2263 #if CONFIG_SDP_DEMUXER
2264 static int sdp_probe(AVProbeData *p1)
2265 {
2266  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2267 
2268  /* we look for a line beginning "c=IN IP" */
2269  while (p < p_end && *p != '\0') {
2270  if (sizeof("c=IN IP") - 1 < p_end - p &&
2271  av_strstart(p, "c=IN IP", NULL))
2272  return AVPROBE_SCORE_EXTENSION;
2273 
2274  while (p < p_end - 1 && *p != '\n') p++;
2275  if (++p >= p_end)
2276  break;
2277  if (*p == '\r')
2278  p++;
2279  }
2280  return 0;
2281 }
2282 
2283 static void append_source_addrs(char *buf, int size, const char *name,
2284  int count, struct RTSPSource **addrs)
2285 {
2286  int i;
2287  if (!count)
2288  return;
2289  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2290  for (i = 1; i < count; i++)
2291  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2292 }
2293 
2294 static int sdp_read_header(AVFormatContext *s)
2295 {
2296  RTSPState *rt = s->priv_data;
2297  RTSPStream *rtsp_st;
2298  int size, i, err;
2299  char *content;
2300  char url[1024];
2301 
2302  if (!ff_network_init())
2303  return AVERROR(EIO);
2304 
2305  if (s->max_delay < 0) /* Not set by the caller */
2307  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2309 
2310  /* read the whole sdp file */
2311  /* XXX: better loading */
2312  content = av_malloc(SDP_MAX_SIZE);
2313  if (!content)
2314  return AVERROR(ENOMEM);
2315  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2316  if (size <= 0) {
2317  av_free(content);
2318  return AVERROR_INVALIDDATA;
2319  }
2320  content[size] ='\0';
2321 
2322  err = ff_sdp_parse(s, content);
2323  av_freep(&content);
2324  if (err) goto fail;
2325 
2326  /* open each RTP stream */
2327  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2328  char namebuf[50];
2329  rtsp_st = rt->rtsp_streams[i];
2330 
2331  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2332  AVDictionary *opts = map_to_opts(rt);
2333 
2334  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2335  sizeof(rtsp_st->sdp_ip),
2336  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2337  if (err) {
2338  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2339  err = AVERROR(EIO);
2340  av_dict_free(&opts);
2341  goto fail;
2342  }
2343  ff_url_join(url, sizeof(url), "rtp", NULL,
2344  namebuf, rtsp_st->sdp_port,
2345  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2346  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2347  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2348  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2349 
2350  append_source_addrs(url, sizeof(url), "sources",
2351  rtsp_st->nb_include_source_addrs,
2352  rtsp_st->include_source_addrs);
2353  append_source_addrs(url, sizeof(url), "block",
2354  rtsp_st->nb_exclude_source_addrs,
2355  rtsp_st->exclude_source_addrs);
2356  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2358 
2359  av_dict_free(&opts);
2360 
2361  if (err < 0) {
2362  err = AVERROR_INVALIDDATA;
2363  goto fail;
2364  }
2365  }
2366  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2367  goto fail;
2368  }
2369  return 0;
2370 fail:
2372  ff_network_close();
2373  return err;
2374 }
2375 
2376 static int sdp_read_close(AVFormatContext *s)
2377 {
2379  ff_network_close();
2380  return 0;
2381 }
2382 
2383 static const AVClass sdp_demuxer_class = {
2384  .class_name = "SDP demuxer",
2385  .item_name = av_default_item_name,
2386  .option = sdp_options,
2387  .version = LIBAVUTIL_VERSION_INT,
2388 };
2389 
2390 AVInputFormat ff_sdp_demuxer = {
2391  .name = "sdp",
2392  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2393  .priv_data_size = sizeof(RTSPState),
2394  .read_probe = sdp_probe,
2395  .read_header = sdp_read_header,
2397  .read_close = sdp_read_close,
2398  .priv_class = &sdp_demuxer_class,
2399 };
2400 #endif /* CONFIG_SDP_DEMUXER */
2401 
2402 #if CONFIG_RTP_DEMUXER
2403 static int rtp_probe(AVProbeData *p)
2404 {
2405  if (av_strstart(p->filename, "rtp:", NULL))
2406  return AVPROBE_SCORE_MAX;
2407  return 0;
2408 }
2409 
2410 static int rtp_read_header(AVFormatContext *s)
2411 {
2412  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2413  char host[500], sdp[500];
2414  int ret, port;
2415  URLContext* in = NULL;
2416  int payload_type;
2417  AVCodecParameters *par = NULL;
2418  struct sockaddr_storage addr;
2419  AVIOContext pb;
2420  socklen_t addrlen = sizeof(addr);
2421  RTSPState *rt = s->priv_data;
2422 
2423  if (!ff_network_init())
2424  return AVERROR(EIO);
2425 
2428  if (ret)
2429  goto fail;
2430 
2431  while (1) {
2432  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2433  if (ret == AVERROR(EAGAIN))
2434  continue;
2435  if (ret < 0)
2436  goto fail;
2437  if (ret < 12) {
2438  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2439  continue;
2440  }
2441 
2442  if ((recvbuf[0] & 0xc0) != 0x80) {
2443  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2444  "received\n");
2445  continue;
2446  }
2447 
2448  if (RTP_PT_IS_RTCP(recvbuf[1]))
2449  continue;
2450 
2451  payload_type = recvbuf[1] & 0x7f;
2452  break;
2453  }
2454  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2455  ffurl_close(in);
2456  in = NULL;
2457 
2458  par = avcodec_parameters_alloc();
2459  if (!par) {
2460  ret = AVERROR(ENOMEM);
2461  goto fail;
2462  }
2463 
2464  if (ff_rtp_get_codec_info(par, payload_type)) {
2465  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2466  "without an SDP file describing it\n",
2467  payload_type);
2468  goto fail;
2469  }
2470  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2471  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2472  "properly you need an SDP file "
2473  "describing it\n");
2474  }
2475 
2476  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2477  NULL, 0, s->filename);
2478 
2479  snprintf(sdp, sizeof(sdp),
2480  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2481  addr.ss_family == AF_INET ? 4 : 6, host,
2482  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2483  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2484  port, payload_type);
2485  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2487 
2488  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2489  s->pb = &pb;
2490 
2491  /* sdp_read_header initializes this again */
2492  ff_network_close();
2493 
2494  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2495 
2496  ret = sdp_read_header(s);
2497  s->pb = NULL;
2498  return ret;
2499 
2500 fail:
2502  if (in)
2503  ffurl_close(in);
2504  ff_network_close();
2505  return ret;
2506 }
2507 
2508 static const AVClass rtp_demuxer_class = {
2509  .class_name = "RTP demuxer",
2510  .item_name = av_default_item_name,
2511  .option = rtp_options,
2512  .version = LIBAVUTIL_VERSION_INT,
2513 };
2514 
2515 AVInputFormat ff_rtp_demuxer = {
2516  .name = "rtp",
2517  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2518  .priv_data_size = sizeof(RTSPState),
2519  .read_probe = rtp_probe,
2520  .read_header = rtp_read_header,
2522  .read_close = sdp_read_close,
2523  .flags = AVFMT_NOFILE,
2524  .priv_class = &rtp_demuxer_class,
2525 };
2526 #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:4621
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:565
const char * s
Definition: avisynth_c.h:768
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:1580
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:1605
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:86
const char * filename
Definition: avformat.h:462
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
Definition: rtsp.c:165
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:317
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4741
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:419
const char * desc
Definition: nvenc.c:60
#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:793
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2887
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:4152
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:890
#define AVIO_FLAG_READ
read-only
Definition: avio.h:660
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:661
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:706
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:223
void ff_network_close(void)
Definition: network.c:102
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:144
initialized and sending/receiving data
Definition: rtsp.h:197
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:270
#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:222
#define freeaddrinfo
Definition: network.h:208
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:453
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1398
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:177
This struct describes the properties of an encoded stream.
Definition: avcodec.h:4144
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:522
#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:1349
#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
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:187
int ff_network_init(void)
Definition: network.c:55
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1302
AVOptions.
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: utils.c:2258
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:106
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
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 id
Format-specific stream ID.
Definition: avformat.h:896
enum AVStreamParseType need_parsing
Definition: avformat.h:1081
#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)
#define SPACE_CHARS
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4371
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:1417
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:1887
static int flags
Definition: log.c:57
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:174
#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
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
Normal RTSP.
Definition: rtsp.h:68
const OptionDef options[]
Definition: ffserver.c:3952
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:637
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:1368
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:1567
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:631
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
const char * protocol_whitelist
Definition: url.h:49
av_default_item_name
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:179
#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:4148
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
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:2268
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:109
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:145
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:3121
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:464
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:463
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:1405
#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:185
#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:202
char filename[1024]
input or output filename
Definition: avformat.h:1425
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:130
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:140
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:156
int n
Definition: avisynth_c.h:684
AVDictionary * metadata
Definition: avformat.h:961
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
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:624
#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
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:558
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:1372
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:528
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:454
Stream structure.
Definition: avformat.h:889
#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:237
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:723
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:660
int rtp_port_max
Definition: rtsp.h:388
#define NTP_OFFSET
Definition: internal.h:240
Definition: rtp.h:100
AVIOContext * pb
I/O context.
Definition: avformat.h:1391
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:383
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
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
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:662
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
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:215
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:2906
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:739
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:757
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:283
#define s1
Definition: regdef.h:38
const char * name
Name of the codec described by this descriptor.
Definition: avcodec.h:711
#define snprintf
Definition: snprintf.h:34
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:471
int max_p
Definition: rtsp.h:355
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4306
int buffer_size
Definition: rtsp.h:411
This structure contains the data a format has to probe a file.
Definition: avformat.h:461
#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
static void interleave(short *output, short **input, int channels, int samples)
Definition: resample.c:161
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:703
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:465
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:1434
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:4262
#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:473
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:34
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:207
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:103
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
uint32_t ssrc
Definition: rtpdec.h:153
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:119
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:478
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:132
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
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:464
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:412
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:1361
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4343
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:209
#define av_free(p)
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:149
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:276
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:78
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:447
void * priv_data
Format private data.
Definition: avformat.h:1377
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:866
int channels
Audio only.
Definition: avcodec.h:4258
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:1301
char * protocol_blacklist
',' separated list of disallowed protocols.
Definition: avformat.h:1922
int ai_flags
Definition: network.h:128
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1444
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:664
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
Definition: avformat.h:1252
#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:926
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:853
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:2931
AVStream * st
Definition: rtpdec.h:151
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:445
This structure stores compressed data.
Definition: avcodec.h:1656
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:1137
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:112
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
Definition: avio.c:405
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
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
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