FFmpeg  4.2.2
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  { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
81 
82 
84  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
85  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
86  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
87  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
89  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
90  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
91  { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
92  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
93  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
94  { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
95  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
96  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
97  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
98  { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
99 #if FF_API_OLD_RTSP_OPTIONS
100  { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
101  { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
102 #else
103  { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
104 #endif
105  COMMON_OPTS(),
106  { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
107 #if FF_API_OLD_RTSP_OPTIONS
108  { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
109 #endif
110  { NULL },
111 };
112 
113 static const AVOption sdp_options[] = {
114  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
115  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
116  { "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" },
117  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
118  COMMON_OPTS(),
119  { NULL },
120 };
121 
122 static const AVOption rtp_options[] = {
123  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
124  COMMON_OPTS(),
125  { NULL },
126 };
127 
128 
130 {
132  char buf[256];
133 
134  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
135  av_dict_set(&opts, "buffer_size", buf, 0);
136  snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
137  av_dict_set(&opts, "pkt_size", buf, 0);
138 
139  return opts;
140 }
141 
142 static void get_word_until_chars(char *buf, int buf_size,
143  const char *sep, const char **pp)
144 {
145  const char *p;
146  char *q;
147 
148  p = *pp;
149  p += strspn(p, SPACE_CHARS);
150  q = buf;
151  while (!strchr(sep, *p) && *p != '\0') {
152  if ((q - buf) < buf_size - 1)
153  *q++ = *p;
154  p++;
155  }
156  if (buf_size > 0)
157  *q = '\0';
158  *pp = p;
159 }
160 
161 static void get_word_sep(char *buf, int buf_size, const char *sep,
162  const char **pp)
163 {
164  if (**pp == '/') (*pp)++;
165  get_word_until_chars(buf, buf_size, sep, pp);
166 }
167 
168 static void get_word(char *buf, int buf_size, const char **pp)
169 {
170  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
171 }
172 
173 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
174  * and end time.
175  * Used for seeking in the rtp stream.
176  */
177 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
178 {
179  char buf[256];
180 
181  p += strspn(p, SPACE_CHARS);
182  if (!av_stristart(p, "npt=", &p))
183  return;
184 
185  *start = AV_NOPTS_VALUE;
186  *end = AV_NOPTS_VALUE;
187 
188  get_word_sep(buf, sizeof(buf), "-", &p);
189  if (av_parse_time(start, buf, 1) < 0)
190  return;
191  if (*p == '-') {
192  p++;
193  get_word_sep(buf, sizeof(buf), "-", &p);
194  if (av_parse_time(end, buf, 1) < 0)
195  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
196  }
197 }
198 
200  const char *buf, struct sockaddr_storage *sock)
201 {
202  struct addrinfo hints = { 0 }, *ai = NULL;
203  int ret;
204 
205  hints.ai_flags = AI_NUMERICHOST;
206  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
207  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
208  buf,
209  gai_strerror(ret));
210  return -1;
211  }
212  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
213  freeaddrinfo(ai);
214  return 0;
215 }
216 
217 #if CONFIG_RTPDEC
218 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
219  RTSPStream *rtsp_st, AVStream *st)
220 {
221  AVCodecParameters *par = st ? st->codecpar : NULL;
222  if (!handler)
223  return;
224  if (par)
225  par->codec_id = handler->codec_id;
226  rtsp_st->dynamic_handler = handler;
227  if (st)
228  st->need_parsing = handler->need_parsing;
229  if (handler->priv_data_size) {
231  if (!rtsp_st->dynamic_protocol_context)
232  rtsp_st->dynamic_handler = NULL;
233  }
234 }
235 
236 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
237  AVStream *st)
238 {
239  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
240  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
241  rtsp_st->dynamic_protocol_context);
242  if (ret < 0) {
243  if (rtsp_st->dynamic_protocol_context) {
244  if (rtsp_st->dynamic_handler->close)
245  rtsp_st->dynamic_handler->close(
246  rtsp_st->dynamic_protocol_context);
248  }
249  rtsp_st->dynamic_protocol_context = NULL;
250  rtsp_st->dynamic_handler = NULL;
251  }
252  }
253 }
254 
255 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
256 static int sdp_parse_rtpmap(AVFormatContext *s,
257  AVStream *st, RTSPStream *rtsp_st,
258  int payload_type, const char *p)
259 {
260  AVCodecParameters *par = st->codecpar;
261  char buf[256];
262  int i;
263  const AVCodecDescriptor *desc;
264  const char *c_name;
265 
266  /* See if we can handle this kind of payload.
267  * The space should normally not be there but some Real streams or
268  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
269  * have a trailing space. */
270  get_word_sep(buf, sizeof(buf), "/ ", &p);
271  if (payload_type < RTP_PT_PRIVATE) {
272  /* We are in a standard case
273  * (from http://www.iana.org/assignments/rtp-parameters). */
274  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
275  }
276 
277  if (par->codec_id == AV_CODEC_ID_NONE) {
278  const RTPDynamicProtocolHandler *handler =
280  init_rtp_handler(handler, rtsp_st, st);
281  /* If no dynamic handler was found, check with the list of standard
282  * allocated types, if such a stream for some reason happens to
283  * use a private payload type. This isn't handled in rtpdec.c, since
284  * the format name from the rtpmap line never is passed into rtpdec. */
285  if (!rtsp_st->dynamic_handler)
286  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
287  }
288 
289  desc = avcodec_descriptor_get(par->codec_id);
290  if (desc && desc->name)
291  c_name = desc->name;
292  else
293  c_name = "(null)";
294 
295  get_word_sep(buf, sizeof(buf), "/", &p);
296  i = atoi(buf);
297  switch (par->codec_type) {
298  case AVMEDIA_TYPE_AUDIO:
299  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
302  if (i > 0) {
303  par->sample_rate = i;
304  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
305  get_word_sep(buf, sizeof(buf), "/", &p);
306  i = atoi(buf);
307  if (i > 0)
308  par->channels = i;
309  }
310  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
311  par->sample_rate);
312  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
313  par->channels);
314  break;
315  case AVMEDIA_TYPE_VIDEO:
316  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
317  if (i > 0)
318  avpriv_set_pts_info(st, 32, 1, i);
319  break;
320  default:
321  break;
322  }
323  finalize_rtp_handler_init(s, rtsp_st, st);
324  return 0;
325 }
326 
327 /* parse the attribute line from the fmtp a line of an sdp response. This
328  * is broken out as a function because it is used in rtp_h264.c, which is
329  * forthcoming. */
330 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
331  char *value, int value_size)
332 {
333  *p += strspn(*p, SPACE_CHARS);
334  if (**p) {
335  get_word_sep(attr, attr_size, "=", p);
336  if (**p == '=')
337  (*p)++;
338  get_word_sep(value, value_size, ";", p);
339  if (**p == ';')
340  (*p)++;
341  return 1;
342  }
343  return 0;
344 }
345 
346 typedef struct SDPParseState {
347  /* SDP only */
348  struct sockaddr_storage default_ip;
349  int default_ttl;
350  int skip_media; ///< set if an unknown m= line occurs
351  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
352  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
353  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
354  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
355  int seen_rtpmap;
356  int seen_fmtp;
357  char delayed_fmtp[2048];
358 } SDPParseState;
359 
360 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
361  struct RTSPSource ***dest, int *dest_count)
362 {
363  RTSPSource *rtsp_src, *rtsp_src2;
364  int i;
365  for (i = 0; i < count; i++) {
366  rtsp_src = addrs[i];
367  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
368  if (!rtsp_src2)
369  continue;
370  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
371  dynarray_add(dest, dest_count, rtsp_src2);
372  }
373 }
374 
375 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
376  int payload_type, const char *line)
377 {
378  int i;
379 
380  for (i = 0; i < rt->nb_rtsp_streams; i++) {
381  RTSPStream *rtsp_st = rt->rtsp_streams[i];
382  if (rtsp_st->sdp_payload_type == payload_type &&
383  rtsp_st->dynamic_handler &&
384  rtsp_st->dynamic_handler->parse_sdp_a_line) {
385  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
386  rtsp_st->dynamic_protocol_context, line);
387  }
388  }
389 }
390 
391 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
392  int letter, const char *buf)
393 {
394  RTSPState *rt = s->priv_data;
395  char buf1[64], st_type[64];
396  const char *p;
397  enum AVMediaType codec_type;
398  int payload_type;
399  AVStream *st;
400  RTSPStream *rtsp_st;
401  RTSPSource *rtsp_src;
402  struct sockaddr_storage sdp_ip;
403  int ttl;
404 
405  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
406 
407  p = buf;
408  if (s1->skip_media && letter != 'm')
409  return;
410  switch (letter) {
411  case 'c':
412  get_word(buf1, sizeof(buf1), &p);
413  if (strcmp(buf1, "IN") != 0)
414  return;
415  get_word(buf1, sizeof(buf1), &p);
416  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
417  return;
418  get_word_sep(buf1, sizeof(buf1), "/", &p);
419  if (get_sockaddr(s, buf1, &sdp_ip))
420  return;
421  ttl = 16;
422  if (*p == '/') {
423  p++;
424  get_word_sep(buf1, sizeof(buf1), "/", &p);
425  ttl = atoi(buf1);
426  }
427  if (s->nb_streams == 0) {
428  s1->default_ip = sdp_ip;
429  s1->default_ttl = ttl;
430  } else {
431  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
432  rtsp_st->sdp_ip = sdp_ip;
433  rtsp_st->sdp_ttl = ttl;
434  }
435  break;
436  case 's':
437  av_dict_set(&s->metadata, "title", p, 0);
438  break;
439  case 'i':
440  if (s->nb_streams == 0) {
441  av_dict_set(&s->metadata, "comment", p, 0);
442  break;
443  }
444  break;
445  case 'm':
446  /* new stream */
447  s1->skip_media = 0;
448  s1->seen_fmtp = 0;
449  s1->seen_rtpmap = 0;
450  codec_type = AVMEDIA_TYPE_UNKNOWN;
451  get_word(st_type, sizeof(st_type), &p);
452  if (!strcmp(st_type, "audio")) {
453  codec_type = AVMEDIA_TYPE_AUDIO;
454  } else if (!strcmp(st_type, "video")) {
455  codec_type = AVMEDIA_TYPE_VIDEO;
456  } else if (!strcmp(st_type, "application")) {
457  codec_type = AVMEDIA_TYPE_DATA;
458  } else if (!strcmp(st_type, "text")) {
459  codec_type = AVMEDIA_TYPE_SUBTITLE;
460  }
461  if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
462  !(rt->media_type_mask & (1 << codec_type)) ||
463  rt->nb_rtsp_streams >= s->max_streams
464  ) {
465  s1->skip_media = 1;
466  return;
467  }
468  rtsp_st = av_mallocz(sizeof(RTSPStream));
469  if (!rtsp_st)
470  return;
471  rtsp_st->stream_index = -1;
472  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
473 
474  rtsp_st->sdp_ip = s1->default_ip;
475  rtsp_st->sdp_ttl = s1->default_ttl;
476 
477  copy_default_source_addrs(s1->default_include_source_addrs,
478  s1->nb_default_include_source_addrs,
479  &rtsp_st->include_source_addrs,
480  &rtsp_st->nb_include_source_addrs);
481  copy_default_source_addrs(s1->default_exclude_source_addrs,
482  s1->nb_default_exclude_source_addrs,
483  &rtsp_st->exclude_source_addrs,
484  &rtsp_st->nb_exclude_source_addrs);
485 
486  get_word(buf1, sizeof(buf1), &p); /* port */
487  rtsp_st->sdp_port = atoi(buf1);
488 
489  get_word(buf1, sizeof(buf1), &p); /* protocol */
490  if (!strcmp(buf1, "udp"))
492  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
493  rtsp_st->feedback = 1;
494 
495  /* XXX: handle list of formats */
496  get_word(buf1, sizeof(buf1), &p); /* format list */
497  rtsp_st->sdp_payload_type = atoi(buf1);
498 
499  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
500  /* no corresponding stream */
501  if (rt->transport == RTSP_TRANSPORT_RAW) {
502  if (CONFIG_RTPDEC && !rt->ts)
503  rt->ts = avpriv_mpegts_parse_open(s);
504  } else {
506  handler = ff_rtp_handler_find_by_id(
508  init_rtp_handler(handler, rtsp_st, NULL);
509  finalize_rtp_handler_init(s, rtsp_st, NULL);
510  }
511  } else if (rt->server_type == RTSP_SERVER_WMS &&
512  codec_type == AVMEDIA_TYPE_DATA) {
513  /* RTX stream, a stream that carries all the other actual
514  * audio/video streams. Don't expose this to the callers. */
515  } else {
516  st = avformat_new_stream(s, NULL);
517  if (!st)
518  return;
519  st->id = rt->nb_rtsp_streams - 1;
520  rtsp_st->stream_index = st->index;
522  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
524  /* if standard payload type, we can find the codec right now */
526  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
527  st->codecpar->sample_rate > 0)
528  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
529  /* Even static payload types may need a custom depacketizer */
530  handler = ff_rtp_handler_find_by_id(
531  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
532  init_rtp_handler(handler, rtsp_st, st);
533  finalize_rtp_handler_init(s, rtsp_st, st);
534  }
535  if (rt->default_lang[0])
536  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
537  }
538  /* put a default control url */
539  av_strlcpy(rtsp_st->control_url, rt->control_uri,
540  sizeof(rtsp_st->control_url));
541  break;
542  case 'a':
543  if (av_strstart(p, "control:", &p)) {
544  if (s->nb_streams == 0) {
545  if (!strncmp(p, "rtsp://", 7))
546  av_strlcpy(rt->control_uri, p,
547  sizeof(rt->control_uri));
548  } else {
549  char proto[32];
550  /* get the control url */
551  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
552 
553  /* XXX: may need to add full url resolution */
554  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
555  NULL, NULL, 0, p);
556  if (proto[0] == '\0') {
557  /* relative control URL */
558  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
559  av_strlcat(rtsp_st->control_url, "/",
560  sizeof(rtsp_st->control_url));
561  av_strlcat(rtsp_st->control_url, p,
562  sizeof(rtsp_st->control_url));
563  } else
564  av_strlcpy(rtsp_st->control_url, p,
565  sizeof(rtsp_st->control_url));
566  }
567  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
568  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
569  get_word(buf1, sizeof(buf1), &p);
570  payload_type = atoi(buf1);
571  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
572  if (rtsp_st->stream_index >= 0) {
573  st = s->streams[rtsp_st->stream_index];
574  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
575  }
576  s1->seen_rtpmap = 1;
577  if (s1->seen_fmtp) {
578  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
579  }
580  } else if (av_strstart(p, "fmtp:", &p) ||
581  av_strstart(p, "framesize:", &p)) {
582  // let dynamic protocol handlers have a stab at the line.
583  get_word(buf1, sizeof(buf1), &p);
584  payload_type = atoi(buf1);
585  if (s1->seen_rtpmap) {
586  parse_fmtp(s, rt, payload_type, buf);
587  } else {
588  s1->seen_fmtp = 1;
589  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
590  }
591  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
592  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593  get_word(buf1, sizeof(buf1), &p);
594  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
595  } else if (av_strstart(p, "range:", &p)) {
596  int64_t start, end;
597 
598  // this is so that seeking on a streamed file can work.
599  rtsp_parse_range_npt(p, &start, &end);
600  s->start_time = start;
601  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
602  s->duration = (end == AV_NOPTS_VALUE) ?
603  AV_NOPTS_VALUE : end - start;
604  } else if (av_strstart(p, "lang:", &p)) {
605  if (s->nb_streams > 0) {
606  get_word(buf1, sizeof(buf1), &p);
607  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
608  if (rtsp_st->stream_index >= 0) {
609  st = s->streams[rtsp_st->stream_index];
610  av_dict_set(&st->metadata, "language", buf1, 0);
611  }
612  } else
613  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
614  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
615  if (atoi(p) == 1)
617  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
618  s->nb_streams > 0) {
619  st = s->streams[s->nb_streams - 1];
620  st->codecpar->sample_rate = atoi(p);
621  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
622  // RFC 4568
623  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
624  get_word(buf1, sizeof(buf1), &p); // ignore tag
625  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
626  p += strspn(p, SPACE_CHARS);
627  if (av_strstart(p, "inline:", &p))
628  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
629  } else if (av_strstart(p, "source-filter:", &p)) {
630  int exclude = 0;
631  get_word(buf1, sizeof(buf1), &p);
632  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
633  return;
634  exclude = !strcmp(buf1, "excl");
635 
636  get_word(buf1, sizeof(buf1), &p);
637  if (strcmp(buf1, "IN") != 0)
638  return;
639  get_word(buf1, sizeof(buf1), &p);
640  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
641  return;
642  // not checking that the destination address actually matches or is wildcard
643  get_word(buf1, sizeof(buf1), &p);
644 
645  while (*p != '\0') {
646  rtsp_src = av_mallocz(sizeof(*rtsp_src));
647  if (!rtsp_src)
648  return;
649  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
650  if (exclude) {
651  if (s->nb_streams == 0) {
652  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
653  } else {
654  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
655  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
656  }
657  } else {
658  if (s->nb_streams == 0) {
659  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
660  } else {
661  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
662  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
663  }
664  }
665  }
666  } else {
667  if (rt->server_type == RTSP_SERVER_WMS)
669  if (s->nb_streams > 0) {
670  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
671 
672  if (rt->server_type == RTSP_SERVER_REAL)
673  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
674 
675  if (rtsp_st->dynamic_handler &&
677  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
678  rtsp_st->stream_index,
679  rtsp_st->dynamic_protocol_context, buf);
680  }
681  }
682  break;
683  }
684 }
685 
686 int ff_sdp_parse(AVFormatContext *s, const char *content)
687 {
688  const char *p;
689  int letter, i;
690  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
691  * contain long SDP lines containing complete ASF Headers (several
692  * kB) or arrays of MDPR (RM stream descriptor) headers plus
693  * "rulebooks" describing their properties. Therefore, the SDP line
694  * buffer is large.
695  *
696  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
697  * in rtpdec_xiph.c. */
698  char buf[16384], *q;
699  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
700 
701  p = content;
702  for (;;) {
703  p += strspn(p, SPACE_CHARS);
704  letter = *p;
705  if (letter == '\0')
706  break;
707  p++;
708  if (*p != '=')
709  goto next_line;
710  p++;
711  /* get the content */
712  q = buf;
713  while (*p != '\n' && *p != '\r' && *p != '\0') {
714  if ((q - buf) < sizeof(buf) - 1)
715  *q++ = *p;
716  p++;
717  }
718  *q = '\0';
719  sdp_parse_line(s, s1, letter, buf);
720  next_line:
721  while (*p != '\n' && *p != '\0')
722  p++;
723  if (*p == '\n')
724  p++;
725  }
726 
727  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
728  av_freep(&s1->default_include_source_addrs[i]);
729  av_freep(&s1->default_include_source_addrs);
730  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
731  av_freep(&s1->default_exclude_source_addrs[i]);
732  av_freep(&s1->default_exclude_source_addrs);
733 
734  return 0;
735 }
736 #endif /* CONFIG_RTPDEC */
737 
738 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
739 {
740  RTSPState *rt = s->priv_data;
741  int i;
742 
743  for (i = 0; i < rt->nb_rtsp_streams; i++) {
744  RTSPStream *rtsp_st = rt->rtsp_streams[i];
745  if (!rtsp_st)
746  continue;
747  if (rtsp_st->transport_priv) {
748  if (s->oformat) {
749  AVFormatContext *rtpctx = rtsp_st->transport_priv;
750  av_write_trailer(rtpctx);
752  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
753  ff_rtsp_tcp_write_packet(s, rtsp_st);
754  ffio_free_dyn_buf(&rtpctx->pb);
755  } else {
756  avio_closep(&rtpctx->pb);
757  }
758  avformat_free_context(rtpctx);
759  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
761  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
763  }
764  rtsp_st->transport_priv = NULL;
765  if (rtsp_st->rtp_handle)
766  ffurl_close(rtsp_st->rtp_handle);
767  rtsp_st->rtp_handle = NULL;
768  }
769 }
770 
771 /* close and free RTSP streams */
773 {
774  RTSPState *rt = s->priv_data;
775  int i, j;
776  RTSPStream *rtsp_st;
777 
778  ff_rtsp_undo_setup(s, 0);
779  for (i = 0; i < rt->nb_rtsp_streams; i++) {
780  rtsp_st = rt->rtsp_streams[i];
781  if (rtsp_st) {
782  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
783  if (rtsp_st->dynamic_handler->close)
784  rtsp_st->dynamic_handler->close(
785  rtsp_st->dynamic_protocol_context);
787  }
788  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
789  av_freep(&rtsp_st->include_source_addrs[j]);
790  av_freep(&rtsp_st->include_source_addrs);
791  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
792  av_freep(&rtsp_st->exclude_source_addrs[j]);
793  av_freep(&rtsp_st->exclude_source_addrs);
794 
795  av_freep(&rtsp_st);
796  }
797  }
798  av_freep(&rt->rtsp_streams);
799  if (rt->asf_ctx) {
801  }
802  if (CONFIG_RTPDEC && rt->ts)
804  av_freep(&rt->p);
805  av_freep(&rt->recvbuf);
806 }
807 
809 {
810  RTSPState *rt = s->priv_data;
811  AVStream *st = NULL;
812  int reordering_queue_size = rt->reordering_queue_size;
813  if (reordering_queue_size < 0) {
815  reordering_queue_size = 0;
816  else
817  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
818  }
819 
820  /* open the RTP context */
821  if (rtsp_st->stream_index >= 0)
822  st = s->streams[rtsp_st->stream_index];
823  if (!st)
825 
826  if (CONFIG_RTSP_MUXER && s->oformat && st) {
827  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
828  s, st, rtsp_st->rtp_handle,
830  rtsp_st->stream_index);
831  /* Ownership of rtp_handle is passed to the rtp mux context */
832  rtsp_st->rtp_handle = NULL;
833  if (ret < 0)
834  return ret;
835  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
836  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
837  return 0; // Don't need to open any parser here
838  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
839  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
840  rtsp_st->dynamic_protocol_context,
841  rtsp_st->dynamic_handler);
842  else if (CONFIG_RTPDEC)
843  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
844  rtsp_st->sdp_payload_type,
845  reordering_queue_size);
846 
847  if (!rtsp_st->transport_priv) {
848  return AVERROR(ENOMEM);
849  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
850  s->iformat) {
851  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
852  rtpctx->ssrc = rtsp_st->ssrc;
853  if (rtsp_st->dynamic_handler) {
855  rtsp_st->dynamic_protocol_context,
856  rtsp_st->dynamic_handler);
857  }
858  if (rtsp_st->crypto_suite[0])
860  rtsp_st->crypto_suite,
861  rtsp_st->crypto_params);
862  }
863 
864  return 0;
865 }
866 
867 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
868 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
869 {
870  const char *q;
871  char *p;
872  int v;
873 
874  q = *pp;
875  q += strspn(q, SPACE_CHARS);
876  v = strtol(q, &p, 10);
877  if (*p == '-') {
878  p++;
879  *min_ptr = v;
880  v = strtol(p, &p, 10);
881  *max_ptr = v;
882  } else {
883  *min_ptr = v;
884  *max_ptr = v;
885  }
886  *pp = p;
887 }
888 
889 /* XXX: only one transport specification is parsed */
890 static void rtsp_parse_transport(AVFormatContext *s,
891  RTSPMessageHeader *reply, const char *p)
892 {
893  char transport_protocol[16];
894  char profile[16];
895  char lower_transport[16];
896  char parameter[16];
898  char buf[256];
899 
900  reply->nb_transports = 0;
901 
902  for (;;) {
903  p += strspn(p, SPACE_CHARS);
904  if (*p == '\0')
905  break;
906 
907  th = &reply->transports[reply->nb_transports];
908 
909  get_word_sep(transport_protocol, sizeof(transport_protocol),
910  "/", &p);
911  if (!av_strcasecmp (transport_protocol, "rtp")) {
912  get_word_sep(profile, sizeof(profile), "/;,", &p);
913  lower_transport[0] = '\0';
914  /* rtp/avp/<protocol> */
915  if (*p == '/') {
916  get_word_sep(lower_transport, sizeof(lower_transport),
917  ";,", &p);
918  }
920  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
921  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
922  /* x-pn-tng/<protocol> */
923  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
924  profile[0] = '\0';
926  } else if (!av_strcasecmp(transport_protocol, "raw")) {
927  get_word_sep(profile, sizeof(profile), "/;,", &p);
928  lower_transport[0] = '\0';
929  /* raw/raw/<protocol> */
930  if (*p == '/') {
931  get_word_sep(lower_transport, sizeof(lower_transport),
932  ";,", &p);
933  }
935  }
936  if (!av_strcasecmp(lower_transport, "TCP"))
938  else
940 
941  if (*p == ';')
942  p++;
943  /* get each parameter */
944  while (*p != '\0' && *p != ',') {
945  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
946  if (!strcmp(parameter, "port")) {
947  if (*p == '=') {
948  p++;
949  rtsp_parse_range(&th->port_min, &th->port_max, &p);
950  }
951  } else if (!strcmp(parameter, "client_port")) {
952  if (*p == '=') {
953  p++;
954  rtsp_parse_range(&th->client_port_min,
955  &th->client_port_max, &p);
956  }
957  } else if (!strcmp(parameter, "server_port")) {
958  if (*p == '=') {
959  p++;
960  rtsp_parse_range(&th->server_port_min,
961  &th->server_port_max, &p);
962  }
963  } else if (!strcmp(parameter, "interleaved")) {
964  if (*p == '=') {
965  p++;
966  rtsp_parse_range(&th->interleaved_min,
967  &th->interleaved_max, &p);
968  }
969  } else if (!strcmp(parameter, "multicast")) {
972  } else if (!strcmp(parameter, "ttl")) {
973  if (*p == '=') {
974  char *end;
975  p++;
976  th->ttl = strtol(p, &end, 10);
977  p = end;
978  }
979  } else if (!strcmp(parameter, "destination")) {
980  if (*p == '=') {
981  p++;
982  get_word_sep(buf, sizeof(buf), ";,", &p);
983  get_sockaddr(s, buf, &th->destination);
984  }
985  } else if (!strcmp(parameter, "source")) {
986  if (*p == '=') {
987  p++;
988  get_word_sep(buf, sizeof(buf), ";,", &p);
989  av_strlcpy(th->source, buf, sizeof(th->source));
990  }
991  } else if (!strcmp(parameter, "mode")) {
992  if (*p == '=') {
993  p++;
994  get_word_sep(buf, sizeof(buf), ";, ", &p);
995  if (!strcmp(buf, "record") ||
996  !strcmp(buf, "receive"))
997  th->mode_record = 1;
998  }
999  }
1000 
1001  while (*p != ';' && *p != '\0' && *p != ',')
1002  p++;
1003  if (*p == ';')
1004  p++;
1005  }
1006  if (*p == ',')
1007  p++;
1008 
1009  reply->nb_transports++;
1010  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1011  break;
1012  }
1013 }
1014 
1015 static void handle_rtp_info(RTSPState *rt, const char *url,
1016  uint32_t seq, uint32_t rtptime)
1017 {
1018  int i;
1019  if (!rtptime || !url[0])
1020  return;
1021  if (rt->transport != RTSP_TRANSPORT_RTP)
1022  return;
1023  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1024  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1025  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1026  if (!rtpctx)
1027  continue;
1028  if (!strcmp(rtsp_st->control_url, url)) {
1029  rtpctx->base_timestamp = rtptime;
1030  break;
1031  }
1032  }
1033 }
1034 
1035 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1036 {
1037  int read = 0;
1038  char key[20], value[1024], url[1024] = "";
1039  uint32_t seq = 0, rtptime = 0;
1040 
1041  for (;;) {
1042  p += strspn(p, SPACE_CHARS);
1043  if (!*p)
1044  break;
1045  get_word_sep(key, sizeof(key), "=", &p);
1046  if (*p != '=')
1047  break;
1048  p++;
1049  get_word_sep(value, sizeof(value), ";, ", &p);
1050  read++;
1051  if (!strcmp(key, "url"))
1052  av_strlcpy(url, value, sizeof(url));
1053  else if (!strcmp(key, "seq"))
1054  seq = strtoul(value, NULL, 10);
1055  else if (!strcmp(key, "rtptime"))
1056  rtptime = strtoul(value, NULL, 10);
1057  if (*p == ',') {
1058  handle_rtp_info(rt, url, seq, rtptime);
1059  url[0] = '\0';
1060  seq = rtptime = 0;
1061  read = 0;
1062  }
1063  if (*p)
1064  p++;
1065  }
1066  if (read > 0)
1067  handle_rtp_info(rt, url, seq, rtptime);
1068 }
1069 
1071  RTSPMessageHeader *reply, const char *buf,
1072  RTSPState *rt, const char *method)
1073 {
1074  const char *p;
1075 
1076  /* NOTE: we do case independent match for broken servers */
1077  p = buf;
1078  if (av_stristart(p, "Session:", &p)) {
1079  int t;
1080  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1081  if (av_stristart(p, ";timeout=", &p) &&
1082  (t = strtol(p, NULL, 10)) > 0) {
1083  reply->timeout = t;
1084  }
1085  } else if (av_stristart(p, "Content-Length:", &p)) {
1086  reply->content_length = strtol(p, NULL, 10);
1087  } else if (av_stristart(p, "Transport:", &p)) {
1088  rtsp_parse_transport(s, reply, p);
1089  } else if (av_stristart(p, "CSeq:", &p)) {
1090  reply->seq = strtol(p, NULL, 10);
1091  } else if (av_stristart(p, "Range:", &p)) {
1092  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1093  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1094  p += strspn(p, SPACE_CHARS);
1095  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1096  } else if (av_stristart(p, "Server:", &p)) {
1097  p += strspn(p, SPACE_CHARS);
1098  av_strlcpy(reply->server, p, sizeof(reply->server));
1099  } else if (av_stristart(p, "Notice:", &p) ||
1100  av_stristart(p, "X-Notice:", &p)) {
1101  reply->notice = strtol(p, NULL, 10);
1102  } else if (av_stristart(p, "Location:", &p)) {
1103  p += strspn(p, SPACE_CHARS);
1104  av_strlcpy(reply->location, p , sizeof(reply->location));
1105  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1106  p += strspn(p, SPACE_CHARS);
1107  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1108  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1109  p += strspn(p, SPACE_CHARS);
1110  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1111  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1112  p += strspn(p, SPACE_CHARS);
1113  if (method && !strcmp(method, "DESCRIBE"))
1114  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1115  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1116  p += strspn(p, SPACE_CHARS);
1117  if (method && !strcmp(method, "PLAY"))
1118  rtsp_parse_rtp_info(rt, p);
1119  } else if (av_stristart(p, "Public:", &p) && rt) {
1120  if (strstr(p, "GET_PARAMETER") &&
1121  method && !strcmp(method, "OPTIONS"))
1122  rt->get_parameter_supported = 1;
1123  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1124  p += strspn(p, SPACE_CHARS);
1125  rt->accept_dynamic_rate = atoi(p);
1126  } else if (av_stristart(p, "Content-Type:", &p)) {
1127  p += strspn(p, SPACE_CHARS);
1128  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1129  }
1130 }
1131 
1132 /* skip a RTP/TCP interleaved packet */
1134 {
1135  RTSPState *rt = s->priv_data;
1136  int ret, len, len1;
1137  uint8_t buf[1024];
1138 
1139  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1140  if (ret != 3)
1141  return;
1142  len = AV_RB16(buf + 1);
1143 
1144  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1145 
1146  /* skip payload */
1147  while (len > 0) {
1148  len1 = len;
1149  if (len1 > sizeof(buf))
1150  len1 = sizeof(buf);
1151  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1152  if (ret != len1)
1153  return;
1154  len -= len1;
1155  }
1156 }
1157 
1159  unsigned char **content_ptr,
1160  int return_on_interleaved_data, const char *method)
1161 {
1162  RTSPState *rt = s->priv_data;
1163  char buf[4096], buf1[1024], *q;
1164  unsigned char ch;
1165  const char *p;
1166  int ret, content_length, line_count = 0, request = 0;
1167  unsigned char *content = NULL;
1168 
1169 start:
1170  line_count = 0;
1171  request = 0;
1172  content = NULL;
1173  memset(reply, 0, sizeof(*reply));
1174 
1175  /* parse reply (XXX: use buffers) */
1176  rt->last_reply[0] = '\0';
1177  for (;;) {
1178  q = buf;
1179  for (;;) {
1180  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1181  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1182  if (ret != 1)
1183  return AVERROR_EOF;
1184  if (ch == '\n')
1185  break;
1186  if (ch == '$' && q == buf) {
1187  if (return_on_interleaved_data) {
1188  return 1;
1189  } else
1191  } else if (ch != '\r') {
1192  if ((q - buf) < sizeof(buf) - 1)
1193  *q++ = ch;
1194  }
1195  }
1196  *q = '\0';
1197 
1198  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1199 
1200  /* test if last line */
1201  if (buf[0] == '\0')
1202  break;
1203  p = buf;
1204  if (line_count == 0) {
1205  /* get reply code */
1206  get_word(buf1, sizeof(buf1), &p);
1207  if (!strncmp(buf1, "RTSP/", 5)) {
1208  get_word(buf1, sizeof(buf1), &p);
1209  reply->status_code = atoi(buf1);
1210  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1211  } else {
1212  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1213  get_word(buf1, sizeof(buf1), &p); // object
1214  request = 1;
1215  }
1216  } else {
1217  ff_rtsp_parse_line(s, reply, p, rt, method);
1218  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1219  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1220  }
1221  line_count++;
1222  }
1223 
1224  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1225  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1226 
1227  content_length = reply->content_length;
1228  if (content_length > 0) {
1229  /* leave some room for a trailing '\0' (useful for simple parsing) */
1230  content = av_malloc(content_length + 1);
1231  if (!content)
1232  return AVERROR(ENOMEM);
1233  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1234  content[content_length] = '\0';
1235  }
1236  if (content_ptr)
1237  *content_ptr = content;
1238  else
1239  av_freep(&content);
1240 
1241  if (request) {
1242  char buf[1024];
1243  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1244  const char* ptr = buf;
1245 
1246  if (!strcmp(reply->reason, "OPTIONS")) {
1247  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1248  if (reply->seq)
1249  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1250  if (reply->session_id[0])
1251  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1252  reply->session_id);
1253  } else {
1254  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1255  }
1256  av_strlcat(buf, "\r\n", sizeof(buf));
1257 
1258  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1259  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1260  ptr = base64buf;
1261  }
1262  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1263 
1265  /* Even if the request from the server had data, it is not the data
1266  * that the caller wants or expects. The memory could also be leaked
1267  * if the actual following reply has content data. */
1268  if (content_ptr)
1269  av_freep(content_ptr);
1270  /* If method is set, this is called from ff_rtsp_send_cmd,
1271  * where a reply to exactly this request is awaited. For
1272  * callers from within packet receiving, we just want to
1273  * return to the caller and go back to receiving packets. */
1274  if (method)
1275  goto start;
1276  return 0;
1277  }
1278 
1279  if (rt->seq != reply->seq) {
1280  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1281  rt->seq, reply->seq);
1282  }
1283 
1284  /* EOS */
1285  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1286  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1287  reply->notice == 2306 /* Continuous Feed Terminated */) {
1288  rt->state = RTSP_STATE_IDLE;
1289  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1290  return AVERROR(EIO); /* data or server error */
1291  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1292  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1293  return AVERROR(EPERM);
1294 
1295  return 0;
1296 }
1297 
1298 /**
1299  * Send a command to the RTSP server without waiting for the reply.
1300  *
1301  * @param s RTSP (de)muxer context
1302  * @param method the method for the request
1303  * @param url the target url for the request
1304  * @param headers extra header lines to include in the request
1305  * @param send_content if non-null, the data to send as request body content
1306  * @param send_content_length the length of the send_content data, or 0 if
1307  * send_content is null
1308  *
1309  * @return zero if success, nonzero otherwise
1310  */
1311 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1312  const char *method, const char *url,
1313  const char *headers,
1314  const unsigned char *send_content,
1315  int send_content_length)
1316 {
1317  RTSPState *rt = s->priv_data;
1318  char buf[4096], *out_buf;
1319  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1320 
1321  if (!rt->rtsp_hd_out)
1322  return ENOTCONN;
1323 
1324  /* Add in RTSP headers */
1325  out_buf = buf;
1326  rt->seq++;
1327  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1328  if (headers)
1329  av_strlcat(buf, headers, sizeof(buf));
1330  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1331  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1332  if (rt->session_id[0] != '\0' && (!headers ||
1333  !strstr(headers, "\nIf-Match:"))) {
1334  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1335  }
1336  if (rt->auth[0]) {
1337  char *str = ff_http_auth_create_response(&rt->auth_state,
1338  rt->auth, url, method);
1339  if (str)
1340  av_strlcat(buf, str, sizeof(buf));
1341  av_free(str);
1342  }
1343  if (send_content_length > 0 && send_content)
1344  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1345  av_strlcat(buf, "\r\n", sizeof(buf));
1346 
1347  /* base64 encode rtsp if tunneling */
1348  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1349  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1350  out_buf = base64buf;
1351  }
1352 
1353  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1354 
1355  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1356  if (send_content_length > 0 && send_content) {
1357  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1358  avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1359  return AVERROR_PATCHWELCOME;
1360  }
1361  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1362  }
1364 
1365  return 0;
1366 }
1367 
1368 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1369  const char *url, const char *headers)
1370 {
1371  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1372 }
1373 
1374 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1375  const char *headers, RTSPMessageHeader *reply,
1376  unsigned char **content_ptr)
1377 {
1378  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1379  content_ptr, NULL, 0);
1380 }
1381 
1383  const char *method, const char *url,
1384  const char *header,
1385  RTSPMessageHeader *reply,
1386  unsigned char **content_ptr,
1387  const unsigned char *send_content,
1388  int send_content_length)
1389 {
1390  RTSPState *rt = s->priv_data;
1391  HTTPAuthType cur_auth_type;
1392  int ret, attempts = 0;
1393 
1394 retry:
1395  cur_auth_type = rt->auth_state.auth_type;
1396  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1397  send_content,
1398  send_content_length)))
1399  return ret;
1400 
1401  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1402  return ret;
1403  attempts++;
1404 
1405  if (reply->status_code == 401 &&
1406  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1407  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1408  goto retry;
1409 
1410  if (reply->status_code > 400){
1411  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1412  method,
1413  reply->status_code,
1414  reply->reason);
1415  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1416  }
1417 
1418  return 0;
1419 }
1420 
1421 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1422  int lower_transport, const char *real_challenge)
1423 {
1424  RTSPState *rt = s->priv_data;
1425  int rtx = 0, j, i, err, interleave = 0, port_off;
1426  RTSPStream *rtsp_st;
1427  RTSPMessageHeader reply1, *reply = &reply1;
1428  char cmd[2048];
1429  const char *trans_pref;
1430 
1431  if (rt->transport == RTSP_TRANSPORT_RDT)
1432  trans_pref = "x-pn-tng";
1433  else if (rt->transport == RTSP_TRANSPORT_RAW)
1434  trans_pref = "RAW/RAW";
1435  else
1436  trans_pref = "RTP/AVP";
1437 
1438  /* default timeout: 1 minute */
1439  rt->timeout = 60;
1440 
1441  /* Choose a random starting offset within the first half of the
1442  * port range, to allow for a number of ports to try even if the offset
1443  * happens to be at the end of the random range. */
1444  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1445  /* even random offset */
1446  port_off -= port_off & 0x01;
1447 
1448  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1449  char transport[2048];
1450 
1451  /*
1452  * WMS serves all UDP data over a single connection, the RTX, which
1453  * isn't necessarily the first in the SDP but has to be the first
1454  * to be set up, else the second/third SETUP will fail with a 461.
1455  */
1456  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1457  rt->server_type == RTSP_SERVER_WMS) {
1458  if (i == 0) {
1459  /* rtx first */
1460  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1461  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1462  if (len >= 4 &&
1463  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1464  "/rtx"))
1465  break;
1466  }
1467  if (rtx == rt->nb_rtsp_streams)
1468  return -1; /* no RTX found */
1469  rtsp_st = rt->rtsp_streams[rtx];
1470  } else
1471  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1472  } else
1473  rtsp_st = rt->rtsp_streams[i];
1474 
1475  /* RTP/UDP */
1476  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1477  char buf[256];
1478 
1479  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1480  port = reply->transports[0].client_port_min;
1481  goto have_port;
1482  }
1483 
1484  /* first try in specified port range */
1485  while (j <= rt->rtp_port_max) {
1486  AVDictionary *opts = map_to_opts(rt);
1487 
1488  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1489  "?localport=%d", j);
1490  /* we will use two ports per rtp stream (rtp and rtcp) */
1491  j += 2;
1494 
1495  av_dict_free(&opts);
1496 
1497  if (!err)
1498  goto rtp_opened;
1499  }
1500  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1501  err = AVERROR(EIO);
1502  goto fail;
1503 
1504  rtp_opened:
1505  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1506  have_port:
1507  snprintf(transport, sizeof(transport) - 1,
1508  "%s/UDP;", trans_pref);
1509  if (rt->server_type != RTSP_SERVER_REAL)
1510  av_strlcat(transport, "unicast;", sizeof(transport));
1511  av_strlcatf(transport, sizeof(transport),
1512  "client_port=%d", port);
1513  if (rt->transport == RTSP_TRANSPORT_RTP &&
1514  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1515  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1516  }
1517 
1518  /* RTP/TCP */
1519  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1520  /* For WMS streams, the application streams are only used for
1521  * UDP. When trying to set it up for TCP streams, the server
1522  * will return an error. Therefore, we skip those streams. */
1523  if (rt->server_type == RTSP_SERVER_WMS &&
1524  (rtsp_st->stream_index < 0 ||
1525  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1527  continue;
1528  snprintf(transport, sizeof(transport) - 1,
1529  "%s/TCP;", trans_pref);
1530  if (rt->transport != RTSP_TRANSPORT_RDT)
1531  av_strlcat(transport, "unicast;", sizeof(transport));
1532  av_strlcatf(transport, sizeof(transport),
1533  "interleaved=%d-%d",
1534  interleave, interleave + 1);
1535  interleave += 2;
1536  }
1537 
1538  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1539  snprintf(transport, sizeof(transport) - 1,
1540  "%s/UDP;multicast", trans_pref);
1541  }
1542  if (s->oformat) {
1543  av_strlcat(transport, ";mode=record", sizeof(transport));
1544  } else if (rt->server_type == RTSP_SERVER_REAL ||
1546  av_strlcat(transport, ";mode=play", sizeof(transport));
1547  snprintf(cmd, sizeof(cmd),
1548  "Transport: %s\r\n",
1549  transport);
1550  if (rt->accept_dynamic_rate)
1551  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1552  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1553  char real_res[41], real_csum[9];
1554  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1555  real_challenge);
1556  av_strlcatf(cmd, sizeof(cmd),
1557  "If-Match: %s\r\n"
1558  "RealChallenge2: %s, sd=%s\r\n",
1559  rt->session_id, real_res, real_csum);
1560  }
1561  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1562  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1563  err = 1;
1564  goto fail;
1565  } else if (reply->status_code != RTSP_STATUS_OK ||
1566  reply->nb_transports != 1) {
1568  goto fail;
1569  }
1570 
1571  /* XXX: same protocol for all streams is required */
1572  if (i > 0) {
1573  if (reply->transports[0].lower_transport != rt->lower_transport ||
1574  reply->transports[0].transport != rt->transport) {
1575  err = AVERROR_INVALIDDATA;
1576  goto fail;
1577  }
1578  } else {
1579  rt->lower_transport = reply->transports[0].lower_transport;
1580  rt->transport = reply->transports[0].transport;
1581  }
1582 
1583  /* Fail if the server responded with another lower transport mode
1584  * than what we requested. */
1585  if (reply->transports[0].lower_transport != lower_transport) {
1586  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1587  err = AVERROR_INVALIDDATA;
1588  goto fail;
1589  }
1590 
1591  switch(reply->transports[0].lower_transport) {
1593  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1594  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1595  break;
1596 
1597  case RTSP_LOWER_TRANSPORT_UDP: {
1598  char url[1024], options[30] = "";
1599  const char *peer = host;
1600 
1601  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1602  av_strlcpy(options, "?connect=1", sizeof(options));
1603  /* Use source address if specified */
1604  if (reply->transports[0].source[0])
1605  peer = reply->transports[0].source;
1606  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1607  reply->transports[0].server_port_min, "%s", options);
1608  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1609  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1610  err = AVERROR_INVALIDDATA;
1611  goto fail;
1612  }
1613  break;
1614  }
1616  char url[1024], namebuf[50], optbuf[20] = "";
1617  struct sockaddr_storage addr;
1618  int port, ttl;
1619 
1620  if (reply->transports[0].destination.ss_family) {
1621  addr = reply->transports[0].destination;
1622  port = reply->transports[0].port_min;
1623  ttl = reply->transports[0].ttl;
1624  } else {
1625  addr = rtsp_st->sdp_ip;
1626  port = rtsp_st->sdp_port;
1627  ttl = rtsp_st->sdp_ttl;
1628  }
1629  if (ttl > 0)
1630  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1631  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1632  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1633  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1634  port, "%s", optbuf);
1637  err = AVERROR_INVALIDDATA;
1638  goto fail;
1639  }
1640  break;
1641  }
1642  }
1643 
1644  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1645  goto fail;
1646  }
1647 
1648  if (rt->nb_rtsp_streams && reply->timeout > 0)
1649  rt->timeout = reply->timeout;
1650 
1651  if (rt->server_type == RTSP_SERVER_REAL)
1652  rt->need_subscription = 1;
1653 
1654  return 0;
1655 
1656 fail:
1657  ff_rtsp_undo_setup(s, 0);
1658  return err;
1659 }
1660 
1662 {
1663  RTSPState *rt = s->priv_data;
1664  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1665  ffurl_close(rt->rtsp_hd);
1666  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1667 }
1668 
1670 {
1671  RTSPState *rt = s->priv_data;
1672  char proto[128], host[1024], path[1024];
1673  char tcpname[1024], cmd[2048], auth[128];
1674  const char *lower_rtsp_proto = "tcp";
1675  int port, err, tcp_fd;
1676  RTSPMessageHeader reply1, *reply = &reply1;
1677  int lower_transport_mask = 0;
1678  int default_port = RTSP_DEFAULT_PORT;
1679  int https_tunnel = 0;
1680  char real_challenge[64] = "";
1681  struct sockaddr_storage peer;
1682  socklen_t peer_len = sizeof(peer);
1683 
1684  if (rt->rtp_port_max < rt->rtp_port_min) {
1685  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1686  "than min port %d\n", rt->rtp_port_max,
1687  rt->rtp_port_min);
1688  return AVERROR(EINVAL);
1689  }
1690 
1691  if (!ff_network_init())
1692  return AVERROR(EIO);
1693 
1694  if (s->max_delay < 0) /* Not set by the caller */
1696 
1699  (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1700  https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1703  }
1704  /* Only pass through valid flags from here */
1706 
1707 redirect:
1708  memset(&reply1, 0, sizeof(reply1));
1709  /* extract hostname and port */
1710  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1711  host, sizeof(host), &port, path, sizeof(path), s->url);
1712 
1713  if (!strcmp(proto, "rtsps")) {
1714  lower_rtsp_proto = "tls";
1715  default_port = RTSPS_DEFAULT_PORT;
1717  }
1718 
1719  if (*auth) {
1720  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1721  }
1722  if (port < 0)
1723  port = default_port;
1724 
1725  lower_transport_mask = rt->lower_transport_mask;
1726 
1727  if (!lower_transport_mask)
1728  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1729 
1730  if (s->oformat) {
1731  /* Only UDP or TCP - UDP multicast isn't supported. */
1732  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1733  (1 << RTSP_LOWER_TRANSPORT_TCP);
1734  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1735  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1736  "only UDP and TCP are supported for output.\n");
1737  err = AVERROR(EINVAL);
1738  goto fail;
1739  }
1740  }
1741 
1742  /* Construct the URI used in request; this is similar to s->url,
1743  * but with authentication credentials removed and RTSP specific options
1744  * stripped out. */
1745  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1746  host, port, "%s", path);
1747 
1748  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1749  /* set up initial handshake for tunneling */
1750  char httpname[1024];
1751  char sessioncookie[17];
1752  char headers[1024];
1754 
1755  av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1756 
1757  ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1758  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1760 
1761  /* GET requests */
1762  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1763  &s->interrupt_callback) < 0) {
1764  err = AVERROR(EIO);
1765  goto fail;
1766  }
1767 
1768  /* generate GET headers */
1769  snprintf(headers, sizeof(headers),
1770  "x-sessioncookie: %s\r\n"
1771  "Accept: application/x-rtsp-tunnelled\r\n"
1772  "Pragma: no-cache\r\n"
1773  "Cache-Control: no-cache\r\n",
1774  sessioncookie);
1775  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1776 
1777  if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1779  if (!rt->rtsp_hd->protocol_whitelist) {
1780  err = AVERROR(ENOMEM);
1781  goto fail;
1782  }
1783  }
1784 
1785  /* complete the connection */
1786  if (ffurl_connect(rt->rtsp_hd, &options)) {
1787  av_dict_free(&options);
1788  err = AVERROR(EIO);
1789  goto fail;
1790  }
1791 
1792  /* POST requests */
1793  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1794  &s->interrupt_callback) < 0 ) {
1795  err = AVERROR(EIO);
1796  goto fail;
1797  }
1798 
1799  /* generate POST headers */
1800  snprintf(headers, sizeof(headers),
1801  "x-sessioncookie: %s\r\n"
1802  "Content-Type: application/x-rtsp-tunnelled\r\n"
1803  "Pragma: no-cache\r\n"
1804  "Cache-Control: no-cache\r\n"
1805  "Content-Length: 32767\r\n"
1806  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1807  sessioncookie);
1808  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1809  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1810  av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1811 
1812  /* Initialize the authentication state for the POST session. The HTTP
1813  * protocol implementation doesn't properly handle multi-pass
1814  * authentication for POST requests, since it would require one of
1815  * the following:
1816  * - implementing Expect: 100-continue, which many HTTP servers
1817  * don't support anyway, even less the RTSP servers that do HTTP
1818  * tunneling
1819  * - sending the whole POST data until getting a 401 reply specifying
1820  * what authentication method to use, then resending all that data
1821  * - waiting for potential 401 replies directly after sending the
1822  * POST header (waiting for some unspecified time)
1823  * Therefore, we copy the full auth state, which works for both basic
1824  * and digest. (For digest, we would have to synchronize the nonce
1825  * count variable between the two sessions, if we'd do more requests
1826  * with the original session, though.)
1827  */
1829 
1830  /* complete the connection */
1831  if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1832  av_dict_free(&options);
1833  err = AVERROR(EIO);
1834  goto fail;
1835  }
1836  av_dict_free(&options);
1837  } else {
1838  int ret;
1839  /* open the tcp connection */
1840  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1841  host, port,
1842  "?timeout=%d", rt->stimeout);
1843  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1845  err = ret;
1846  goto fail;
1847  }
1848  rt->rtsp_hd_out = rt->rtsp_hd;
1849  }
1850  rt->seq = 0;
1851 
1852  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1853  if (tcp_fd < 0) {
1854  err = tcp_fd;
1855  goto fail;
1856  }
1857  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1858  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1859  NULL, 0, NI_NUMERICHOST);
1860  }
1861 
1862  /* request options supported by the server; this also detects server
1863  * type */
1864  for (rt->server_type = RTSP_SERVER_RTP;;) {
1865  cmd[0] = 0;
1866  if (rt->server_type == RTSP_SERVER_REAL)
1867  av_strlcat(cmd,
1868  /*
1869  * The following entries are required for proper
1870  * streaming from a Realmedia server. They are
1871  * interdependent in some way although we currently
1872  * don't quite understand how. Values were copied
1873  * from mplayer SVN r23589.
1874  * ClientChallenge is a 16-byte ID in hex
1875  * CompanyID is a 16-byte ID in base64
1876  */
1877  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1878  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1879  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1880  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1881  sizeof(cmd));
1882  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1883  if (reply->status_code != RTSP_STATUS_OK) {
1885  goto fail;
1886  }
1887 
1888  /* detect server type if not standard-compliant RTP */
1889  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1891  continue;
1892  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1894  } else if (rt->server_type == RTSP_SERVER_REAL)
1895  strcpy(real_challenge, reply->real_challenge);
1896  break;
1897  }
1898 
1899  if (CONFIG_RTSP_DEMUXER && s->iformat)
1900  err = ff_rtsp_setup_input_streams(s, reply);
1901  else if (CONFIG_RTSP_MUXER)
1902  err = ff_rtsp_setup_output_streams(s, host);
1903  else
1904  av_assert0(0);
1905  if (err)
1906  goto fail;
1907 
1908  do {
1909  int lower_transport = ff_log2_tab[lower_transport_mask &
1910  ~(lower_transport_mask - 1)];
1911 
1912  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1913  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1914  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1915 
1916  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1917  rt->server_type == RTSP_SERVER_REAL ?
1918  real_challenge : NULL);
1919  if (err < 0)
1920  goto fail;
1921  lower_transport_mask &= ~(1 << lower_transport);
1922  if (lower_transport_mask == 0 && err == 1) {
1923  err = AVERROR(EPROTONOSUPPORT);
1924  goto fail;
1925  }
1926  } while (err);
1927 
1928  rt->lower_transport_mask = lower_transport_mask;
1929  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1930  rt->state = RTSP_STATE_IDLE;
1931  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1932  return 0;
1933  fail:
1936  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1937  char *new_url = av_strdup(reply->location);
1938  if (!new_url) {
1939  err = AVERROR(ENOMEM);
1940  goto fail2;
1941  }
1942  ff_format_set_url(s, new_url);
1943  rt->session_id[0] = '\0';
1944  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1945  reply->status_code,
1946  s->url);
1947  goto redirect;
1948  }
1949  fail2:
1950  ff_network_close();
1951  return err;
1952 }
1953 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1954 
1955 #if CONFIG_RTPDEC
1956 static int parse_rtsp_message(AVFormatContext *s)
1957 {
1958  RTSPState *rt = s->priv_data;
1959  int ret;
1960 
1961  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1962  if (rt->state == RTSP_STATE_STREAMING) {
1964  return AVERROR_EOF;
1965  else
1967  "Unable to answer to TEARDOWN\n");
1968  } else
1969  return 0;
1970  } else {
1971  RTSPMessageHeader reply;
1972  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1973  if (ret < 0)
1974  return ret;
1975  /* XXX: parse message */
1976  if (rt->state != RTSP_STATE_STREAMING)
1977  return 0;
1978  }
1979 
1980  return 0;
1981 }
1982 
1983 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1984  uint8_t *buf, int buf_size, int64_t wait_end)
1985 {
1986  RTSPState *rt = s->priv_data;
1987  RTSPStream *rtsp_st;
1988  int n, i, ret, timeout_cnt = 0;
1989  struct pollfd *p = rt->p;
1990  int *fds = NULL, fdsnum, fdsidx;
1991 
1992  if (!p) {
1993  p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
1994  if (!p)
1995  return AVERROR(ENOMEM);
1996 
1997  if (rt->rtsp_hd) {
1998  p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1999  p[rt->max_p++].events = POLLIN;
2000  }
2001  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2002  rtsp_st = rt->rtsp_streams[i];
2003  if (rtsp_st->rtp_handle) {
2004  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2005  &fds, &fdsnum)) {
2006  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2007  return ret;
2008  }
2009  if (fdsnum != 2) {
2010  av_log(s, AV_LOG_ERROR,
2011  "Number of fds %d not supported\n", fdsnum);
2012  return AVERROR_INVALIDDATA;
2013  }
2014  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2015  p[rt->max_p].fd = fds[fdsidx];
2016  p[rt->max_p++].events = POLLIN;
2017  }
2018  av_freep(&fds);
2019  }
2020  }
2021  }
2022 
2023  for (;;) {
2025  return AVERROR_EXIT;
2026  if (wait_end && wait_end - av_gettime_relative() < 0)
2027  return AVERROR(EAGAIN);
2028  n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2029  if (n > 0) {
2030  int j = rt->rtsp_hd ? 1 : 0;
2031  timeout_cnt = 0;
2032  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2033  rtsp_st = rt->rtsp_streams[i];
2034  if (rtsp_st->rtp_handle) {
2035  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2036  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2037  if (ret > 0) {
2038  *prtsp_st = rtsp_st;
2039  return ret;
2040  }
2041  }
2042  j+=2;
2043  }
2044  }
2045 #if CONFIG_RTSP_DEMUXER
2046  if (rt->rtsp_hd && p[0].revents & POLLIN) {
2047  if ((ret = parse_rtsp_message(s)) < 0) {
2048  return ret;
2049  }
2050  }
2051 #endif
2052  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2053  return AVERROR(ETIMEDOUT);
2054  } else if (n < 0 && errno != EINTR)
2055  return AVERROR(errno);
2056  }
2057 }
2058 
2059 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2060  const uint8_t *buf, int len)
2061 {
2062  RTSPState *rt = s->priv_data;
2063  int i;
2064  if (len < 0)
2065  return len;
2066  if (rt->nb_rtsp_streams == 1) {
2067  *rtsp_st = rt->rtsp_streams[0];
2068  return len;
2069  }
2070  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2071  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2072  int no_ssrc = 0;
2073  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2075  if (!rtpctx)
2076  continue;
2077  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2078  *rtsp_st = rt->rtsp_streams[i];
2079  return len;
2080  }
2081  if (!rtpctx->ssrc)
2082  no_ssrc = 1;
2083  }
2084  if (no_ssrc) {
2086  "Unable to pick stream for packet - SSRC not known for "
2087  "all streams\n");
2088  return AVERROR(EAGAIN);
2089  }
2090  } else {
2091  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2092  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2093  *rtsp_st = rt->rtsp_streams[i];
2094  return len;
2095  }
2096  }
2097  }
2098  }
2099  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2100  return AVERROR(EAGAIN);
2101 }
2102 
2103 static int read_packet(AVFormatContext *s,
2104  RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2105  int64_t wait_end)
2106 {
2107  RTSPState *rt = s->priv_data;
2108  int len;
2109 
2110  switch(rt->lower_transport) {
2111  default:
2112 #if CONFIG_RTSP_DEMUXER
2114  len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2115  break;
2116 #endif
2119  len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2120  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2121  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2122  break;
2124  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2125  wait_end && wait_end < av_gettime_relative())
2126  len = AVERROR(EAGAIN);
2127  else
2128  len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2129  len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2130  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2131  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2132  break;
2133  }
2134 
2135  if (len == 0)
2136  return AVERROR_EOF;
2137 
2138  return len;
2139 }
2140 
2142 {
2143  RTSPState *rt = s->priv_data;
2144  int ret, len;
2145  RTSPStream *rtsp_st, *first_queue_st = NULL;
2146  int64_t wait_end = 0;
2147 
2148  if (rt->nb_byes == rt->nb_rtsp_streams)
2149  return AVERROR_EOF;
2150 
2151  /* get next frames from the same RTP packet */
2152  if (rt->cur_transport_priv) {
2153  if (rt->transport == RTSP_TRANSPORT_RDT) {
2154  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2155  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2156  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2157  } else if (CONFIG_RTPDEC && rt->ts) {
2158  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2159  if (ret >= 0) {
2160  rt->recvbuf_pos += ret;
2161  ret = rt->recvbuf_pos < rt->recvbuf_len;
2162  }
2163  } else
2164  ret = -1;
2165  if (ret == 0) {
2166  rt->cur_transport_priv = NULL;
2167  return 0;
2168  } else if (ret == 1) {
2169  return 0;
2170  } else
2171  rt->cur_transport_priv = NULL;
2172  }
2173 
2174 redo:
2175  if (rt->transport == RTSP_TRANSPORT_RTP) {
2176  int i;
2177  int64_t first_queue_time = 0;
2178  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2179  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2180  int64_t queue_time;
2181  if (!rtpctx)
2182  continue;
2183  queue_time = ff_rtp_queued_packet_time(rtpctx);
2184  if (queue_time && (queue_time - first_queue_time < 0 ||
2185  !first_queue_time)) {
2186  first_queue_time = queue_time;
2187  first_queue_st = rt->rtsp_streams[i];
2188  }
2189  }
2190  if (first_queue_time) {
2191  wait_end = first_queue_time + s->max_delay;
2192  } else {
2193  wait_end = 0;
2194  first_queue_st = NULL;
2195  }
2196  }
2197 
2198  /* read next RTP packet */
2199  if (!rt->recvbuf) {
2201  if (!rt->recvbuf)
2202  return AVERROR(ENOMEM);
2203  }
2204 
2205  len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2206  if (len == AVERROR(EAGAIN) && first_queue_st &&
2207  rt->transport == RTSP_TRANSPORT_RTP) {
2209  "max delay reached. need to consume packet\n");
2210  rtsp_st = first_queue_st;
2211  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2212  goto end;
2213  }
2214  if (len < 0)
2215  return len;
2216 
2217  if (rt->transport == RTSP_TRANSPORT_RDT) {
2218  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2219  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2220  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2221  if (rtsp_st->feedback) {
2222  AVIOContext *pb = NULL;
2224  pb = s->pb;
2225  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2226  }
2227  if (ret < 0) {
2228  /* Either bad packet, or a RTCP packet. Check if the
2229  * first_rtcp_ntp_time field was initialized. */
2230  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2231  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2232  /* first_rtcp_ntp_time has been initialized for this stream,
2233  * copy the same value to all other uninitialized streams,
2234  * in order to map their timestamp origin to the same ntp time
2235  * as this one. */
2236  int i;
2237  AVStream *st = NULL;
2238  if (rtsp_st->stream_index >= 0)
2239  st = s->streams[rtsp_st->stream_index];
2240  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2241  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2242  AVStream *st2 = NULL;
2243  if (rt->rtsp_streams[i]->stream_index >= 0)
2244  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2245  if (rtpctx2 && st && st2 &&
2246  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2247  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2248  rtpctx2->rtcp_ts_offset = av_rescale_q(
2249  rtpctx->rtcp_ts_offset, st->time_base,
2250  st2->time_base);
2251  }
2252  }
2253  // Make real NTP start time available in AVFormatContext
2254  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2255  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2256  if (rtpctx->st) {
2257  s->start_time_realtime -=
2258  av_rescale (rtpctx->rtcp_ts_offset,
2259  (uint64_t) rtpctx->st->time_base.num * 1000000,
2260  rtpctx->st->time_base.den);
2261  }
2262  }
2263  }
2264  if (ret == -RTCP_BYE) {
2265  rt->nb_byes++;
2266 
2267  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2268  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2269 
2270  if (rt->nb_byes == rt->nb_rtsp_streams)
2271  return AVERROR_EOF;
2272  }
2273  }
2274  } else if (CONFIG_RTPDEC && rt->ts) {
2275  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2276  if (ret >= 0) {
2277  if (ret < len) {
2278  rt->recvbuf_len = len;
2279  rt->recvbuf_pos = ret;
2280  rt->cur_transport_priv = rt->ts;
2281  return 1;
2282  } else {
2283  ret = 0;
2284  }
2285  }
2286  } else {
2287  return AVERROR_INVALIDDATA;
2288  }
2289 end:
2290  if (ret < 0)
2291  goto redo;
2292  if (ret == 1)
2293  /* more packets may follow, so we save the RTP context */
2294  rt->cur_transport_priv = rtsp_st->transport_priv;
2295 
2296  return ret;
2297 }
2298 #endif /* CONFIG_RTPDEC */
2299 
2300 #if CONFIG_SDP_DEMUXER
2301 static int sdp_probe(const AVProbeData *p1)
2302 {
2303  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2304 
2305  /* we look for a line beginning "c=IN IP" */
2306  while (p < p_end && *p != '\0') {
2307  if (sizeof("c=IN IP") - 1 < p_end - p &&
2308  av_strstart(p, "c=IN IP", NULL))
2309  return AVPROBE_SCORE_EXTENSION;
2310 
2311  while (p < p_end - 1 && *p != '\n') p++;
2312  if (++p >= p_end)
2313  break;
2314  if (*p == '\r')
2315  p++;
2316  }
2317  return 0;
2318 }
2319 
2320 static void append_source_addrs(char *buf, int size, const char *name,
2321  int count, struct RTSPSource **addrs)
2322 {
2323  int i;
2324  if (!count)
2325  return;
2326  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2327  for (i = 1; i < count; i++)
2328  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2329 }
2330 
2331 static int sdp_read_header(AVFormatContext *s)
2332 {
2333  RTSPState *rt = s->priv_data;
2334  RTSPStream *rtsp_st;
2335  int size, i, err;
2336  char *content;
2337  char url[1024];
2338 
2339  if (!ff_network_init())
2340  return AVERROR(EIO);
2341 
2342  if (s->max_delay < 0) /* Not set by the caller */
2344  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2346 
2347  /* read the whole sdp file */
2348  /* XXX: better loading */
2349  content = av_malloc(SDP_MAX_SIZE);
2350  if (!content)
2351  return AVERROR(ENOMEM);
2352  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2353  if (size <= 0) {
2354  av_free(content);
2355  return AVERROR_INVALIDDATA;
2356  }
2357  content[size] ='\0';
2358 
2359  err = ff_sdp_parse(s, content);
2360  av_freep(&content);
2361  if (err) goto fail;
2362 
2363  /* open each RTP stream */
2364  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2365  char namebuf[50];
2366  rtsp_st = rt->rtsp_streams[i];
2367 
2368  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2369  AVDictionary *opts = map_to_opts(rt);
2370 
2371  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2372  sizeof(rtsp_st->sdp_ip),
2373  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2374  if (err) {
2375  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2376  err = AVERROR(EIO);
2377  av_dict_free(&opts);
2378  goto fail;
2379  }
2380  ff_url_join(url, sizeof(url), "rtp", NULL,
2381  namebuf, rtsp_st->sdp_port,
2382  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2383  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2384  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2385  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2386 
2387  append_source_addrs(url, sizeof(url), "sources",
2388  rtsp_st->nb_include_source_addrs,
2389  rtsp_st->include_source_addrs);
2390  append_source_addrs(url, sizeof(url), "block",
2391  rtsp_st->nb_exclude_source_addrs,
2392  rtsp_st->exclude_source_addrs);
2393  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2395 
2396  av_dict_free(&opts);
2397 
2398  if (err < 0) {
2399  err = AVERROR_INVALIDDATA;
2400  goto fail;
2401  }
2402  }
2403  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2404  goto fail;
2405  }
2406  return 0;
2407 fail:
2409  ff_network_close();
2410  return err;
2411 }
2412 
2413 static int sdp_read_close(AVFormatContext *s)
2414 {
2416  ff_network_close();
2417  return 0;
2418 }
2419 
2420 static const AVClass sdp_demuxer_class = {
2421  .class_name = "SDP demuxer",
2422  .item_name = av_default_item_name,
2423  .option = sdp_options,
2424  .version = LIBAVUTIL_VERSION_INT,
2425 };
2426 
2428  .name = "sdp",
2429  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2430  .priv_data_size = sizeof(RTSPState),
2431  .read_probe = sdp_probe,
2432  .read_header = sdp_read_header,
2434  .read_close = sdp_read_close,
2435  .priv_class = &sdp_demuxer_class,
2436 };
2437 #endif /* CONFIG_SDP_DEMUXER */
2438 
2439 #if CONFIG_RTP_DEMUXER
2440 static int rtp_probe(const AVProbeData *p)
2441 {
2442  if (av_strstart(p->filename, "rtp:", NULL))
2443  return AVPROBE_SCORE_MAX;
2444  return 0;
2445 }
2446 
2447 static int rtp_read_header(AVFormatContext *s)
2448 {
2449  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2450  char host[500], sdp[500];
2451  int ret, port;
2452  URLContext* in = NULL;
2453  int payload_type;
2454  AVCodecParameters *par = NULL;
2455  struct sockaddr_storage addr;
2456  AVIOContext pb;
2457  socklen_t addrlen = sizeof(addr);
2458  RTSPState *rt = s->priv_data;
2459 
2460  if (!ff_network_init())
2461  return AVERROR(EIO);
2462 
2463  ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2465  if (ret)
2466  goto fail;
2467 
2468  while (1) {
2469  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2470  if (ret == AVERROR(EAGAIN))
2471  continue;
2472  if (ret < 0)
2473  goto fail;
2474  if (ret < 12) {
2475  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2476  continue;
2477  }
2478 
2479  if ((recvbuf[0] & 0xc0) != 0x80) {
2480  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2481  "received\n");
2482  continue;
2483  }
2484 
2485  if (RTP_PT_IS_RTCP(recvbuf[1]))
2486  continue;
2487 
2488  payload_type = recvbuf[1] & 0x7f;
2489  break;
2490  }
2491  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2492  ffurl_close(in);
2493  in = NULL;
2494 
2495  par = avcodec_parameters_alloc();
2496  if (!par) {
2497  ret = AVERROR(ENOMEM);
2498  goto fail;
2499  }
2500 
2501  if (ff_rtp_get_codec_info(par, payload_type)) {
2502  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2503  "without an SDP file describing it\n",
2504  payload_type);
2505  goto fail;
2506  }
2507  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2508  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2509  "properly you need an SDP file "
2510  "describing it\n");
2511  }
2512 
2513  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2514  NULL, 0, s->url);
2515 
2516  snprintf(sdp, sizeof(sdp),
2517  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2518  addr.ss_family == AF_INET ? 4 : 6, host,
2519  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2520  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2521  port, payload_type);
2522  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2524 
2525  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2526  s->pb = &pb;
2527 
2528  /* sdp_read_header initializes this again */
2529  ff_network_close();
2530 
2531  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2532 
2533  ret = sdp_read_header(s);
2534  s->pb = NULL;
2535  return ret;
2536 
2537 fail:
2539  if (in)
2540  ffurl_close(in);
2541  ff_network_close();
2542  return ret;
2543 }
2544 
2545 static const AVClass rtp_demuxer_class = {
2546  .class_name = "RTP demuxer",
2547  .item_name = av_default_item_name,
2548  .option = rtp_options,
2549  .version = LIBAVUTIL_VERSION_INT,
2550 };
2551 
2553  .name = "rtp",
2554  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2555  .priv_data_size = sizeof(RTSPState),
2556  .read_probe = rtp_probe,
2557  .read_header = rtp_read_header,
2559  .read_close = sdp_read_close,
2560  .flags = AVFMT_NOFILE,
2561  .priv_class = &rtp_demuxer_class,
2562 };
2563 #endif /* CONFIG_RTP_DEMUXER */
const char * name
Definition: avisynth_c.h:867
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:274
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a &#39;$&#39;, stream length and stre...
Definition: rtsp.h:94
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:4732
char crypto_suite[40]
Definition: rtsp.h:478
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:389
#define NULL
Definition: coverity.c:32
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:100
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:581
Bytestream IO Context.
Definition: avio.h:161
Realmedia Data Transport.
Definition: rtsp.h:59
#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:521
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:1611
int size
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:1636
AVOption.
Definition: opt.h:246
HTTPS tunneled.
Definition: rtsp.h:45
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:116
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:188
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
const char * filename
Definition: avformat.h:447
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:177
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:318
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:4886
int av_parse_time(int64_t *timeval, const char *timestr, int duration)
Parse timestr and return in *time a corresponding number of microseconds.
Definition: parseutils.c:587
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:421
const char * desc
Definition: nvenc.c:68
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:73
#define CONFIG_RTPDEC
Definition: config.h:665
Windows Media server.
Definition: rtsp.h:210
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:355
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:808
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:3232
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:3957
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:882
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { 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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) 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);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=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) { int planes=out->planar ? out->ch_count :1;unsigned m=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){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
#define AVIO_FLAG_READ
read-only
Definition: avio.h:654
char * user_agent
User-Agent string.
Definition: rtsp.h:409
char location[4096]
the "Location:" field.
Definition: rtsp.h:153
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:655
int mode_record
transport set to record data
Definition: rtsp.h:113
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:716
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:116
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:145
initialized and sending/receiving data
Definition: rtsp.h:198
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:271
const char * key
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:423
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:80
#define freeaddrinfo
Definition: network.h:215
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:455
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1407
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:421
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:246
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:240
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:132
#define AI_NUMERICHOST
Definition: network.h:184
This struct describes the properties of an encoded stream.
Definition: avcodec.h:3949
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:122
This describes the server response to each RTSP command.
Definition: rtsp.h:128
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream &#39;st&#39;.
Definition: rtpdec.c:538
#define RECVBUF_SIZE
Definition: rtsp.c:59
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:143
Format I/O context.
Definition: avformat.h:1358
#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:208
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:404
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:426
int recvbuf_len
Definition: rtsp.h:324
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:179
#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:361
#define CONFIG_RTSP_DEMUXER
Definition: config.h:2205
Standards-compliant RTP.
Definition: rtsp.h:58
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:149
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:75
#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:110
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:199
int ff_network_init(void)
Definition: network.c:58
#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:2019
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:473
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:113
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int id
Format-specific stream ID.
Definition: avformat.h:888
enum AVStreamParseType need_parsing
Definition: avformat.h:1099
#define POLL_TIMEOUT_MS
Definition: rtsp.c:55
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:60
static void handler(vbi_event *ev, void *user_data)
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4459
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1426
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:374
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:329
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:438
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:46
char * protocol_whitelist
&#39;,&#39; separated list of allowed protocols.
Definition: avformat.h:1918
#define CONFIG_RTSP_MUXER
Definition: config.h:2432
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:132
#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:180
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
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:69
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 &#39;transports&#39; variable below
Definition: rtsp.h:135
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:647
AVInputFormat ff_rtp_demuxer
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:178
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:78
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:458
Private data for the RTSP demuxer.
Definition: rtsp.h:219
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:256
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 i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#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:1598
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:633
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:251
const char * protocol_whitelist
Definition: url.h:49
#define AV_RB16
Definition: intreadwrite.h:53
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
ff_const59 struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:1370
char * url
input or output URL.
Definition: avformat.h:1454
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const AVOption ff_rtsp_options[]
Definition: rtsp.c:83
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:3953
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:183
Definition: graph2dot.c:48
URLContext * rtsp_hd
Definition: rtsp.h:221
simple assert() macros that are a bit more flexible than ISO C assert().
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:332
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:456
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, const RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:574
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:2029
int64_t rtcp_ts_offset
Definition: rtpdec.h:181
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:120
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:226
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:165
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
Definition: codec_desc.c:3257
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:443
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:449
int seq
RTSP command sequence number.
Definition: rtsp.h:242
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:448
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:340
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1414
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:422
AVDictionary * opts
Definition: movenc.c:50
#define NI_NUMERICHOST
Definition: network.h:192
#define th
Definition: regdef.h:75
#define LIBAVFORMAT_IDENT
Definition: version.h:46
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:308
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
Definition: rtpdec.h:126
int recvbuf_pos
Definition: rtsp.h:323
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:198
int nb_rtsp_streams
number of items in the &#39;rtsp_streams&#39; variable
Definition: rtsp.h:224
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:284
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
int content_length
length of the data following this header
Definition: rtsp.h:130
int max_streams
The maximum number of streams.
Definition: avformat.h:1960
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:173
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:76
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:89
RTSP over HTTP (tunneling)
Definition: rtsp.h:70
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:142
#define s(width, name)
Definition: cbs_vp9.c:257
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
void ff_format_set_url(AVFormatContext *s, char *url)
Set AVFormatContext url field to the provided pointer.
Definition: utils.c:5801
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:168
int n
Definition: avisynth_c.h:760
AVDictionary * metadata
Definition: avformat.h:945
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:466
char crypto_params[100]
Definition: rtsp.h:479
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:200
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, const RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:626
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define ENC
Definition: rtsp.c:64
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:453
Raw data (over UDP)
Definition: rtsp.h:60
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:322
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
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
int sdp_payload_type
payload type
Definition: rtsp.h:460
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:457
ff_const59 struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1377
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1443
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:530
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:470
Stream structure.
Definition: avformat.h:881
#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:337
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:263
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:429
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:454
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:251
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:738
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:664
int rtp_port_max
Definition: rtsp.h:389
#define NTP_OFFSET
Definition: internal.h:244
Definition: rtp.h:100
AVIOContext * pb
I/O context.
Definition: avformat.h:1400
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:384
AVInputFormat ff_sdp_demuxer
int server_port_max
Definition: rtsp.h:106
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:66
static av_always_inline void RENAME() interleave(TYPE *dst, TYPE *src0, TYPE *src1, int w2, int add, int shift)
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:476
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:416
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:259
void * buf
Definition: avisynth_c.h:766
Definition: url.h:38
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
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:74
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:656
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:70
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:379
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
double value
Definition: eval.c:98
int client_port_max
Definition: rtsp.h:102
Describe the class of an AVClass context structure.
Definition: log.h:67
#define SDP_MAX_SIZE
Definition: rtsp.c:58
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:515
#define SPACE_CHARS
Definition: internal.h:354
void * priv_data
Definition: url.h:41
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:469
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:280
#define gai_strerror
Definition: network.h:222
not initialized
Definition: rtsp.h:197
int64_t range_end
Definition: rtsp.h:139
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:119
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:3251
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:156
AVMediaType
Definition: avutil.h:199
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:70
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:755
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
Definition: rtspdec.c:750
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:772
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&#39;t hear from them...
Definition: rtpdec.c:299
#define s1
Definition: regdef.h:38
const char * name
Name of the codec described by this descriptor.
Definition: avcodec.h:724
#define snprintf
Definition: snprintf.h:34
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:456
int max_p
Definition: rtsp.h:356
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4393
int buffer_size
Definition: rtsp.h:412
This structure contains the data a format has to probe a file.
Definition: avformat.h:446
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:77
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
Definition: avstring.c:93
int interleaved_max
Definition: rtsp.h:94
#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:716
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
#define flags(name, subs,...)
Definition: cbs_av1.c:561
enum RTSPServerType server_type
brand of server that we&#39;re talking to; e.g.
Definition: rtsp.h:268
int ffurl_close(URLContext *h)
Definition: avio.c:467
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:139
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1463
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:232
static int read_probe(const AVProbeData *pd)
Definition: jvdec.c:55
int sample_rate
Audio only.
Definition: avcodec.h:4067
#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:458
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:34
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:214
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:113
const OptionDef options[]
Definition: ffmpeg_opt.c:3364
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:129
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:463
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:289
int av_dict_set_int(AVDictionary **pm, const char *key, int64_t value, int flags)
Convenience wrapper for av_dict_set that converts the value to a string and stores it...
Definition: dict.c:147
int ffurl_read_complete(URLContext *h, unsigned char *buf, int size)
Read as many bytes as possible (up to size), calling the read function multiple times if necessary...
Definition: avio.c:414
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
Definition: rdt.c:94
int den
Denominator.
Definition: rational.h:60
char default_lang[4]
Definition: rtsp.h:411
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4431
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&#39;re reading data interleave...
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:399
#define getnameinfo
Definition: network.h:216
#define av_free(p)
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:161
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:277
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:79
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:449
void * priv_data
Format private data.
Definition: avformat.h:1386
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:593
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:882
int channels
Audio only.
Definition: avcodec.h:4063
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:459
#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:1254
char * protocol_blacklist
&#39;,&#39; separated list of disallowed protocols.
Definition: avformat.h:1953
int ai_flags
Definition: network.h:135
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1473
Realmedia-style server.
Definition: rtsp.h:209
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:345
#define av_freep(p)
void INT64 INT64 count
Definition: avisynth_c.h:766
void INT64 start
Definition: avisynth_c.h:766
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:654
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:120
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1028
#define av_malloc_array(a, b)
int pkt_size
Definition: rtsp.h:413
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:910
int interleaved_max
Definition: rtsp.h:447
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
Definition: rtpdec.c:869
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:115
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:101
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:3276
AVStream * st
Definition: rtpdec.h:151
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with a matching codec ID.
Definition: rtpdec.c:160
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:447
This structure stores compressed data.
Definition: avcodec.h:1454
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:1208
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:106
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:122
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
Definition: avio.c:407
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:439
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
#define OFFSET(x)
Definition: rtsp.c:62
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with the specified name.
Definition: rtpdec.c:146
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:98
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:440
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:102