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rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
32 #include "avformat.h"
33 #include "avio_internal.h"
34 
35 #if HAVE_POLL_H
36 #include <poll.h>
37 #endif
38 #include "internal.h"
39 #include "network.h"
40 #include "os_support.h"
41 #include "http.h"
42 #include "rtsp.h"
43 
44 #include "rtpdec.h"
45 #include "rdt.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
48 #include "url.h"
49 #include "rtpenc.h"
50 #include "mpegts.h"
51 
52 /* Timeout values for socket poll, in ms,
53  * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
60 
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 
65 #define RTSP_FLAG_OPTS(name, longname) \
66  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67  { "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" }, \
68  { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 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_DATA+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 
76 #define RTSP_REORDERING_OPTS() \
77  { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
78 
80  { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
81  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
82  { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
83  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
86  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
87  RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
88  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
89  { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
90  { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
91  { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
92  { "stimeout", "timeout (in micro seconds) of socket i/o operations.", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
94  { NULL },
95 };
96 
97 static const AVOption sdp_options[] = {
98  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
99  { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
100  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
102  { NULL },
103 };
104 
105 static const AVOption rtp_options[] = {
106  RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
108  { NULL },
109 };
110 
111 static void get_word_until_chars(char *buf, int buf_size,
112  const char *sep, const char **pp)
113 {
114  const char *p;
115  char *q;
116 
117  p = *pp;
118  p += strspn(p, SPACE_CHARS);
119  q = buf;
120  while (!strchr(sep, *p) && *p != '\0') {
121  if ((q - buf) < buf_size - 1)
122  *q++ = *p;
123  p++;
124  }
125  if (buf_size > 0)
126  *q = '\0';
127  *pp = p;
128 }
129 
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
131  const char **pp)
132 {
133  if (**pp == '/') (*pp)++;
134  get_word_until_chars(buf, buf_size, sep, pp);
135 }
136 
137 static void get_word(char *buf, int buf_size, const char **pp)
138 {
139  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
140 }
141 
142 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
143  * and end time.
144  * Used for seeking in the rtp stream.
145  */
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
147 {
148  char buf[256];
149 
150  p += strspn(p, SPACE_CHARS);
151  if (!av_stristart(p, "npt=", &p))
152  return;
153 
154  *start = AV_NOPTS_VALUE;
155  *end = AV_NOPTS_VALUE;
156 
157  get_word_sep(buf, sizeof(buf), "-", &p);
158  av_parse_time(start, buf, 1);
159  if (*p == '-') {
160  p++;
161  get_word_sep(buf, sizeof(buf), "-", &p);
162  av_parse_time(end, buf, 1);
163  }
164 }
165 
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
167 {
168  struct addrinfo hints = { 0 }, *ai = NULL;
169  hints.ai_flags = AI_NUMERICHOST;
170  if (getaddrinfo(buf, NULL, &hints, &ai))
171  return -1;
172  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
173  freeaddrinfo(ai);
174  return 0;
175 }
176 
177 #if CONFIG_RTPDEC
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179  RTSPStream *rtsp_st, AVCodecContext *codec)
180 {
181  if (!handler)
182  return;
183  if (codec)
184  codec->codec_id = handler->codec_id;
185  rtsp_st->dynamic_handler = handler;
186  if (handler->alloc) {
187  rtsp_st->dynamic_protocol_context = handler->alloc();
188  if (!rtsp_st->dynamic_protocol_context)
189  rtsp_st->dynamic_handler = NULL;
190  }
191 }
192 
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195  AVStream *st, RTSPStream *rtsp_st,
196  int payload_type, const char *p)
197 {
198  AVCodecContext *codec = st->codec;
199  char buf[256];
200  int i;
201  AVCodec *c;
202  const char *c_name;
203 
204  /* See if we can handle this kind of payload.
205  * The space should normally not be there but some Real streams or
206  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207  * have a trailing space. */
208  get_word_sep(buf, sizeof(buf), "/ ", &p);
209  if (payload_type < RTP_PT_PRIVATE) {
210  /* We are in a standard case
211  * (from http://www.iana.org/assignments/rtp-parameters). */
212  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
213  }
214 
215  if (codec->codec_id == AV_CODEC_ID_NONE) {
216  RTPDynamicProtocolHandler *handler =
218  init_rtp_handler(handler, rtsp_st, codec);
219  /* If no dynamic handler was found, check with the list of standard
220  * allocated types, if such a stream for some reason happens to
221  * use a private payload type. This isn't handled in rtpdec.c, since
222  * the format name from the rtpmap line never is passed into rtpdec. */
223  if (!rtsp_st->dynamic_handler)
224  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
225  }
226 
227  c = avcodec_find_decoder(codec->codec_id);
228  if (c && c->name)
229  c_name = c->name;
230  else
231  c_name = "(null)";
232 
233  get_word_sep(buf, sizeof(buf), "/", &p);
234  i = atoi(buf);
235  switch (codec->codec_type) {
236  case AVMEDIA_TYPE_AUDIO:
237  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
240  if (i > 0) {
241  codec->sample_rate = i;
242  avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243  get_word_sep(buf, sizeof(buf), "/", &p);
244  i = atoi(buf);
245  if (i > 0)
246  codec->channels = i;
247  }
248  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
249  codec->sample_rate);
250  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
251  codec->channels);
252  break;
253  case AVMEDIA_TYPE_VIDEO:
254  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
255  if (i > 0)
256  avpriv_set_pts_info(st, 32, 1, i);
257  break;
258  default:
259  break;
260  }
261  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262  rtsp_st->dynamic_handler->init(s, st->index,
263  rtsp_st->dynamic_protocol_context);
264  return 0;
265 }
266 
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268  * is broken out as a function because it is used in rtp_h264.c, which is
269  * forthcoming. */
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271  char *value, int value_size)
272 {
273  *p += strspn(*p, SPACE_CHARS);
274  if (**p) {
275  get_word_sep(attr, attr_size, "=", p);
276  if (**p == '=')
277  (*p)++;
278  get_word_sep(value, value_size, ";", p);
279  if (**p == ';')
280  (*p)++;
281  return 1;
282  }
283  return 0;
284 }
285 
286 typedef struct SDPParseState {
287  /* SDP only */
288  struct sockaddr_storage default_ip;
289  int default_ttl;
290  int skip_media; ///< set if an unknown m= line occurs
291 } SDPParseState;
292 
293 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
294  int letter, const char *buf)
295 {
296  RTSPState *rt = s->priv_data;
297  char buf1[64], st_type[64];
298  const char *p;
299  enum AVMediaType codec_type;
300  int payload_type, i;
301  AVStream *st;
302  RTSPStream *rtsp_st;
303  struct sockaddr_storage sdp_ip;
304  int ttl;
305 
306  av_dlog(s, "sdp: %c='%s'\n", letter, buf);
307 
308  p = buf;
309  if (s1->skip_media && letter != 'm')
310  return;
311  switch (letter) {
312  case 'c':
313  get_word(buf1, sizeof(buf1), &p);
314  if (strcmp(buf1, "IN") != 0)
315  return;
316  get_word(buf1, sizeof(buf1), &p);
317  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
318  return;
319  get_word_sep(buf1, sizeof(buf1), "/", &p);
320  if (get_sockaddr(buf1, &sdp_ip))
321  return;
322  ttl = 16;
323  if (*p == '/') {
324  p++;
325  get_word_sep(buf1, sizeof(buf1), "/", &p);
326  ttl = atoi(buf1);
327  }
328  if (s->nb_streams == 0) {
329  s1->default_ip = sdp_ip;
330  s1->default_ttl = ttl;
331  } else {
332  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
333  rtsp_st->sdp_ip = sdp_ip;
334  rtsp_st->sdp_ttl = ttl;
335  }
336  break;
337  case 's':
338  av_dict_set(&s->metadata, "title", p, 0);
339  break;
340  case 'i':
341  if (s->nb_streams == 0) {
342  av_dict_set(&s->metadata, "comment", p, 0);
343  break;
344  }
345  break;
346  case 'm':
347  /* new stream */
348  s1->skip_media = 0;
349  codec_type = AVMEDIA_TYPE_UNKNOWN;
350  get_word(st_type, sizeof(st_type), &p);
351  if (!strcmp(st_type, "audio")) {
352  codec_type = AVMEDIA_TYPE_AUDIO;
353  } else if (!strcmp(st_type, "video")) {
354  codec_type = AVMEDIA_TYPE_VIDEO;
355  } else if (!strcmp(st_type, "application")) {
356  codec_type = AVMEDIA_TYPE_DATA;
357  }
358  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
359  s1->skip_media = 1;
360  return;
361  }
362  rtsp_st = av_mallocz(sizeof(RTSPStream));
363  if (!rtsp_st)
364  return;
365  rtsp_st->stream_index = -1;
366  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
367 
368  rtsp_st->sdp_ip = s1->default_ip;
369  rtsp_st->sdp_ttl = s1->default_ttl;
370 
371  get_word(buf1, sizeof(buf1), &p); /* port */
372  rtsp_st->sdp_port = atoi(buf1);
373 
374  get_word(buf1, sizeof(buf1), &p); /* protocol */
375  if (!strcmp(buf1, "udp"))
377  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
378  rtsp_st->feedback = 1;
379 
380  /* XXX: handle list of formats */
381  get_word(buf1, sizeof(buf1), &p); /* format list */
382  rtsp_st->sdp_payload_type = atoi(buf1);
383 
384  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
385  /* no corresponding stream */
386  if (rt->transport == RTSP_TRANSPORT_RAW) {
387  if (!rt->ts && CONFIG_RTPDEC)
388  rt->ts = ff_mpegts_parse_open(s);
389  } else {
390  RTPDynamicProtocolHandler *handler;
391  handler = ff_rtp_handler_find_by_id(
393  init_rtp_handler(handler, rtsp_st, NULL);
394  if (handler && handler->init)
395  handler->init(s, -1, rtsp_st->dynamic_protocol_context);
396  }
397  } else if (rt->server_type == RTSP_SERVER_WMS &&
398  codec_type == AVMEDIA_TYPE_DATA) {
399  /* RTX stream, a stream that carries all the other actual
400  * audio/video streams. Don't expose this to the callers. */
401  } else {
402  st = avformat_new_stream(s, NULL);
403  if (!st)
404  return;
405  st->id = rt->nb_rtsp_streams - 1;
406  rtsp_st->stream_index = st->index;
407  st->codec->codec_type = codec_type;
408  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
409  RTPDynamicProtocolHandler *handler;
410  /* if standard payload type, we can find the codec right now */
412  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
413  st->codec->sample_rate > 0)
414  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
415  /* Even static payload types may need a custom depacketizer */
416  handler = ff_rtp_handler_find_by_id(
417  rtsp_st->sdp_payload_type, st->codec->codec_type);
418  init_rtp_handler(handler, rtsp_st, st->codec);
419  if (handler && handler->init)
420  handler->init(s, st->index,
421  rtsp_st->dynamic_protocol_context);
422  }
423  }
424  /* put a default control url */
425  av_strlcpy(rtsp_st->control_url, rt->control_uri,
426  sizeof(rtsp_st->control_url));
427  break;
428  case 'a':
429  if (av_strstart(p, "control:", &p)) {
430  if (s->nb_streams == 0) {
431  if (!strncmp(p, "rtsp://", 7))
432  av_strlcpy(rt->control_uri, p,
433  sizeof(rt->control_uri));
434  } else {
435  char proto[32];
436  /* get the control url */
437  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
438 
439  /* XXX: may need to add full url resolution */
440  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
441  NULL, NULL, 0, p);
442  if (proto[0] == '\0') {
443  /* relative control URL */
444  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
445  av_strlcat(rtsp_st->control_url, "/",
446  sizeof(rtsp_st->control_url));
447  av_strlcat(rtsp_st->control_url, p,
448  sizeof(rtsp_st->control_url));
449  } else
450  av_strlcpy(rtsp_st->control_url, p,
451  sizeof(rtsp_st->control_url));
452  }
453  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
454  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
455  get_word(buf1, sizeof(buf1), &p);
456  payload_type = atoi(buf1);
457  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
458  if (rtsp_st->stream_index >= 0) {
459  st = s->streams[rtsp_st->stream_index];
460  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
461  }
462  } else if (av_strstart(p, "fmtp:", &p) ||
463  av_strstart(p, "framesize:", &p)) {
464  /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
465  // let dynamic protocol handlers have a stab at the line.
466  get_word(buf1, sizeof(buf1), &p);
467  payload_type = atoi(buf1);
468  for (i = 0; i < rt->nb_rtsp_streams; i++) {
469  rtsp_st = rt->rtsp_streams[i];
470  if (rtsp_st->sdp_payload_type == payload_type &&
471  rtsp_st->dynamic_handler &&
473  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
474  rtsp_st->dynamic_protocol_context, buf);
475  }
476  } else if (av_strstart(p, "range:", &p)) {
477  int64_t start, end;
478 
479  // this is so that seeking on a streamed file can work.
480  rtsp_parse_range_npt(p, &start, &end);
481  s->start_time = start;
482  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
483  s->duration = (end == AV_NOPTS_VALUE) ?
484  AV_NOPTS_VALUE : end - start;
485  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
486  if (atoi(p) == 1)
488  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
489  s->nb_streams > 0) {
490  st = s->streams[s->nb_streams - 1];
491  st->codec->sample_rate = atoi(p);
492  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
493  // RFC 4568
494  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
495  get_word(buf1, sizeof(buf1), &p); // ignore tag
496  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
497  p += strspn(p, SPACE_CHARS);
498  if (av_strstart(p, "inline:", &p))
499  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
500  } else {
501  if (rt->server_type == RTSP_SERVER_WMS)
503  if (s->nb_streams > 0) {
504  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
505 
506  if (rt->server_type == RTSP_SERVER_REAL)
507  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
508 
509  if (rtsp_st->dynamic_handler &&
511  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
512  rtsp_st->stream_index,
513  rtsp_st->dynamic_protocol_context, buf);
514  }
515  }
516  break;
517  }
518 }
519 
520 int ff_sdp_parse(AVFormatContext *s, const char *content)
521 {
522  RTSPState *rt = s->priv_data;
523  const char *p;
524  int letter;
525  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
526  * contain long SDP lines containing complete ASF Headers (several
527  * kB) or arrays of MDPR (RM stream descriptor) headers plus
528  * "rulebooks" describing their properties. Therefore, the SDP line
529  * buffer is large.
530  *
531  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
532  * in rtpdec_xiph.c. */
533  char buf[16384], *q;
534  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
535 
536  p = content;
537  for (;;) {
538  p += strspn(p, SPACE_CHARS);
539  letter = *p;
540  if (letter == '\0')
541  break;
542  p++;
543  if (*p != '=')
544  goto next_line;
545  p++;
546  /* get the content */
547  q = buf;
548  while (*p != '\n' && *p != '\r' && *p != '\0') {
549  if ((q - buf) < sizeof(buf) - 1)
550  *q++ = *p;
551  p++;
552  }
553  *q = '\0';
554  sdp_parse_line(s, s1, letter, buf);
555  next_line:
556  while (*p != '\n' && *p != '\0')
557  p++;
558  if (*p == '\n')
559  p++;
560  }
561  rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
562  if (!rt->p) return AVERROR(ENOMEM);
563  return 0;
564 }
565 #endif /* CONFIG_RTPDEC */
566 
568 {
569  RTSPState *rt = s->priv_data;
570  int i;
571 
572  for (i = 0; i < rt->nb_rtsp_streams; i++) {
573  RTSPStream *rtsp_st = rt->rtsp_streams[i];
574  if (!rtsp_st)
575  continue;
576  if (rtsp_st->transport_priv) {
577  if (s->oformat) {
578  AVFormatContext *rtpctx = rtsp_st->transport_priv;
579  av_write_trailer(rtpctx);
581  uint8_t *ptr;
582  avio_close_dyn_buf(rtpctx->pb, &ptr);
583  av_free(ptr);
584  } else {
585  avio_close(rtpctx->pb);
586  }
587  avformat_free_context(rtpctx);
588  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
590  else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
592  }
593  rtsp_st->transport_priv = NULL;
594  if (rtsp_st->rtp_handle)
595  ffurl_close(rtsp_st->rtp_handle);
596  rtsp_st->rtp_handle = NULL;
597  }
598 }
599 
600 /* close and free RTSP streams */
602 {
603  RTSPState *rt = s->priv_data;
604  int i;
605  RTSPStream *rtsp_st;
606 
608  for (i = 0; i < rt->nb_rtsp_streams; i++) {
609  rtsp_st = rt->rtsp_streams[i];
610  if (rtsp_st) {
611  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
612  rtsp_st->dynamic_handler->free(
613  rtsp_st->dynamic_protocol_context);
614  av_free(rtsp_st);
615  }
616  }
617  av_free(rt->rtsp_streams);
618  if (rt->asf_ctx) {
620  }
621  if (rt->ts && CONFIG_RTPDEC)
623  av_free(rt->p);
624  av_free(rt->recvbuf);
625 }
626 
628 {
629  RTSPState *rt = s->priv_data;
630  AVStream *st = NULL;
631  int reordering_queue_size = rt->reordering_queue_size;
632  if (reordering_queue_size < 0) {
634  reordering_queue_size = 0;
635  else
636  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
637  }
638 
639  /* open the RTP context */
640  if (rtsp_st->stream_index >= 0)
641  st = s->streams[rtsp_st->stream_index];
642  if (!st)
644 
645  if (s->oformat && CONFIG_RTSP_MUXER) {
646  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
647  rtsp_st->rtp_handle,
649  rtsp_st->stream_index);
650  /* Ownership of rtp_handle is passed to the rtp mux context */
651  rtsp_st->rtp_handle = NULL;
652  if (ret < 0)
653  return ret;
654  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
655  return 0; // Don't need to open any parser here
656  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
657  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
658  rtsp_st->dynamic_protocol_context,
659  rtsp_st->dynamic_handler);
660  else if (CONFIG_RTPDEC)
661  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
662  rtsp_st->sdp_payload_type,
663  reordering_queue_size);
664 
665  if (!rtsp_st->transport_priv) {
666  return AVERROR(ENOMEM);
667  } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
668  if (rtsp_st->dynamic_handler) {
670  rtsp_st->dynamic_protocol_context,
671  rtsp_st->dynamic_handler);
672  }
673  if (rtsp_st->crypto_suite[0])
675  rtsp_st->crypto_suite,
676  rtsp_st->crypto_params);
677  }
678 
679  return 0;
680 }
681 
682 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
683 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
684 {
685  const char *q;
686  char *p;
687  int v;
688 
689  q = *pp;
690  q += strspn(q, SPACE_CHARS);
691  v = strtol(q, &p, 10);
692  if (*p == '-') {
693  p++;
694  *min_ptr = v;
695  v = strtol(p, &p, 10);
696  *max_ptr = v;
697  } else {
698  *min_ptr = v;
699  *max_ptr = v;
700  }
701  *pp = p;
702 }
703 
704 /* XXX: only one transport specification is parsed */
705 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
706 {
707  char transport_protocol[16];
708  char profile[16];
709  char lower_transport[16];
710  char parameter[16];
712  char buf[256];
713 
714  reply->nb_transports = 0;
715 
716  for (;;) {
717  p += strspn(p, SPACE_CHARS);
718  if (*p == '\0')
719  break;
720 
721  th = &reply->transports[reply->nb_transports];
722 
723  get_word_sep(transport_protocol, sizeof(transport_protocol),
724  "/", &p);
725  if (!av_strcasecmp (transport_protocol, "rtp")) {
726  get_word_sep(profile, sizeof(profile), "/;,", &p);
727  lower_transport[0] = '\0';
728  /* rtp/avp/<protocol> */
729  if (*p == '/') {
730  get_word_sep(lower_transport, sizeof(lower_transport),
731  ";,", &p);
732  }
734  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
735  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
736  /* x-pn-tng/<protocol> */
737  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
738  profile[0] = '\0';
740  } else if (!av_strcasecmp(transport_protocol, "raw")) {
741  get_word_sep(profile, sizeof(profile), "/;,", &p);
742  lower_transport[0] = '\0';
743  /* raw/raw/<protocol> */
744  if (*p == '/') {
745  get_word_sep(lower_transport, sizeof(lower_transport),
746  ";,", &p);
747  }
749  }
750  if (!av_strcasecmp(lower_transport, "TCP"))
752  else
754 
755  if (*p == ';')
756  p++;
757  /* get each parameter */
758  while (*p != '\0' && *p != ',') {
759  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
760  if (!strcmp(parameter, "port")) {
761  if (*p == '=') {
762  p++;
763  rtsp_parse_range(&th->port_min, &th->port_max, &p);
764  }
765  } else if (!strcmp(parameter, "client_port")) {
766  if (*p == '=') {
767  p++;
768  rtsp_parse_range(&th->client_port_min,
769  &th->client_port_max, &p);
770  }
771  } else if (!strcmp(parameter, "server_port")) {
772  if (*p == '=') {
773  p++;
774  rtsp_parse_range(&th->server_port_min,
775  &th->server_port_max, &p);
776  }
777  } else if (!strcmp(parameter, "interleaved")) {
778  if (*p == '=') {
779  p++;
780  rtsp_parse_range(&th->interleaved_min,
781  &th->interleaved_max, &p);
782  }
783  } else if (!strcmp(parameter, "multicast")) {
786  } else if (!strcmp(parameter, "ttl")) {
787  if (*p == '=') {
788  char *end;
789  p++;
790  th->ttl = strtol(p, &end, 10);
791  p = end;
792  }
793  } else if (!strcmp(parameter, "destination")) {
794  if (*p == '=') {
795  p++;
796  get_word_sep(buf, sizeof(buf), ";,", &p);
797  get_sockaddr(buf, &th->destination);
798  }
799  } else if (!strcmp(parameter, "source")) {
800  if (*p == '=') {
801  p++;
802  get_word_sep(buf, sizeof(buf), ";,", &p);
803  av_strlcpy(th->source, buf, sizeof(th->source));
804  }
805  } else if (!strcmp(parameter, "mode")) {
806  if (*p == '=') {
807  p++;
808  get_word_sep(buf, sizeof(buf), ";, ", &p);
809  if (!strcmp(buf, "record") ||
810  !strcmp(buf, "receive"))
811  th->mode_record = 1;
812  }
813  }
814 
815  while (*p != ';' && *p != '\0' && *p != ',')
816  p++;
817  if (*p == ';')
818  p++;
819  }
820  if (*p == ',')
821  p++;
822 
823  reply->nb_transports++;
824  }
825 }
826 
827 static void handle_rtp_info(RTSPState *rt, const char *url,
828  uint32_t seq, uint32_t rtptime)
829 {
830  int i;
831  if (!rtptime || !url[0])
832  return;
833  if (rt->transport != RTSP_TRANSPORT_RTP)
834  return;
835  for (i = 0; i < rt->nb_rtsp_streams; i++) {
836  RTSPStream *rtsp_st = rt->rtsp_streams[i];
837  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
838  if (!rtpctx)
839  continue;
840  if (!strcmp(rtsp_st->control_url, url)) {
841  rtpctx->base_timestamp = rtptime;
842  break;
843  }
844  }
845 }
846 
847 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
848 {
849  int read = 0;
850  char key[20], value[1024], url[1024] = "";
851  uint32_t seq = 0, rtptime = 0;
852 
853  for (;;) {
854  p += strspn(p, SPACE_CHARS);
855  if (!*p)
856  break;
857  get_word_sep(key, sizeof(key), "=", &p);
858  if (*p != '=')
859  break;
860  p++;
861  get_word_sep(value, sizeof(value), ";, ", &p);
862  read++;
863  if (!strcmp(key, "url"))
864  av_strlcpy(url, value, sizeof(url));
865  else if (!strcmp(key, "seq"))
866  seq = strtoul(value, NULL, 10);
867  else if (!strcmp(key, "rtptime"))
868  rtptime = strtoul(value, NULL, 10);
869  if (*p == ',') {
870  handle_rtp_info(rt, url, seq, rtptime);
871  url[0] = '\0';
872  seq = rtptime = 0;
873  read = 0;
874  }
875  if (*p)
876  p++;
877  }
878  if (read > 0)
879  handle_rtp_info(rt, url, seq, rtptime);
880 }
881 
882 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
883  RTSPState *rt, const char *method)
884 {
885  const char *p;
886 
887  /* NOTE: we do case independent match for broken servers */
888  p = buf;
889  if (av_stristart(p, "Session:", &p)) {
890  int t;
891  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
892  if (av_stristart(p, ";timeout=", &p) &&
893  (t = strtol(p, NULL, 10)) > 0) {
894  reply->timeout = t;
895  }
896  } else if (av_stristart(p, "Content-Length:", &p)) {
897  reply->content_length = strtol(p, NULL, 10);
898  } else if (av_stristart(p, "Transport:", &p)) {
899  rtsp_parse_transport(reply, p);
900  } else if (av_stristart(p, "CSeq:", &p)) {
901  reply->seq = strtol(p, NULL, 10);
902  } else if (av_stristart(p, "Range:", &p)) {
903  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
904  } else if (av_stristart(p, "RealChallenge1:", &p)) {
905  p += strspn(p, SPACE_CHARS);
906  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
907  } else if (av_stristart(p, "Server:", &p)) {
908  p += strspn(p, SPACE_CHARS);
909  av_strlcpy(reply->server, p, sizeof(reply->server));
910  } else if (av_stristart(p, "Notice:", &p) ||
911  av_stristart(p, "X-Notice:", &p)) {
912  reply->notice = strtol(p, NULL, 10);
913  } else if (av_stristart(p, "Location:", &p)) {
914  p += strspn(p, SPACE_CHARS);
915  av_strlcpy(reply->location, p , sizeof(reply->location));
916  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
917  p += strspn(p, SPACE_CHARS);
918  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
919  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
920  p += strspn(p, SPACE_CHARS);
921  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
922  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
923  p += strspn(p, SPACE_CHARS);
924  if (method && !strcmp(method, "DESCRIBE"))
925  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
926  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
927  p += strspn(p, SPACE_CHARS);
928  if (method && !strcmp(method, "PLAY"))
929  rtsp_parse_rtp_info(rt, p);
930  } else if (av_stristart(p, "Public:", &p) && rt) {
931  if (strstr(p, "GET_PARAMETER") &&
932  method && !strcmp(method, "OPTIONS"))
933  rt->get_parameter_supported = 1;
934  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
935  p += strspn(p, SPACE_CHARS);
936  rt->accept_dynamic_rate = atoi(p);
937  } else if (av_stristart(p, "Content-Type:", &p)) {
938  p += strspn(p, SPACE_CHARS);
939  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
940  }
941 }
942 
943 /* skip a RTP/TCP interleaved packet */
945 {
946  RTSPState *rt = s->priv_data;
947  int ret, len, len1;
948  uint8_t buf[1024];
949 
950  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
951  if (ret != 3)
952  return;
953  len = AV_RB16(buf + 1);
954 
955  av_dlog(s, "skipping RTP packet len=%d\n", len);
956 
957  /* skip payload */
958  while (len > 0) {
959  len1 = len;
960  if (len1 > sizeof(buf))
961  len1 = sizeof(buf);
962  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
963  if (ret != len1)
964  return;
965  len -= len1;
966  }
967 }
968 
970  unsigned char **content_ptr,
971  int return_on_interleaved_data, const char *method)
972 {
973  RTSPState *rt = s->priv_data;
974  char buf[4096], buf1[1024], *q;
975  unsigned char ch;
976  const char *p;
977  int ret, content_length, line_count = 0, request = 0;
978  unsigned char *content = NULL;
979 
980 start:
981  line_count = 0;
982  request = 0;
983  content = NULL;
984  memset(reply, 0, sizeof(*reply));
985 
986  /* parse reply (XXX: use buffers) */
987  rt->last_reply[0] = '\0';
988  for (;;) {
989  q = buf;
990  for (;;) {
991  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
992  av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
993  if (ret != 1)
994  return AVERROR_EOF;
995  if (ch == '\n')
996  break;
997  if (ch == '$') {
998  /* XXX: only parse it if first char on line ? */
999  if (return_on_interleaved_data) {
1000  return 1;
1001  } else
1003  } else if (ch != '\r') {
1004  if ((q - buf) < sizeof(buf) - 1)
1005  *q++ = ch;
1006  }
1007  }
1008  *q = '\0';
1009 
1010  av_dlog(s, "line='%s'\n", buf);
1011 
1012  /* test if last line */
1013  if (buf[0] == '\0')
1014  break;
1015  p = buf;
1016  if (line_count == 0) {
1017  /* get reply code */
1018  get_word(buf1, sizeof(buf1), &p);
1019  if (!strncmp(buf1, "RTSP/", 5)) {
1020  get_word(buf1, sizeof(buf1), &p);
1021  reply->status_code = atoi(buf1);
1022  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1023  } else {
1024  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1025  get_word(buf1, sizeof(buf1), &p); // object
1026  request = 1;
1027  }
1028  } else {
1029  ff_rtsp_parse_line(reply, p, rt, method);
1030  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1031  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1032  }
1033  line_count++;
1034  }
1035 
1036  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1037  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1038 
1039  content_length = reply->content_length;
1040  if (content_length > 0) {
1041  /* leave some room for a trailing '\0' (useful for simple parsing) */
1042  content = av_malloc(content_length + 1);
1043  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1044  content[content_length] = '\0';
1045  }
1046  if (content_ptr)
1047  *content_ptr = content;
1048  else
1049  av_free(content);
1050 
1051  if (request) {
1052  char buf[1024];
1053  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1054  const char* ptr = buf;
1055 
1056  if (!strcmp(reply->reason, "OPTIONS")) {
1057  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1058  if (reply->seq)
1059  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1060  if (reply->session_id[0])
1061  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1062  reply->session_id);
1063  } else {
1064  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1065  }
1066  av_strlcat(buf, "\r\n", sizeof(buf));
1067 
1068  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1069  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1070  ptr = base64buf;
1071  }
1072  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1073 
1074  rt->last_cmd_time = av_gettime();
1075  /* Even if the request from the server had data, it is not the data
1076  * that the caller wants or expects. The memory could also be leaked
1077  * if the actual following reply has content data. */
1078  if (content_ptr)
1079  av_freep(content_ptr);
1080  /* If method is set, this is called from ff_rtsp_send_cmd,
1081  * where a reply to exactly this request is awaited. For
1082  * callers from within packet receiving, we just want to
1083  * return to the caller and go back to receiving packets. */
1084  if (method)
1085  goto start;
1086  return 0;
1087  }
1088 
1089  if (rt->seq != reply->seq) {
1090  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1091  rt->seq, reply->seq);
1092  }
1093 
1094  /* EOS */
1095  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1096  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1097  reply->notice == 2306 /* Continuous Feed Terminated */) {
1098  rt->state = RTSP_STATE_IDLE;
1099  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1100  return AVERROR(EIO); /* data or server error */
1101  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1102  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1103  return AVERROR(EPERM);
1104 
1105  return 0;
1106 }
1107 
1108 /**
1109  * Send a command to the RTSP server without waiting for the reply.
1110  *
1111  * @param s RTSP (de)muxer context
1112  * @param method the method for the request
1113  * @param url the target url for the request
1114  * @param headers extra header lines to include in the request
1115  * @param send_content if non-null, the data to send as request body content
1116  * @param send_content_length the length of the send_content data, or 0 if
1117  * send_content is null
1118  *
1119  * @return zero if success, nonzero otherwise
1120  */
1121 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1122  const char *method, const char *url,
1123  const char *headers,
1124  const unsigned char *send_content,
1125  int send_content_length)
1126 {
1127  RTSPState *rt = s->priv_data;
1128  char buf[4096], *out_buf;
1129  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1130 
1131  /* Add in RTSP headers */
1132  out_buf = buf;
1133  rt->seq++;
1134  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1135  if (headers)
1136  av_strlcat(buf, headers, sizeof(buf));
1137  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1138  if (rt->session_id[0] != '\0' && (!headers ||
1139  !strstr(headers, "\nIf-Match:"))) {
1140  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1141  }
1142  if (rt->auth[0]) {
1143  char *str = ff_http_auth_create_response(&rt->auth_state,
1144  rt->auth, url, method);
1145  if (str)
1146  av_strlcat(buf, str, sizeof(buf));
1147  av_free(str);
1148  }
1149  if (send_content_length > 0 && send_content)
1150  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1151  av_strlcat(buf, "\r\n", sizeof(buf));
1152 
1153  /* base64 encode rtsp if tunneling */
1154  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1155  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1156  out_buf = base64buf;
1157  }
1158 
1159  av_dlog(s, "Sending:\n%s--\n", buf);
1160 
1161  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1162  if (send_content_length > 0 && send_content) {
1163  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1164  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1165  "with content data not supported\n");
1166  return AVERROR_PATCHWELCOME;
1167  }
1168  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1169  }
1170  rt->last_cmd_time = av_gettime();
1171 
1172  return 0;
1173 }
1174 
1175 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1176  const char *url, const char *headers)
1177 {
1178  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1179 }
1180 
1181 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1182  const char *headers, RTSPMessageHeader *reply,
1183  unsigned char **content_ptr)
1184 {
1185  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1186  content_ptr, NULL, 0);
1187 }
1188 
1190  const char *method, const char *url,
1191  const char *header,
1192  RTSPMessageHeader *reply,
1193  unsigned char **content_ptr,
1194  const unsigned char *send_content,
1195  int send_content_length)
1196 {
1197  RTSPState *rt = s->priv_data;
1198  HTTPAuthType cur_auth_type;
1199  int ret, attempts = 0;
1200 
1201 retry:
1202  cur_auth_type = rt->auth_state.auth_type;
1203  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1204  send_content,
1205  send_content_length)))
1206  return ret;
1207 
1208  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1209  return ret;
1210  attempts++;
1211 
1212  if (reply->status_code == 401 &&
1213  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1214  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1215  goto retry;
1216 
1217  if (reply->status_code > 400){
1218  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1219  method,
1220  reply->status_code,
1221  reply->reason);
1222  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1223  }
1224 
1225  return 0;
1226 }
1227 
1228 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1229  int lower_transport, const char *real_challenge)
1230 {
1231  RTSPState *rt = s->priv_data;
1232  int rtx = 0, j, i, err, interleave = 0, port_off;
1233  RTSPStream *rtsp_st;
1234  RTSPMessageHeader reply1, *reply = &reply1;
1235  char cmd[2048];
1236  const char *trans_pref;
1237 
1238  if (rt->transport == RTSP_TRANSPORT_RDT)
1239  trans_pref = "x-pn-tng";
1240  else if (rt->transport == RTSP_TRANSPORT_RAW)
1241  trans_pref = "RAW/RAW";
1242  else
1243  trans_pref = "RTP/AVP";
1244 
1245  /* default timeout: 1 minute */
1246  rt->timeout = 60;
1247 
1248  /* Choose a random starting offset within the first half of the
1249  * port range, to allow for a number of ports to try even if the offset
1250  * happens to be at the end of the random range. */
1251  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1252  /* even random offset */
1253  port_off -= port_off & 0x01;
1254 
1255  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1256  char transport[2048];
1257 
1258  /*
1259  * WMS serves all UDP data over a single connection, the RTX, which
1260  * isn't necessarily the first in the SDP but has to be the first
1261  * to be set up, else the second/third SETUP will fail with a 461.
1262  */
1263  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1264  rt->server_type == RTSP_SERVER_WMS) {
1265  if (i == 0) {
1266  /* rtx first */
1267  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1268  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1269  if (len >= 4 &&
1270  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1271  "/rtx"))
1272  break;
1273  }
1274  if (rtx == rt->nb_rtsp_streams)
1275  return -1; /* no RTX found */
1276  rtsp_st = rt->rtsp_streams[rtx];
1277  } else
1278  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1279  } else
1280  rtsp_st = rt->rtsp_streams[i];
1281 
1282  /* RTP/UDP */
1283  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1284  char buf[256];
1285 
1286  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1287  port = reply->transports[0].client_port_min;
1288  goto have_port;
1289  }
1290 
1291  /* first try in specified port range */
1292  while (j <= rt->rtp_port_max) {
1293  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1294  "?localport=%d", j);
1295  /* we will use two ports per rtp stream (rtp and rtcp) */
1296  j += 2;
1297  if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1298  &s->interrupt_callback, NULL))
1299  goto rtp_opened;
1300  }
1301  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1302  err = AVERROR(EIO);
1303  goto fail;
1304 
1305  rtp_opened:
1306  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1307  have_port:
1308  snprintf(transport, sizeof(transport) - 1,
1309  "%s/UDP;", trans_pref);
1310  if (rt->server_type != RTSP_SERVER_REAL)
1311  av_strlcat(transport, "unicast;", sizeof(transport));
1312  av_strlcatf(transport, sizeof(transport),
1313  "client_port=%d", port);
1314  if (rt->transport == RTSP_TRANSPORT_RTP &&
1315  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1316  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1317  }
1318 
1319  /* RTP/TCP */
1320  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1321  /* For WMS streams, the application streams are only used for
1322  * UDP. When trying to set it up for TCP streams, the server
1323  * will return an error. Therefore, we skip those streams. */
1324  if (rt->server_type == RTSP_SERVER_WMS &&
1325  (rtsp_st->stream_index < 0 ||
1326  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1328  continue;
1329  snprintf(transport, sizeof(transport) - 1,
1330  "%s/TCP;", trans_pref);
1331  if (rt->transport != RTSP_TRANSPORT_RDT)
1332  av_strlcat(transport, "unicast;", sizeof(transport));
1333  av_strlcatf(transport, sizeof(transport),
1334  "interleaved=%d-%d",
1335  interleave, interleave + 1);
1336  interleave += 2;
1337  }
1338 
1339  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1340  snprintf(transport, sizeof(transport) - 1,
1341  "%s/UDP;multicast", trans_pref);
1342  }
1343  if (s->oformat) {
1344  av_strlcat(transport, ";mode=record", sizeof(transport));
1345  } else if (rt->server_type == RTSP_SERVER_REAL ||
1347  av_strlcat(transport, ";mode=play", sizeof(transport));
1348  snprintf(cmd, sizeof(cmd),
1349  "Transport: %s\r\n",
1350  transport);
1351  if (rt->accept_dynamic_rate)
1352  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1353  if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1354  char real_res[41], real_csum[9];
1355  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1356  real_challenge);
1357  av_strlcatf(cmd, sizeof(cmd),
1358  "If-Match: %s\r\n"
1359  "RealChallenge2: %s, sd=%s\r\n",
1360  rt->session_id, real_res, real_csum);
1361  }
1362  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1363  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1364  err = 1;
1365  goto fail;
1366  } else if (reply->status_code != RTSP_STATUS_OK ||
1367  reply->nb_transports != 1) {
1368  err = AVERROR_INVALIDDATA;
1369  goto fail;
1370  }
1371 
1372  /* XXX: same protocol for all streams is required */
1373  if (i > 0) {
1374  if (reply->transports[0].lower_transport != rt->lower_transport ||
1375  reply->transports[0].transport != rt->transport) {
1376  err = AVERROR_INVALIDDATA;
1377  goto fail;
1378  }
1379  } else {
1380  rt->lower_transport = reply->transports[0].lower_transport;
1381  rt->transport = reply->transports[0].transport;
1382  }
1383 
1384  /* Fail if the server responded with another lower transport mode
1385  * than what we requested. */
1386  if (reply->transports[0].lower_transport != lower_transport) {
1387  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1388  err = AVERROR_INVALIDDATA;
1389  goto fail;
1390  }
1391 
1392  switch(reply->transports[0].lower_transport) {
1394  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1395  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1396  break;
1397 
1398  case RTSP_LOWER_TRANSPORT_UDP: {
1399  char url[1024], options[30] = "";
1400 
1401  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1402  av_strlcpy(options, "?connect=1", sizeof(options));
1403  /* Use source address if specified */
1404  if (reply->transports[0].source[0]) {
1405  ff_url_join(url, sizeof(url), "rtp", NULL,
1406  reply->transports[0].source,
1407  reply->transports[0].server_port_min, "%s", options);
1408  } else {
1409  ff_url_join(url, sizeof(url), "rtp", NULL, host,
1410  reply->transports[0].server_port_min, "%s", options);
1411  }
1412  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1413  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1414  err = AVERROR_INVALIDDATA;
1415  goto fail;
1416  }
1417  /* Try to initialize the connection state in a
1418  * potential NAT router by sending dummy packets.
1419  * RTP/RTCP dummy packets are used for RDT, too.
1420  */
1421  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1422  CONFIG_RTPDEC)
1424  break;
1425  }
1427  char url[1024], namebuf[50], optbuf[20] = "";
1428  struct sockaddr_storage addr;
1429  int port, ttl;
1430 
1431  if (reply->transports[0].destination.ss_family) {
1432  addr = reply->transports[0].destination;
1433  port = reply->transports[0].port_min;
1434  ttl = reply->transports[0].ttl;
1435  } else {
1436  addr = rtsp_st->sdp_ip;
1437  port = rtsp_st->sdp_port;
1438  ttl = rtsp_st->sdp_ttl;
1439  }
1440  if (ttl > 0)
1441  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1442  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1443  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1444  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1445  port, "%s", optbuf);
1446  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1447  &s->interrupt_callback, NULL) < 0) {
1448  err = AVERROR_INVALIDDATA;
1449  goto fail;
1450  }
1451  break;
1452  }
1453  }
1454 
1455  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1456  goto fail;
1457  }
1458 
1459  if (rt->nb_rtsp_streams && reply->timeout > 0)
1460  rt->timeout = reply->timeout;
1461 
1462  if (rt->server_type == RTSP_SERVER_REAL)
1463  rt->need_subscription = 1;
1464 
1465  return 0;
1466 
1467 fail:
1468  ff_rtsp_undo_setup(s);
1469  return err;
1470 }
1471 
1473 {
1474  RTSPState *rt = s->priv_data;
1475  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1476  ffurl_close(rt->rtsp_hd);
1477  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1478 }
1479 
1481 {
1482  RTSPState *rt = s->priv_data;
1483  char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1484  int port, err, tcp_fd;
1485  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1486  int lower_transport_mask = 0;
1487  char real_challenge[64] = "";
1488  struct sockaddr_storage peer;
1489  socklen_t peer_len = sizeof(peer);
1490 
1491  if (rt->rtp_port_max < rt->rtp_port_min) {
1492  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1493  "than min port %d\n", rt->rtp_port_max,
1494  rt->rtp_port_min);
1495  return AVERROR(EINVAL);
1496  }
1497 
1498  if (!ff_network_init())
1499  return AVERROR(EIO);
1500 
1501  if (s->max_delay < 0) /* Not set by the caller */
1503 
1508  }
1509  /* Only pass through valid flags from here */
1511 
1512 redirect:
1513  lower_transport_mask = rt->lower_transport_mask;
1514  /* extract hostname and port */
1515  av_url_split(NULL, 0, auth, sizeof(auth),
1516  host, sizeof(host), &port, path, sizeof(path), s->filename);
1517  if (*auth) {
1518  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1519  }
1520  if (port < 0)
1521  port = RTSP_DEFAULT_PORT;
1522 
1523  if (!lower_transport_mask)
1524  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1525 
1526  if (s->oformat) {
1527  /* Only UDP or TCP - UDP multicast isn't supported. */
1528  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1529  (1 << RTSP_LOWER_TRANSPORT_TCP);
1530  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1531  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1532  "only UDP and TCP are supported for output.\n");
1533  err = AVERROR(EINVAL);
1534  goto fail;
1535  }
1536  }
1537 
1538  /* Construct the URI used in request; this is similar to s->filename,
1539  * but with authentication credentials removed and RTSP specific options
1540  * stripped out. */
1541  ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1542  host, port, "%s", path);
1543 
1544  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1545  /* set up initial handshake for tunneling */
1546  char httpname[1024];
1547  char sessioncookie[17];
1548  char headers[1024];
1549 
1550  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1551  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1553 
1554  /* GET requests */
1555  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1556  &s->interrupt_callback) < 0) {
1557  err = AVERROR(EIO);
1558  goto fail;
1559  }
1560 
1561  /* generate GET headers */
1562  snprintf(headers, sizeof(headers),
1563  "x-sessioncookie: %s\r\n"
1564  "Accept: application/x-rtsp-tunnelled\r\n"
1565  "Pragma: no-cache\r\n"
1566  "Cache-Control: no-cache\r\n",
1567  sessioncookie);
1568  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1569 
1570  /* complete the connection */
1571  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1572  err = AVERROR(EIO);
1573  goto fail;
1574  }
1575 
1576  /* POST requests */
1577  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1578  &s->interrupt_callback) < 0 ) {
1579  err = AVERROR(EIO);
1580  goto fail;
1581  }
1582 
1583  /* generate POST headers */
1584  snprintf(headers, sizeof(headers),
1585  "x-sessioncookie: %s\r\n"
1586  "Content-Type: application/x-rtsp-tunnelled\r\n"
1587  "Pragma: no-cache\r\n"
1588  "Cache-Control: no-cache\r\n"
1589  "Content-Length: 32767\r\n"
1590  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1591  sessioncookie);
1592  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1593  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1594 
1595  /* Initialize the authentication state for the POST session. The HTTP
1596  * protocol implementation doesn't properly handle multi-pass
1597  * authentication for POST requests, since it would require one of
1598  * the following:
1599  * - implementing Expect: 100-continue, which many HTTP servers
1600  * don't support anyway, even less the RTSP servers that do HTTP
1601  * tunneling
1602  * - sending the whole POST data until getting a 401 reply specifying
1603  * what authentication method to use, then resending all that data
1604  * - waiting for potential 401 replies directly after sending the
1605  * POST header (waiting for some unspecified time)
1606  * Therefore, we copy the full auth state, which works for both basic
1607  * and digest. (For digest, we would have to synchronize the nonce
1608  * count variable between the two sessions, if we'd do more requests
1609  * with the original session, though.)
1610  */
1612 
1613  /* complete the connection */
1614  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1615  err = AVERROR(EIO);
1616  goto fail;
1617  }
1618  } else {
1619  /* open the tcp connection */
1620  ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1621  "?timeout=%d", rt->stimeout);
1622  if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1623  &s->interrupt_callback, NULL) < 0) {
1624  err = AVERROR(EIO);
1625  goto fail;
1626  }
1627  rt->rtsp_hd_out = rt->rtsp_hd;
1628  }
1629  rt->seq = 0;
1630 
1631  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1632  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1633  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1634  NULL, 0, NI_NUMERICHOST);
1635  }
1636 
1637  /* request options supported by the server; this also detects server
1638  * type */
1639  for (rt->server_type = RTSP_SERVER_RTP;;) {
1640  cmd[0] = 0;
1641  if (rt->server_type == RTSP_SERVER_REAL)
1642  av_strlcat(cmd,
1643  /*
1644  * The following entries are required for proper
1645  * streaming from a Realmedia server. They are
1646  * interdependent in some way although we currently
1647  * don't quite understand how. Values were copied
1648  * from mplayer SVN r23589.
1649  * ClientChallenge is a 16-byte ID in hex
1650  * CompanyID is a 16-byte ID in base64
1651  */
1652  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1653  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1654  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1655  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1656  sizeof(cmd));
1657  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1658  if (reply->status_code != RTSP_STATUS_OK) {
1659  err = AVERROR_INVALIDDATA;
1660  goto fail;
1661  }
1662 
1663  /* detect server type if not standard-compliant RTP */
1664  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1666  continue;
1667  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1669  } else if (rt->server_type == RTSP_SERVER_REAL)
1670  strcpy(real_challenge, reply->real_challenge);
1671  break;
1672  }
1673 
1674  if (s->iformat && CONFIG_RTSP_DEMUXER)
1675  err = ff_rtsp_setup_input_streams(s, reply);
1676  else if (CONFIG_RTSP_MUXER)
1677  err = ff_rtsp_setup_output_streams(s, host);
1678  if (err)
1679  goto fail;
1680 
1681  do {
1682  int lower_transport = ff_log2_tab[lower_transport_mask &
1683  ~(lower_transport_mask - 1)];
1684 
1685  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1686  rt->server_type == RTSP_SERVER_REAL ?
1687  real_challenge : NULL);
1688  if (err < 0)
1689  goto fail;
1690  lower_transport_mask &= ~(1 << lower_transport);
1691  if (lower_transport_mask == 0 && err == 1) {
1692  err = AVERROR(EPROTONOSUPPORT);
1693  goto fail;
1694  }
1695  } while (err);
1696 
1697  rt->lower_transport_mask = lower_transport_mask;
1698  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1699  rt->state = RTSP_STATE_IDLE;
1700  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1701  return 0;
1702  fail:
1705  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1706  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1707  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1708  reply->status_code,
1709  s->filename);
1710  goto redirect;
1711  }
1712  ff_network_close();
1713  return err;
1714 }
1715 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1716 
1717 #if CONFIG_RTPDEC
1718 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1719  uint8_t *buf, int buf_size, int64_t wait_end)
1720 {
1721  RTSPState *rt = s->priv_data;
1722  RTSPStream *rtsp_st;
1723  int n, i, ret, tcp_fd, timeout_cnt = 0;
1724  int max_p = 0;
1725  struct pollfd *p = rt->p;
1726  int *fds = NULL, fdsnum, fdsidx;
1727 
1728  for (;;) {
1730  return AVERROR_EXIT;
1731  if (wait_end && wait_end - av_gettime() < 0)
1732  return AVERROR(EAGAIN);
1733  max_p = 0;
1734  if (rt->rtsp_hd) {
1735  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1736  p[max_p].fd = tcp_fd;
1737  p[max_p++].events = POLLIN;
1738  } else {
1739  tcp_fd = -1;
1740  }
1741  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1742  rtsp_st = rt->rtsp_streams[i];
1743  if (rtsp_st->rtp_handle) {
1744  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1745  &fds, &fdsnum)) {
1746  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1747  return ret;
1748  }
1749  if (fdsnum != 2) {
1750  av_log(s, AV_LOG_ERROR,
1751  "Number of fds %d not supported\n", fdsnum);
1752  return AVERROR_INVALIDDATA;
1753  }
1754  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1755  p[max_p].fd = fds[fdsidx];
1756  p[max_p++].events = POLLIN;
1757  }
1758  av_free(fds);
1759  }
1760  }
1761  n = poll(p, max_p, POLL_TIMEOUT_MS);
1762  if (n > 0) {
1763  int j = 1 - (tcp_fd == -1);
1764  timeout_cnt = 0;
1765  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1766  rtsp_st = rt->rtsp_streams[i];
1767  if (rtsp_st->rtp_handle) {
1768  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1769  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1770  if (ret > 0) {
1771  *prtsp_st = rtsp_st;
1772  return ret;
1773  }
1774  }
1775  j+=2;
1776  }
1777  }
1778 #if CONFIG_RTSP_DEMUXER
1779  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1780  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1781  if (rt->state == RTSP_STATE_STREAMING) {
1783  return AVERROR_EOF;
1784  else
1786  "Unable to answer to TEARDOWN\n");
1787  } else
1788  return 0;
1789  } else {
1790  RTSPMessageHeader reply;
1791  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1792  if (ret < 0)
1793  return ret;
1794  /* XXX: parse message */
1795  if (rt->state != RTSP_STATE_STREAMING)
1796  return 0;
1797  }
1798  }
1799 #endif
1800  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1801  return AVERROR(ETIMEDOUT);
1802  } else if (n < 0 && errno != EINTR)
1803  return AVERROR(errno);
1804  }
1805 }
1806 
1807 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1808  const uint8_t *buf, int len)
1809 {
1810  RTSPState *rt = s->priv_data;
1811  int i;
1812  if (len < 0)
1813  return len;
1814  if (rt->nb_rtsp_streams == 1) {
1815  *rtsp_st = rt->rtsp_streams[0];
1816  return len;
1817  }
1818  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1819  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1820  int no_ssrc = 0;
1821  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1822  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1823  if (!rtpctx)
1824  continue;
1825  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1826  *rtsp_st = rt->rtsp_streams[i];
1827  return len;
1828  }
1829  if (!rtpctx->ssrc)
1830  no_ssrc = 1;
1831  }
1832  if (no_ssrc) {
1834  "Unable to pick stream for packet - SSRC not known for "
1835  "all streams\n");
1836  return AVERROR(EAGAIN);
1837  }
1838  } else {
1839  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1840  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1841  *rtsp_st = rt->rtsp_streams[i];
1842  return len;
1843  }
1844  }
1845  }
1846  }
1847  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1848  return AVERROR(EAGAIN);
1849 }
1850 
1852 {
1853  RTSPState *rt = s->priv_data;
1854  int ret, len;
1855  RTSPStream *rtsp_st, *first_queue_st = NULL;
1856  int64_t wait_end = 0;
1857 
1858  if (rt->nb_byes == rt->nb_rtsp_streams)
1859  return AVERROR_EOF;
1860 
1861  /* get next frames from the same RTP packet */
1862  if (rt->cur_transport_priv) {
1863  if (rt->transport == RTSP_TRANSPORT_RDT) {
1864  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1865  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1866  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1867  } else if (rt->ts && CONFIG_RTPDEC) {
1868  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1869  if (ret >= 0) {
1870  rt->recvbuf_pos += ret;
1871  ret = rt->recvbuf_pos < rt->recvbuf_len;
1872  }
1873  } else
1874  ret = -1;
1875  if (ret == 0) {
1876  rt->cur_transport_priv = NULL;
1877  return 0;
1878  } else if (ret == 1) {
1879  return 0;
1880  } else
1881  rt->cur_transport_priv = NULL;
1882  }
1883 
1884 redo:
1885  if (rt->transport == RTSP_TRANSPORT_RTP) {
1886  int i;
1887  int64_t first_queue_time = 0;
1888  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1889  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1890  int64_t queue_time;
1891  if (!rtpctx)
1892  continue;
1893  queue_time = ff_rtp_queued_packet_time(rtpctx);
1894  if (queue_time && (queue_time - first_queue_time < 0 ||
1895  !first_queue_time)) {
1896  first_queue_time = queue_time;
1897  first_queue_st = rt->rtsp_streams[i];
1898  }
1899  }
1900  if (first_queue_time) {
1901  wait_end = first_queue_time + s->max_delay;
1902  } else {
1903  wait_end = 0;
1904  first_queue_st = NULL;
1905  }
1906  }
1907 
1908  /* read next RTP packet */
1909  if (!rt->recvbuf) {
1911  if (!rt->recvbuf)
1912  return AVERROR(ENOMEM);
1913  }
1914 
1915  switch(rt->lower_transport) {
1916  default:
1917 #if CONFIG_RTSP_DEMUXER
1919  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1920  break;
1921 #endif
1924  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1925  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1926  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1927  break;
1929  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1930  wait_end && wait_end < av_gettime())
1931  len = AVERROR(EAGAIN);
1932  else
1933  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1934  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1935  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1936  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1937  break;
1938  }
1939  if (len == AVERROR(EAGAIN) && first_queue_st &&
1940  rt->transport == RTSP_TRANSPORT_RTP) {
1941  rtsp_st = first_queue_st;
1942  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1943  goto end;
1944  }
1945  if (len < 0)
1946  return len;
1947  if (len == 0)
1948  return AVERROR_EOF;
1949  if (rt->transport == RTSP_TRANSPORT_RDT) {
1950  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1951  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1952  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1953  if (rtsp_st->feedback) {
1954  AVIOContext *pb = NULL;
1956  pb = s->pb;
1957  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1958  }
1959  if (ret < 0) {
1960  /* Either bad packet, or a RTCP packet. Check if the
1961  * first_rtcp_ntp_time field was initialized. */
1962  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1963  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1964  /* first_rtcp_ntp_time has been initialized for this stream,
1965  * copy the same value to all other uninitialized streams,
1966  * in order to map their timestamp origin to the same ntp time
1967  * as this one. */
1968  int i;
1969  AVStream *st = NULL;
1970  if (rtsp_st->stream_index >= 0)
1971  st = s->streams[rtsp_st->stream_index];
1972  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1974  AVStream *st2 = NULL;
1975  if (rt->rtsp_streams[i]->stream_index >= 0)
1976  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1977  if (rtpctx2 && st && st2 &&
1978  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1979  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1980  rtpctx2->rtcp_ts_offset = av_rescale_q(
1981  rtpctx->rtcp_ts_offset, st->time_base,
1982  st2->time_base);
1983  }
1984  }
1985  }
1986  if (ret == -RTCP_BYE) {
1987  rt->nb_byes++;
1988 
1989  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1990  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1991 
1992  if (rt->nb_byes == rt->nb_rtsp_streams)
1993  return AVERROR_EOF;
1994  }
1995  }
1996  } else if (rt->ts && CONFIG_RTPDEC) {
1997  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1998  if (ret >= 0) {
1999  if (ret < len) {
2000  rt->recvbuf_len = len;
2001  rt->recvbuf_pos = ret;
2002  rt->cur_transport_priv = rt->ts;
2003  return 1;
2004  } else {
2005  ret = 0;
2006  }
2007  }
2008  } else {
2009  return AVERROR_INVALIDDATA;
2010  }
2011 end:
2012  if (ret < 0)
2013  goto redo;
2014  if (ret == 1)
2015  /* more packets may follow, so we save the RTP context */
2016  rt->cur_transport_priv = rtsp_st->transport_priv;
2017 
2018  return ret;
2019 }
2020 #endif /* CONFIG_RTPDEC */
2021 
2022 #if CONFIG_SDP_DEMUXER
2023 static int sdp_probe(AVProbeData *p1)
2024 {
2025  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2026 
2027  /* we look for a line beginning "c=IN IP" */
2028  while (p < p_end && *p != '\0') {
2029  if (p + sizeof("c=IN IP") - 1 < p_end &&
2030  av_strstart(p, "c=IN IP", NULL))
2031  return AVPROBE_SCORE_EXTENSION;
2032 
2033  while (p < p_end - 1 && *p != '\n') p++;
2034  if (++p >= p_end)
2035  break;
2036  if (*p == '\r')
2037  p++;
2038  }
2039  return 0;
2040 }
2041 
2042 static int sdp_read_header(AVFormatContext *s)
2043 {
2044  RTSPState *rt = s->priv_data;
2045  RTSPStream *rtsp_st;
2046  int size, i, err;
2047  char *content;
2048  char url[1024];
2049 
2050  if (!ff_network_init())
2051  return AVERROR(EIO);
2052 
2053  if (s->max_delay < 0) /* Not set by the caller */
2055  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2057 
2058  /* read the whole sdp file */
2059  /* XXX: better loading */
2060  content = av_malloc(SDP_MAX_SIZE);
2061  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2062  if (size <= 0) {
2063  av_free(content);
2064  return AVERROR_INVALIDDATA;
2065  }
2066  content[size] ='\0';
2067 
2068  err = ff_sdp_parse(s, content);
2069  av_free(content);
2070  if (err) goto fail;
2071 
2072  /* open each RTP stream */
2073  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074  char namebuf[50];
2075  rtsp_st = rt->rtsp_streams[i];
2076 
2077  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2078  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2079  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2080  ff_url_join(url, sizeof(url), "rtp", NULL,
2081  namebuf, rtsp_st->sdp_port,
2082  "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2083  rtsp_st->sdp_ttl,
2084  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2085  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2086  &s->interrupt_callback, NULL) < 0) {
2087  err = AVERROR_INVALIDDATA;
2088  goto fail;
2089  }
2090  }
2091  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2092  goto fail;
2093  }
2094  return 0;
2095 fail:
2097  ff_network_close();
2098  return err;
2099 }
2100 
2101 static int sdp_read_close(AVFormatContext *s)
2102 {
2104  ff_network_close();
2105  return 0;
2106 }
2107 
2108 static const AVClass sdp_demuxer_class = {
2109  .class_name = "SDP demuxer",
2110  .item_name = av_default_item_name,
2111  .option = sdp_options,
2112  .version = LIBAVUTIL_VERSION_INT,
2113 };
2114 
2115 AVInputFormat ff_sdp_demuxer = {
2116  .name = "sdp",
2117  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2118  .priv_data_size = sizeof(RTSPState),
2119  .read_probe = sdp_probe,
2120  .read_header = sdp_read_header,
2122  .read_close = sdp_read_close,
2123  .priv_class = &sdp_demuxer_class,
2124 };
2125 #endif /* CONFIG_SDP_DEMUXER */
2126 
2127 #if CONFIG_RTP_DEMUXER
2128 static int rtp_probe(AVProbeData *p)
2129 {
2130  if (av_strstart(p->filename, "rtp:", NULL))
2131  return AVPROBE_SCORE_MAX;
2132  return 0;
2133 }
2134 
2135 static int rtp_read_header(AVFormatContext *s)
2136 {
2137  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2138  char host[500], sdp[500];
2139  int ret, port;
2140  URLContext* in = NULL;
2141  int payload_type;
2142  AVCodecContext codec = { 0 };
2143  struct sockaddr_storage addr;
2144  AVIOContext pb;
2145  socklen_t addrlen = sizeof(addr);
2146  RTSPState *rt = s->priv_data;
2147 
2148  if (!ff_network_init())
2149  return AVERROR(EIO);
2150 
2151  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2152  &s->interrupt_callback, NULL);
2153  if (ret)
2154  goto fail;
2155 
2156  while (1) {
2157  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2158  if (ret == AVERROR(EAGAIN))
2159  continue;
2160  if (ret < 0)
2161  goto fail;
2162  if (ret < 12) {
2163  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2164  continue;
2165  }
2166 
2167  if ((recvbuf[0] & 0xc0) != 0x80) {
2168  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2169  "received\n");
2170  continue;
2171  }
2172 
2173  if (RTP_PT_IS_RTCP(recvbuf[1]))
2174  continue;
2175 
2176  payload_type = recvbuf[1] & 0x7f;
2177  break;
2178  }
2179  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2180  ffurl_close(in);
2181  in = NULL;
2182 
2183  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2184  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2185  "without an SDP file describing it\n",
2186  payload_type);
2187  goto fail;
2188  }
2189  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2190  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2191  "properly you need an SDP file "
2192  "describing it\n");
2193  }
2194 
2195  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2196  NULL, 0, s->filename);
2197 
2198  snprintf(sdp, sizeof(sdp),
2199  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2200  addr.ss_family == AF_INET ? 4 : 6, host,
2201  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2202  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2203  port, payload_type);
2204  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2205 
2206  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2207  s->pb = &pb;
2208 
2209  /* sdp_read_header initializes this again */
2210  ff_network_close();
2211 
2212  rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2213 
2214  ret = sdp_read_header(s);
2215  s->pb = NULL;
2216  return ret;
2217 
2218 fail:
2219  if (in)
2220  ffurl_close(in);
2221  ff_network_close();
2222  return ret;
2223 }
2224 
2225 static const AVClass rtp_demuxer_class = {
2226  .class_name = "RTP demuxer",
2227  .item_name = av_default_item_name,
2228  .option = rtp_options,
2229  .version = LIBAVUTIL_VERSION_INT,
2230 };
2231 
2232 AVInputFormat ff_rtp_demuxer = {
2233  .name = "rtp",
2234  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2235  .priv_data_size = sizeof(RTSPState),
2236  .read_probe = rtp_probe,
2237  .read_header = rtp_read_header,
2239  .read_close = sdp_read_close,
2240  .flags = AVFMT_NOFILE,
2241  .priv_class = &rtp_demuxer_class,
2242 };
2243 #endif /* CONFIG_RTP_DEMUXER */