3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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.
12 * Libav 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.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
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
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
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" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { 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" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 #define COMMON_OPTS() \
76 { "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 }, \
77 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
82 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
89 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "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 },
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
101 { "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" },
102 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
107 static const AVOption rtp_options[] = {
108 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
114 static AVDictionary *map_to_opts(RTSPState *rt)
116 AVDictionary *opts = NULL;
119 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
120 av_dict_set(&opts, "buffer_size", buf, 0);
125 static void get_word_until_chars(char *buf, int buf_size,
126 const char *sep, const char **pp)
132 p += strspn(p, SPACE_CHARS);
134 while (!strchr(sep, *p) && *p != '\0') {
135 if ((q - buf) < buf_size - 1)
144 static void get_word_sep(char *buf, int buf_size, const char *sep,
147 if (**pp == '/') (*pp)++;
148 get_word_until_chars(buf, buf_size, sep, pp);
151 static void get_word(char *buf, int buf_size, const char **pp)
153 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
156 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
158 * Used for seeking in the rtp stream.
160 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
164 p += strspn(p, SPACE_CHARS);
165 if (!av_stristart(p, "npt=", &p))
168 *start = AV_NOPTS_VALUE;
169 *end = AV_NOPTS_VALUE;
171 get_word_sep(buf, sizeof(buf), "-", &p);
172 if (av_parse_time(start, buf, 1) < 0)
176 get_word_sep(buf, sizeof(buf), "-", &p);
177 av_parse_time(end, buf, 1);
181 static int get_sockaddr(AVFormatContext *s,
182 const char *buf, struct sockaddr_storage *sock)
184 struct addrinfo hints = { 0 }, *ai = NULL;
187 hints.ai_flags = AI_NUMERICHOST;
188 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
189 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
194 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
200 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
201 RTSPStream *rtsp_st, AVStream *st)
203 AVCodecParameters *par = st ? st->codecpar : NULL;
207 par->codec_id = handler->codec_id;
208 rtsp_st->dynamic_handler = handler;
210 st->need_parsing = handler->need_parsing;
211 if (handler->priv_data_size) {
212 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
213 if (!rtsp_st->dynamic_protocol_context)
214 rtsp_st->dynamic_handler = NULL;
218 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
221 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
222 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
223 rtsp_st->dynamic_protocol_context);
225 if (rtsp_st->dynamic_protocol_context) {
226 if (rtsp_st->dynamic_handler->close)
227 rtsp_st->dynamic_handler->close(
228 rtsp_st->dynamic_protocol_context);
229 av_free(rtsp_st->dynamic_protocol_context);
231 rtsp_st->dynamic_protocol_context = NULL;
232 rtsp_st->dynamic_handler = NULL;
237 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
238 static int sdp_parse_rtpmap(AVFormatContext *s,
239 AVStream *st, RTSPStream *rtsp_st,
240 int payload_type, const char *p)
242 AVCodecParameters *par = st->codecpar;
248 /* See if we can handle this kind of payload.
249 * The space should normally not be there but some Real streams or
250 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
251 * have a trailing space. */
252 get_word_sep(buf, sizeof(buf), "/ ", &p);
253 if (payload_type < RTP_PT_PRIVATE) {
254 /* We are in a standard case
255 * (from http://www.iana.org/assignments/rtp-parameters). */
256 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
259 if (par->codec_id == AV_CODEC_ID_NONE) {
260 RTPDynamicProtocolHandler *handler =
261 ff_rtp_handler_find_by_name(buf, par->codec_type);
262 init_rtp_handler(handler, rtsp_st, st);
263 /* If no dynamic handler was found, check with the list of standard
264 * allocated types, if such a stream for some reason happens to
265 * use a private payload type. This isn't handled in rtpdec.c, since
266 * the format name from the rtpmap line never is passed into rtpdec. */
267 if (!rtsp_st->dynamic_handler)
268 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
271 c = avcodec_find_decoder(par->codec_id);
277 get_word_sep(buf, sizeof(buf), "/", &p);
279 switch (par->codec_type) {
280 case AVMEDIA_TYPE_AUDIO:
281 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
282 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
283 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
285 par->sample_rate = i;
286 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
287 get_word_sep(buf, sizeof(buf), "/", &p);
292 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
294 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
297 case AVMEDIA_TYPE_VIDEO:
298 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
300 avpriv_set_pts_info(st, 32, 1, i);
305 finalize_rtp_handler_init(s, rtsp_st, st);
309 /* parse the attribute line from the fmtp a line of an sdp response. This
310 * is broken out as a function because it is used in rtp_h264.c, which is
312 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
313 char *value, int value_size)
315 *p += strspn(*p, SPACE_CHARS);
317 get_word_sep(attr, attr_size, "=", p);
320 get_word_sep(value, value_size, ";", p);
328 typedef struct SDPParseState {
330 struct sockaddr_storage default_ip;
332 int skip_media; ///< set if an unknown m= line occurs
333 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
334 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
335 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
336 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
339 char delayed_fmtp[2048];
342 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
343 struct RTSPSource ***dest, int *dest_count)
345 RTSPSource *rtsp_src, *rtsp_src2;
347 for (i = 0; i < count; i++) {
349 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
352 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
353 dynarray_add(dest, dest_count, rtsp_src2);
357 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
358 int payload_type, const char *line)
362 for (i = 0; i < rt->nb_rtsp_streams; i++) {
363 RTSPStream *rtsp_st = rt->rtsp_streams[i];
364 if (rtsp_st->sdp_payload_type == payload_type &&
365 rtsp_st->dynamic_handler &&
366 rtsp_st->dynamic_handler->parse_sdp_a_line) {
367 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
368 rtsp_st->dynamic_protocol_context, line);
373 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
374 int letter, const char *buf)
376 RTSPState *rt = s->priv_data;
377 char buf1[64], st_type[64];
379 enum AVMediaType codec_type;
383 RTSPSource *rtsp_src;
384 struct sockaddr_storage sdp_ip;
387 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
390 if (s1->skip_media && letter != 'm')
394 get_word(buf1, sizeof(buf1), &p);
395 if (strcmp(buf1, "IN") != 0)
397 get_word(buf1, sizeof(buf1), &p);
398 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
400 get_word_sep(buf1, sizeof(buf1), "/", &p);
401 if (get_sockaddr(s, buf1, &sdp_ip))
406 get_word_sep(buf1, sizeof(buf1), "/", &p);
409 if (s->nb_streams == 0) {
410 s1->default_ip = sdp_ip;
411 s1->default_ttl = ttl;
413 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
414 rtsp_st->sdp_ip = sdp_ip;
415 rtsp_st->sdp_ttl = ttl;
419 av_dict_set(&s->metadata, "title", p, 0);
422 if (s->nb_streams == 0) {
423 av_dict_set(&s->metadata, "comment", p, 0);
432 codec_type = AVMEDIA_TYPE_UNKNOWN;
433 get_word(st_type, sizeof(st_type), &p);
434 if (!strcmp(st_type, "audio")) {
435 codec_type = AVMEDIA_TYPE_AUDIO;
436 } else if (!strcmp(st_type, "video")) {
437 codec_type = AVMEDIA_TYPE_VIDEO;
438 } else if (!strcmp(st_type, "application") || !strcmp(st_type, "text")) {
439 codec_type = AVMEDIA_TYPE_DATA;
441 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
445 rtsp_st = av_mallocz(sizeof(RTSPStream));
448 rtsp_st->stream_index = -1;
449 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
451 rtsp_st->sdp_ip = s1->default_ip;
452 rtsp_st->sdp_ttl = s1->default_ttl;
454 copy_default_source_addrs(s1->default_include_source_addrs,
455 s1->nb_default_include_source_addrs,
456 &rtsp_st->include_source_addrs,
457 &rtsp_st->nb_include_source_addrs);
458 copy_default_source_addrs(s1->default_exclude_source_addrs,
459 s1->nb_default_exclude_source_addrs,
460 &rtsp_st->exclude_source_addrs,
461 &rtsp_st->nb_exclude_source_addrs);
463 get_word(buf1, sizeof(buf1), &p); /* port */
464 rtsp_st->sdp_port = atoi(buf1);
466 get_word(buf1, sizeof(buf1), &p); /* protocol */
467 if (!strcmp(buf1, "udp"))
468 rt->transport = RTSP_TRANSPORT_RAW;
469 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
470 rtsp_st->feedback = 1;
472 /* XXX: handle list of formats */
473 get_word(buf1, sizeof(buf1), &p); /* format list */
474 rtsp_st->sdp_payload_type = atoi(buf1);
476 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
477 /* no corresponding stream */
478 if (rt->transport == RTSP_TRANSPORT_RAW) {
479 if (CONFIG_RTPDEC && !rt->ts)
480 rt->ts = ff_mpegts_parse_open(s);
482 RTPDynamicProtocolHandler *handler;
483 handler = ff_rtp_handler_find_by_id(
484 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
485 init_rtp_handler(handler, rtsp_st, NULL);
486 finalize_rtp_handler_init(s, rtsp_st, NULL);
488 } else if (rt->server_type == RTSP_SERVER_WMS &&
489 codec_type == AVMEDIA_TYPE_DATA) {
490 /* RTX stream, a stream that carries all the other actual
491 * audio/video streams. Don't expose this to the callers. */
493 st = avformat_new_stream(s, NULL);
496 st->id = rt->nb_rtsp_streams - 1;
497 rtsp_st->stream_index = st->index;
498 st->codecpar->codec_type = codec_type;
499 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
500 RTPDynamicProtocolHandler *handler;
501 /* if standard payload type, we can find the codec right now */
502 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
503 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
504 st->codecpar->sample_rate > 0)
505 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
506 /* Even static payload types may need a custom depacketizer */
507 handler = ff_rtp_handler_find_by_id(
508 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
509 init_rtp_handler(handler, rtsp_st, st);
510 finalize_rtp_handler_init(s, rtsp_st, st);
512 if (rt->default_lang[0])
513 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
515 /* put a default control url */
516 av_strlcpy(rtsp_st->control_url, rt->control_uri,
517 sizeof(rtsp_st->control_url));
520 if (av_strstart(p, "control:", &p)) {
521 if (s->nb_streams == 0) {
522 if (!strncmp(p, "rtsp://", 7))
523 av_strlcpy(rt->control_uri, p,
524 sizeof(rt->control_uri));
527 /* get the control url */
528 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
530 /* XXX: may need to add full url resolution */
531 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
533 if (proto[0] == '\0') {
534 /* relative control URL */
535 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
536 av_strlcat(rtsp_st->control_url, "/",
537 sizeof(rtsp_st->control_url));
538 av_strlcat(rtsp_st->control_url, p,
539 sizeof(rtsp_st->control_url));
541 av_strlcpy(rtsp_st->control_url, p,
542 sizeof(rtsp_st->control_url));
544 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
545 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
546 get_word(buf1, sizeof(buf1), &p);
547 payload_type = atoi(buf1);
548 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
549 if (rtsp_st->stream_index >= 0) {
550 st = s->streams[rtsp_st->stream_index];
551 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
555 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
557 } else if (av_strstart(p, "fmtp:", &p) ||
558 av_strstart(p, "framesize:", &p)) {
559 // let dynamic protocol handlers have a stab at the line.
560 get_word(buf1, sizeof(buf1), &p);
561 payload_type = atoi(buf1);
562 if (s1->seen_rtpmap) {
563 parse_fmtp(s, rt, payload_type, buf);
566 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
568 } else if (av_strstart(p, "range:", &p)) {
571 // this is so that seeking on a streamed file can work.
572 rtsp_parse_range_npt(p, &start, &end);
573 s->start_time = start;
574 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
575 s->duration = (end == AV_NOPTS_VALUE) ?
576 AV_NOPTS_VALUE : end - start;
577 } else if (av_strstart(p, "lang:", &p)) {
578 if (s->nb_streams > 0) {
579 get_word(buf1, sizeof(buf1), &p);
580 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
581 if (rtsp_st->stream_index >= 0) {
582 st = s->streams[rtsp_st->stream_index];
583 av_dict_set(&st->metadata, "language", buf1, 0);
586 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
587 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
589 rt->transport = RTSP_TRANSPORT_RDT;
590 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
592 st = s->streams[s->nb_streams - 1];
593 st->codecpar->sample_rate = atoi(p);
594 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
596 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
597 get_word(buf1, sizeof(buf1), &p); // ignore tag
598 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
599 p += strspn(p, SPACE_CHARS);
600 if (av_strstart(p, "inline:", &p))
601 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
602 } else if (av_strstart(p, "source-filter:", &p)) {
604 get_word(buf1, sizeof(buf1), &p);
605 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
607 exclude = !strcmp(buf1, "excl");
609 get_word(buf1, sizeof(buf1), &p);
610 if (strcmp(buf1, "IN") != 0)
612 get_word(buf1, sizeof(buf1), &p);
613 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
615 // not checking that the destination address actually matches or is wildcard
616 get_word(buf1, sizeof(buf1), &p);
619 rtsp_src = av_mallocz(sizeof(*rtsp_src));
622 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
624 if (s->nb_streams == 0) {
625 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
627 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
628 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
631 if (s->nb_streams == 0) {
632 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
634 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
635 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
640 if (rt->server_type == RTSP_SERVER_WMS)
641 ff_wms_parse_sdp_a_line(s, p);
642 if (s->nb_streams > 0) {
643 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
645 if (rt->server_type == RTSP_SERVER_REAL)
646 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
648 if (rtsp_st->dynamic_handler &&
649 rtsp_st->dynamic_handler->parse_sdp_a_line)
650 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
651 rtsp_st->stream_index,
652 rtsp_st->dynamic_protocol_context, buf);
659 int ff_sdp_parse(AVFormatContext *s, const char *content)
661 RTSPState *rt = s->priv_data;
664 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
665 * contain long SDP lines containing complete ASF Headers (several
666 * kB) or arrays of MDPR (RM stream descriptor) headers plus
667 * "rulebooks" describing their properties. Therefore, the SDP line
670 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
671 * in rtpdec_xiph.c. */
673 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
677 p += strspn(p, SPACE_CHARS);
685 /* get the content */
687 while (*p != '\n' && *p != '\r' && *p != '\0') {
688 if ((q - buf) < sizeof(buf) - 1)
693 sdp_parse_line(s, s1, letter, buf);
695 while (*p != '\n' && *p != '\0')
701 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
702 av_free(s1->default_include_source_addrs[i]);
703 av_freep(&s1->default_include_source_addrs);
704 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
705 av_free(s1->default_exclude_source_addrs[i]);
706 av_freep(&s1->default_exclude_source_addrs);
708 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
709 if (!rt->p) return AVERROR(ENOMEM);
712 #endif /* CONFIG_RTPDEC */
714 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
716 RTSPState *rt = s->priv_data;
719 for (i = 0; i < rt->nb_rtsp_streams; i++) {
720 RTSPStream *rtsp_st = rt->rtsp_streams[i];
723 if (rtsp_st->transport_priv) {
725 AVFormatContext *rtpctx = rtsp_st->transport_priv;
726 av_write_trailer(rtpctx);
727 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
728 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
729 ff_rtsp_tcp_write_packet(s, rtsp_st);
730 ffio_free_dyn_buf(&rtpctx->pb);
732 avio_close(rtpctx->pb);
734 avformat_free_context(rtpctx);
735 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
736 ff_rdt_parse_close(rtsp_st->transport_priv);
737 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
738 ff_rtp_parse_close(rtsp_st->transport_priv);
740 rtsp_st->transport_priv = NULL;
741 if (rtsp_st->rtp_handle)
742 ffurl_close(rtsp_st->rtp_handle);
743 rtsp_st->rtp_handle = NULL;
747 /* close and free RTSP streams */
748 void ff_rtsp_close_streams(AVFormatContext *s)
750 RTSPState *rt = s->priv_data;
754 ff_rtsp_undo_setup(s, 0);
755 for (i = 0; i < rt->nb_rtsp_streams; i++) {
756 rtsp_st = rt->rtsp_streams[i];
758 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
759 if (rtsp_st->dynamic_handler->close)
760 rtsp_st->dynamic_handler->close(
761 rtsp_st->dynamic_protocol_context);
762 av_free(rtsp_st->dynamic_protocol_context);
764 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
765 av_free(rtsp_st->include_source_addrs[j]);
766 av_freep(&rtsp_st->include_source_addrs);
767 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
768 av_free(rtsp_st->exclude_source_addrs[j]);
769 av_freep(&rtsp_st->exclude_source_addrs);
774 av_free(rt->rtsp_streams);
776 avformat_close_input(&rt->asf_ctx);
778 if (CONFIG_RTPDEC && rt->ts)
779 ff_mpegts_parse_close(rt->ts);
780 av_freep(&rt->protocols);
782 av_free(rt->recvbuf);
785 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
787 RTSPState *rt = s->priv_data;
789 int reordering_queue_size = rt->reordering_queue_size;
790 if (reordering_queue_size < 0) {
791 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
792 reordering_queue_size = 0;
794 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
797 /* open the RTP context */
798 if (rtsp_st->stream_index >= 0)
799 st = s->streams[rtsp_st->stream_index];
801 s->ctx_flags |= AVFMTCTX_NOHEADER;
803 if (CONFIG_RTSP_MUXER && s->oformat) {
804 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
805 s, st, rtsp_st->rtp_handle,
806 RTSP_TCP_MAX_PACKET_SIZE,
807 rtsp_st->stream_index);
808 /* Ownership of rtp_handle is passed to the rtp mux context */
809 rtsp_st->rtp_handle = NULL;
812 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
813 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
814 return 0; // Don't need to open any parser here
815 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
816 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
817 rtsp_st->dynamic_protocol_context,
818 rtsp_st->dynamic_handler);
819 else if (CONFIG_RTPDEC)
820 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
821 rtsp_st->sdp_payload_type,
822 reordering_queue_size);
824 if (!rtsp_st->transport_priv) {
825 return AVERROR(ENOMEM);
826 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
827 if (rtsp_st->dynamic_handler) {
828 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
829 rtsp_st->dynamic_protocol_context,
830 rtsp_st->dynamic_handler);
832 if (rtsp_st->crypto_suite[0])
833 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
834 rtsp_st->crypto_suite,
835 rtsp_st->crypto_params);
841 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
842 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
849 q += strspn(q, SPACE_CHARS);
850 v = strtol(q, &p, 10);
854 v = strtol(p, &p, 10);
863 /* XXX: only one transport specification is parsed */
864 static void rtsp_parse_transport(AVFormatContext *s,
865 RTSPMessageHeader *reply, const char *p)
867 char transport_protocol[16];
869 char lower_transport[16];
871 RTSPTransportField *th;
874 reply->nb_transports = 0;
877 p += strspn(p, SPACE_CHARS);
881 th = &reply->transports[reply->nb_transports];
883 get_word_sep(transport_protocol, sizeof(transport_protocol),
885 if (!av_strcasecmp (transport_protocol, "rtp")) {
886 get_word_sep(profile, sizeof(profile), "/;,", &p);
887 lower_transport[0] = '\0';
888 /* rtp/avp/<protocol> */
890 get_word_sep(lower_transport, sizeof(lower_transport),
893 th->transport = RTSP_TRANSPORT_RTP;
894 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
895 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
896 /* x-pn-tng/<protocol> */
897 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
899 th->transport = RTSP_TRANSPORT_RDT;
900 } else if (!av_strcasecmp(transport_protocol, "raw")) {
901 get_word_sep(profile, sizeof(profile), "/;,", &p);
902 lower_transport[0] = '\0';
903 /* raw/raw/<protocol> */
905 get_word_sep(lower_transport, sizeof(lower_transport),
908 th->transport = RTSP_TRANSPORT_RAW;
910 if (!av_strcasecmp(lower_transport, "TCP"))
911 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
913 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
917 /* get each parameter */
918 while (*p != '\0' && *p != ',') {
919 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
920 if (!strcmp(parameter, "port")) {
923 rtsp_parse_range(&th->port_min, &th->port_max, &p);
925 } else if (!strcmp(parameter, "client_port")) {
928 rtsp_parse_range(&th->client_port_min,
929 &th->client_port_max, &p);
931 } else if (!strcmp(parameter, "server_port")) {
934 rtsp_parse_range(&th->server_port_min,
935 &th->server_port_max, &p);
937 } else if (!strcmp(parameter, "interleaved")) {
940 rtsp_parse_range(&th->interleaved_min,
941 &th->interleaved_max, &p);
943 } else if (!strcmp(parameter, "multicast")) {
944 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
945 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
946 } else if (!strcmp(parameter, "ttl")) {
950 th->ttl = strtol(p, &end, 10);
953 } else if (!strcmp(parameter, "destination")) {
956 get_word_sep(buf, sizeof(buf), ";,", &p);
957 get_sockaddr(s, buf, &th->destination);
959 } else if (!strcmp(parameter, "source")) {
962 get_word_sep(buf, sizeof(buf), ";,", &p);
963 av_strlcpy(th->source, buf, sizeof(th->source));
965 } else if (!strcmp(parameter, "mode")) {
968 get_word_sep(buf, sizeof(buf), ";, ", &p);
969 if (!strcmp(buf, "record") ||
970 !strcmp(buf, "receive"))
975 while (*p != ';' && *p != '\0' && *p != ',')
983 reply->nb_transports++;
984 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
989 static void handle_rtp_info(RTSPState *rt, const char *url,
990 uint32_t seq, uint32_t rtptime)
993 if (!rtptime || !url[0])
995 if (rt->transport != RTSP_TRANSPORT_RTP)
997 for (i = 0; i < rt->nb_rtsp_streams; i++) {
998 RTSPStream *rtsp_st = rt->rtsp_streams[i];
999 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1002 if (!strcmp(rtsp_st->control_url, url)) {
1003 rtpctx->base_timestamp = rtptime;
1009 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1012 char key[20], value[1024], url[1024] = "";
1013 uint32_t seq = 0, rtptime = 0;
1016 p += strspn(p, SPACE_CHARS);
1019 get_word_sep(key, sizeof(key), "=", &p);
1023 get_word_sep(value, sizeof(value), ";, ", &p);
1025 if (!strcmp(key, "url"))
1026 av_strlcpy(url, value, sizeof(url));
1027 else if (!strcmp(key, "seq"))
1028 seq = strtoul(value, NULL, 10);
1029 else if (!strcmp(key, "rtptime"))
1030 rtptime = strtoul(value, NULL, 10);
1032 handle_rtp_info(rt, url, seq, rtptime);
1041 handle_rtp_info(rt, url, seq, rtptime);
1044 void ff_rtsp_parse_line(AVFormatContext *s,
1045 RTSPMessageHeader *reply, const char *buf,
1046 RTSPState *rt, const char *method)
1050 /* NOTE: we do case independent match for broken servers */
1052 if (av_stristart(p, "Session:", &p)) {
1054 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1055 if (av_stristart(p, ";timeout=", &p) &&
1056 (t = strtol(p, NULL, 10)) > 0) {
1059 } else if (av_stristart(p, "Content-Length:", &p)) {
1060 reply->content_length = strtol(p, NULL, 10);
1061 } else if (av_stristart(p, "Transport:", &p)) {
1062 rtsp_parse_transport(s, reply, p);
1063 } else if (av_stristart(p, "CSeq:", &p)) {
1064 reply->seq = strtol(p, NULL, 10);
1065 } else if (av_stristart(p, "Range:", &p)) {
1066 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1067 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1068 p += strspn(p, SPACE_CHARS);
1069 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1070 } else if (av_stristart(p, "Server:", &p)) {
1071 p += strspn(p, SPACE_CHARS);
1072 av_strlcpy(reply->server, p, sizeof(reply->server));
1073 } else if (av_stristart(p, "Notice:", &p) ||
1074 av_stristart(p, "X-Notice:", &p)) {
1075 reply->notice = strtol(p, NULL, 10);
1076 } else if (av_stristart(p, "Location:", &p)) {
1077 p += strspn(p, SPACE_CHARS);
1078 av_strlcpy(reply->location, p , sizeof(reply->location));
1079 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1080 p += strspn(p, SPACE_CHARS);
1081 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1082 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1083 p += strspn(p, SPACE_CHARS);
1084 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1085 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1086 p += strspn(p, SPACE_CHARS);
1087 if (method && !strcmp(method, "DESCRIBE"))
1088 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1089 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1090 p += strspn(p, SPACE_CHARS);
1091 if (method && !strcmp(method, "PLAY"))
1092 rtsp_parse_rtp_info(rt, p);
1093 } else if (av_stristart(p, "Public:", &p) && rt) {
1094 if (strstr(p, "GET_PARAMETER") &&
1095 method && !strcmp(method, "OPTIONS"))
1096 rt->get_parameter_supported = 1;
1097 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1098 p += strspn(p, SPACE_CHARS);
1099 rt->accept_dynamic_rate = atoi(p);
1100 } else if (av_stristart(p, "Content-Type:", &p)) {
1101 p += strspn(p, SPACE_CHARS);
1102 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1106 /* skip a RTP/TCP interleaved packet */
1107 void ff_rtsp_skip_packet(AVFormatContext *s)
1109 RTSPState *rt = s->priv_data;
1113 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1116 len = AV_RB16(buf + 1);
1118 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1123 if (len1 > sizeof(buf))
1125 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1132 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1133 unsigned char **content_ptr,
1134 int return_on_interleaved_data, const char *method)
1136 RTSPState *rt = s->priv_data;
1137 char buf[4096], buf1[1024], *q;
1140 int ret, content_length, line_count = 0, request = 0;
1141 unsigned char *content = NULL;
1147 memset(reply, 0, sizeof(*reply));
1149 /* parse reply (XXX: use buffers) */
1150 rt->last_reply[0] = '\0';
1154 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1155 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1160 if (ch == '$' && q == buf) {
1161 if (return_on_interleaved_data) {
1164 ff_rtsp_skip_packet(s);
1165 } else if (ch != '\r') {
1166 if ((q - buf) < sizeof(buf) - 1)
1172 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1174 /* test if last line */
1178 if (line_count == 0) {
1179 /* get reply code */
1180 get_word(buf1, sizeof(buf1), &p);
1181 if (!strncmp(buf1, "RTSP/", 5)) {
1182 get_word(buf1, sizeof(buf1), &p);
1183 reply->status_code = atoi(buf1);
1184 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1186 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1187 get_word(buf1, sizeof(buf1), &p); // object
1191 ff_rtsp_parse_line(s, reply, p, rt, method);
1192 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1193 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1198 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1199 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1201 content_length = reply->content_length;
1202 if (content_length > 0) {
1203 /* leave some room for a trailing '\0' (useful for simple parsing) */
1204 content = av_malloc(content_length + 1);
1206 return AVERROR(ENOMEM);
1207 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1208 content[content_length] = '\0';
1211 *content_ptr = content;
1217 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1218 const char* ptr = buf;
1220 if (!strcmp(reply->reason, "OPTIONS")) {
1221 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1223 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1224 if (reply->session_id[0])
1225 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1228 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1230 av_strlcat(buf, "\r\n", sizeof(buf));
1232 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1233 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1236 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1238 rt->last_cmd_time = av_gettime_relative();
1239 /* Even if the request from the server had data, it is not the data
1240 * that the caller wants or expects. The memory could also be leaked
1241 * if the actual following reply has content data. */
1243 av_freep(content_ptr);
1244 /* If method is set, this is called from ff_rtsp_send_cmd,
1245 * where a reply to exactly this request is awaited. For
1246 * callers from within packet receiving, we just want to
1247 * return to the caller and go back to receiving packets. */
1253 if (rt->seq != reply->seq) {
1254 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1255 rt->seq, reply->seq);
1259 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1260 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1261 reply->notice == 2306 /* Continuous Feed Terminated */) {
1262 rt->state = RTSP_STATE_IDLE;
1263 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1264 return AVERROR(EIO); /* data or server error */
1265 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1266 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1267 return AVERROR(EPERM);
1273 * Send a command to the RTSP server without waiting for the reply.
1275 * @param s RTSP (de)muxer context
1276 * @param method the method for the request
1277 * @param url the target url for the request
1278 * @param headers extra header lines to include in the request
1279 * @param send_content if non-null, the data to send as request body content
1280 * @param send_content_length the length of the send_content data, or 0 if
1281 * send_content is null
1283 * @return zero if success, nonzero otherwise
1285 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1286 const char *method, const char *url,
1287 const char *headers,
1288 const unsigned char *send_content,
1289 int send_content_length)
1291 RTSPState *rt = s->priv_data;
1292 char buf[4096], *out_buf;
1293 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1295 /* Add in RTSP headers */
1298 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1300 av_strlcat(buf, headers, sizeof(buf));
1301 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1302 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1303 if (rt->session_id[0] != '\0' && (!headers ||
1304 !strstr(headers, "\nIf-Match:"))) {
1305 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1308 char *str = ff_http_auth_create_response(&rt->auth_state,
1309 rt->auth, url, method);
1311 av_strlcat(buf, str, sizeof(buf));
1314 if (send_content_length > 0 && send_content)
1315 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1316 av_strlcat(buf, "\r\n", sizeof(buf));
1318 /* base64 encode rtsp if tunneling */
1319 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1320 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1321 out_buf = base64buf;
1324 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1326 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1327 if (send_content_length > 0 && send_content) {
1328 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1329 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1330 "with content data not supported\n");
1331 return AVERROR_PATCHWELCOME;
1333 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1335 rt->last_cmd_time = av_gettime_relative();
1340 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1341 const char *url, const char *headers)
1343 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1346 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1347 const char *headers, RTSPMessageHeader *reply,
1348 unsigned char **content_ptr)
1350 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1351 content_ptr, NULL, 0);
1354 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1355 const char *method, const char *url,
1357 RTSPMessageHeader *reply,
1358 unsigned char **content_ptr,
1359 const unsigned char *send_content,
1360 int send_content_length)
1362 RTSPState *rt = s->priv_data;
1363 HTTPAuthType cur_auth_type;
1364 int ret, attempts = 0;
1367 cur_auth_type = rt->auth_state.auth_type;
1368 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1370 send_content_length)))
1373 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1377 if (reply->status_code == 401 &&
1378 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1379 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1382 if (reply->status_code > 400){
1383 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1387 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1393 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1394 int lower_transport, const char *real_challenge)
1396 RTSPState *rt = s->priv_data;
1397 int rtx = 0, j, i, err, interleave = 0, port_off;
1398 RTSPStream *rtsp_st;
1399 RTSPMessageHeader reply1, *reply = &reply1;
1401 const char *trans_pref;
1403 if (rt->transport == RTSP_TRANSPORT_RDT)
1404 trans_pref = "x-pn-tng";
1405 else if (rt->transport == RTSP_TRANSPORT_RAW)
1406 trans_pref = "RAW/RAW";
1408 trans_pref = "RTP/AVP";
1410 /* default timeout: 1 minute */
1413 /* for each stream, make the setup request */
1414 /* XXX: we assume the same server is used for the control of each
1417 /* Choose a random starting offset within the first half of the
1418 * port range, to allow for a number of ports to try even if the offset
1419 * happens to be at the end of the random range. */
1420 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1421 /* even random offset */
1422 port_off -= port_off & 0x01;
1424 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1425 char transport[2048];
1428 * WMS serves all UDP data over a single connection, the RTX, which
1429 * isn't necessarily the first in the SDP but has to be the first
1430 * to be set up, else the second/third SETUP will fail with a 461.
1432 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1433 rt->server_type == RTSP_SERVER_WMS) {
1436 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1437 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1439 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1443 if (rtx == rt->nb_rtsp_streams)
1444 return -1; /* no RTX found */
1445 rtsp_st = rt->rtsp_streams[rtx];
1447 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1449 rtsp_st = rt->rtsp_streams[i];
1452 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1455 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1456 port = reply->transports[0].client_port_min;
1460 /* first try in specified port range */
1461 while (j <= rt->rtp_port_max) {
1462 AVDictionary *opts = map_to_opts(rt);
1464 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1465 "?localport=%d", j);
1466 /* we will use two ports per rtp stream (rtp and rtcp) */
1468 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1469 &s->interrupt_callback, &opts, rt->protocols, NULL);
1471 av_dict_free(&opts);
1477 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1482 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1484 snprintf(transport, sizeof(transport) - 1,
1485 "%s/UDP;", trans_pref);
1486 if (rt->server_type != RTSP_SERVER_REAL)
1487 av_strlcat(transport, "unicast;", sizeof(transport));
1488 av_strlcatf(transport, sizeof(transport),
1489 "client_port=%d", port);
1490 if (rt->transport == RTSP_TRANSPORT_RTP &&
1491 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1492 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1496 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1497 /* For WMS streams, the application streams are only used for
1498 * UDP. When trying to set it up for TCP streams, the server
1499 * will return an error. Therefore, we skip those streams. */
1500 if (rt->server_type == RTSP_SERVER_WMS &&
1501 (rtsp_st->stream_index < 0 ||
1502 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1505 snprintf(transport, sizeof(transport) - 1,
1506 "%s/TCP;", trans_pref);
1507 if (rt->transport != RTSP_TRANSPORT_RDT)
1508 av_strlcat(transport, "unicast;", sizeof(transport));
1509 av_strlcatf(transport, sizeof(transport),
1510 "interleaved=%d-%d",
1511 interleave, interleave + 1);
1515 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1516 snprintf(transport, sizeof(transport) - 1,
1517 "%s/UDP;multicast", trans_pref);
1520 av_strlcat(transport, ";mode=record", sizeof(transport));
1521 } else if (rt->server_type == RTSP_SERVER_REAL ||
1522 rt->server_type == RTSP_SERVER_WMS)
1523 av_strlcat(transport, ";mode=play", sizeof(transport));
1524 snprintf(cmd, sizeof(cmd),
1525 "Transport: %s\r\n",
1527 if (rt->accept_dynamic_rate)
1528 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1529 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1530 char real_res[41], real_csum[9];
1531 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1533 av_strlcatf(cmd, sizeof(cmd),
1535 "RealChallenge2: %s, sd=%s\r\n",
1536 rt->session_id, real_res, real_csum);
1538 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1539 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1542 } else if (reply->status_code != RTSP_STATUS_OK ||
1543 reply->nb_transports != 1) {
1544 err = AVERROR_INVALIDDATA;
1548 /* XXX: same protocol for all streams is required */
1550 if (reply->transports[0].lower_transport != rt->lower_transport ||
1551 reply->transports[0].transport != rt->transport) {
1552 err = AVERROR_INVALIDDATA;
1556 rt->lower_transport = reply->transports[0].lower_transport;
1557 rt->transport = reply->transports[0].transport;
1560 /* Fail if the server responded with another lower transport mode
1561 * than what we requested. */
1562 if (reply->transports[0].lower_transport != lower_transport) {
1563 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1564 err = AVERROR_INVALIDDATA;
1568 switch(reply->transports[0].lower_transport) {
1569 case RTSP_LOWER_TRANSPORT_TCP:
1570 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1571 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1574 case RTSP_LOWER_TRANSPORT_UDP: {
1575 char url[1024], options[30] = "";
1576 const char *peer = host;
1578 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1579 av_strlcpy(options, "?connect=1", sizeof(options));
1580 /* Use source address if specified */
1581 if (reply->transports[0].source[0])
1582 peer = reply->transports[0].source;
1583 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1584 reply->transports[0].server_port_min, "%s", options);
1585 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1586 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1587 err = AVERROR_INVALIDDATA;
1592 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1593 char url[1024], namebuf[50], optbuf[20] = "";
1594 struct sockaddr_storage addr;
1597 if (reply->transports[0].destination.ss_family) {
1598 addr = reply->transports[0].destination;
1599 port = reply->transports[0].port_min;
1600 ttl = reply->transports[0].ttl;
1602 addr = rtsp_st->sdp_ip;
1603 port = rtsp_st->sdp_port;
1604 ttl = rtsp_st->sdp_ttl;
1607 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1608 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1609 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1610 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1611 port, "%s", optbuf);
1612 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1613 &s->interrupt_callback, NULL, rt->protocols, NULL) < 0) {
1614 err = AVERROR_INVALIDDATA;
1621 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1625 if (rt->nb_rtsp_streams && reply->timeout > 0)
1626 rt->timeout = reply->timeout;
1628 if (rt->server_type == RTSP_SERVER_REAL)
1629 rt->need_subscription = 1;
1634 ff_rtsp_undo_setup(s, 0);
1638 void ff_rtsp_close_connections(AVFormatContext *s)
1640 RTSPState *rt = s->priv_data;
1641 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1642 ffurl_close(rt->rtsp_hd);
1643 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1646 int ff_rtsp_connect(AVFormatContext *s)
1648 RTSPState *rt = s->priv_data;
1649 char proto[128], host[1024], path[1024];
1650 char tcpname[1024], cmd[2048], auth[128];
1651 const char *lower_rtsp_proto = "tcp";
1652 int port, err, tcp_fd;
1653 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1654 int lower_transport_mask = 0;
1655 int default_port = RTSP_DEFAULT_PORT;
1656 char real_challenge[64] = "";
1657 struct sockaddr_storage peer;
1658 socklen_t peer_len = sizeof(peer);
1660 if (rt->rtp_port_max < rt->rtp_port_min) {
1661 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1662 "than min port %d\n", rt->rtp_port_max,
1664 return AVERROR(EINVAL);
1667 if (!ff_network_init())
1668 return AVERROR(EIO);
1670 if (!rt->protocols) {
1671 rt->protocols = ffurl_get_protocols(s->protocol_whitelist,
1672 s->protocol_blacklist);
1674 return AVERROR(ENOMEM);
1677 if (s->max_delay < 0) /* Not set by the caller */
1678 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1680 rt->control_transport = RTSP_MODE_PLAIN;
1681 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1682 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1683 rt->control_transport = RTSP_MODE_TUNNEL;
1685 /* Only pass through valid flags from here */
1686 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1689 /* extract hostname and port */
1690 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1691 host, sizeof(host), &port, path, sizeof(path), s->filename);
1693 if (!strcmp(proto, "rtsps")) {
1694 lower_rtsp_proto = "tls";
1695 default_port = RTSPS_DEFAULT_PORT;
1696 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1700 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1703 port = default_port;
1705 lower_transport_mask = rt->lower_transport_mask;
1707 if (!lower_transport_mask)
1708 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1711 /* Only UDP or TCP - UDP multicast isn't supported. */
1712 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1713 (1 << RTSP_LOWER_TRANSPORT_TCP);
1714 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1715 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1716 "only UDP and TCP are supported for output.\n");
1717 err = AVERROR(EINVAL);
1722 /* Construct the URI used in request; this is similar to s->filename,
1723 * but with authentication credentials removed and RTSP specific options
1725 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1726 host, port, "%s", path);
1728 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1729 /* set up initial handshake for tunneling */
1730 char httpname[1024];
1731 char sessioncookie[17];
1734 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1735 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1736 av_get_random_seed(), av_get_random_seed());
1739 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1740 &s->interrupt_callback, rt->protocols) < 0) {
1745 /* generate GET headers */
1746 snprintf(headers, sizeof(headers),
1747 "x-sessioncookie: %s\r\n"
1748 "Accept: application/x-rtsp-tunnelled\r\n"
1749 "Pragma: no-cache\r\n"
1750 "Cache-Control: no-cache\r\n",
1752 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1754 /* complete the connection */
1755 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1761 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1762 &s->interrupt_callback, rt->protocols) < 0 ) {
1767 /* generate POST headers */
1768 snprintf(headers, sizeof(headers),
1769 "x-sessioncookie: %s\r\n"
1770 "Content-Type: application/x-rtsp-tunnelled\r\n"
1771 "Pragma: no-cache\r\n"
1772 "Cache-Control: no-cache\r\n"
1773 "Content-Length: 32767\r\n"
1774 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1776 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1777 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1779 /* Initialize the authentication state for the POST session. The HTTP
1780 * protocol implementation doesn't properly handle multi-pass
1781 * authentication for POST requests, since it would require one of
1783 * - implementing Expect: 100-continue, which many HTTP servers
1784 * don't support anyway, even less the RTSP servers that do HTTP
1786 * - sending the whole POST data until getting a 401 reply specifying
1787 * what authentication method to use, then resending all that data
1788 * - waiting for potential 401 replies directly after sending the
1789 * POST header (waiting for some unspecified time)
1790 * Therefore, we copy the full auth state, which works for both basic
1791 * and digest. (For digest, we would have to synchronize the nonce
1792 * count variable between the two sessions, if we'd do more requests
1793 * with the original session, though.)
1795 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1797 /* complete the connection */
1798 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1803 /* open the tcp connection */
1804 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1806 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1807 &s->interrupt_callback, NULL, rt->protocols, NULL) < 0) {
1811 rt->rtsp_hd_out = rt->rtsp_hd;
1815 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1820 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1821 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1822 NULL, 0, NI_NUMERICHOST);
1825 /* request options supported by the server; this also detects server
1827 for (rt->server_type = RTSP_SERVER_RTP;;) {
1829 if (rt->server_type == RTSP_SERVER_REAL)
1832 * The following entries are required for proper
1833 * streaming from a Realmedia server. They are
1834 * interdependent in some way although we currently
1835 * don't quite understand how. Values were copied
1836 * from mplayer SVN r23589.
1837 * ClientChallenge is a 16-byte ID in hex
1838 * CompanyID is a 16-byte ID in base64
1840 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1841 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1842 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1843 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1845 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1846 if (reply->status_code != RTSP_STATUS_OK) {
1847 err = AVERROR_INVALIDDATA;
1851 /* detect server type if not standard-compliant RTP */
1852 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1853 rt->server_type = RTSP_SERVER_REAL;
1855 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1856 rt->server_type = RTSP_SERVER_WMS;
1857 } else if (rt->server_type == RTSP_SERVER_REAL)
1858 strcpy(real_challenge, reply->real_challenge);
1862 if (CONFIG_RTSP_DEMUXER && s->iformat)
1863 err = ff_rtsp_setup_input_streams(s, reply);
1864 else if (CONFIG_RTSP_MUXER)
1865 err = ff_rtsp_setup_output_streams(s, host);
1870 int lower_transport = ff_log2_tab[lower_transport_mask &
1871 ~(lower_transport_mask - 1)];
1873 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1874 rt->server_type == RTSP_SERVER_REAL ?
1875 real_challenge : NULL);
1878 lower_transport_mask &= ~(1 << lower_transport);
1879 if (lower_transport_mask == 0 && err == 1) {
1880 err = AVERROR(EPROTONOSUPPORT);
1885 rt->lower_transport_mask = lower_transport_mask;
1886 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1887 rt->state = RTSP_STATE_IDLE;
1888 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1891 ff_rtsp_close_streams(s);
1892 ff_rtsp_close_connections(s);
1893 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1894 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1895 rt->session_id[0] = '\0';
1896 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1904 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1907 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1908 uint8_t *buf, int buf_size, int64_t wait_end)
1910 RTSPState *rt = s->priv_data;
1911 RTSPStream *rtsp_st;
1912 int n, i, ret, tcp_fd, timeout_cnt = 0;
1914 struct pollfd *p = rt->p;
1915 int *fds = NULL, fdsnum, fdsidx;
1918 if (ff_check_interrupt(&s->interrupt_callback))
1919 return AVERROR_EXIT;
1920 if (wait_end && wait_end - av_gettime_relative() < 0)
1921 return AVERROR(EAGAIN);
1924 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1925 p[max_p].fd = tcp_fd;
1926 p[max_p++].events = POLLIN;
1930 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1931 rtsp_st = rt->rtsp_streams[i];
1932 if (rtsp_st->rtp_handle) {
1933 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1935 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1939 av_log(s, AV_LOG_ERROR,
1940 "Number of fds %d not supported\n", fdsnum);
1941 return AVERROR_INVALIDDATA;
1943 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1944 p[max_p].fd = fds[fdsidx];
1945 p[max_p++].events = POLLIN;
1950 n = poll(p, max_p, POLL_TIMEOUT_MS);
1952 int j = 1 - (tcp_fd == -1);
1954 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1955 rtsp_st = rt->rtsp_streams[i];
1956 if (rtsp_st->rtp_handle) {
1957 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1958 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1960 *prtsp_st = rtsp_st;
1967 #if CONFIG_RTSP_DEMUXER
1968 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1969 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1970 if (rt->state == RTSP_STATE_STREAMING) {
1971 if (!ff_rtsp_parse_streaming_commands(s))
1974 av_log(s, AV_LOG_WARNING,
1975 "Unable to answer to TEARDOWN\n");
1979 RTSPMessageHeader reply;
1980 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1983 /* XXX: parse message */
1984 if (rt->state != RTSP_STATE_STREAMING)
1989 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1990 return AVERROR(ETIMEDOUT);
1991 } else if (n < 0 && errno != EINTR)
1992 return AVERROR(errno);
1996 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1997 const uint8_t *buf, int len)
1999 RTSPState *rt = s->priv_data;
2003 if (rt->nb_rtsp_streams == 1) {
2004 *rtsp_st = rt->rtsp_streams[0];
2007 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2008 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2010 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2011 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2014 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2015 *rtsp_st = rt->rtsp_streams[i];
2022 av_log(s, AV_LOG_WARNING,
2023 "Unable to pick stream for packet - SSRC not known for "
2025 return AVERROR(EAGAIN);
2028 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2029 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2030 *rtsp_st = rt->rtsp_streams[i];
2036 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2037 return AVERROR(EAGAIN);
2040 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2042 RTSPState *rt = s->priv_data;
2044 RTSPStream *rtsp_st, *first_queue_st = NULL;
2045 int64_t wait_end = 0;
2047 if (rt->nb_byes == rt->nb_rtsp_streams)
2050 /* get next frames from the same RTP packet */
2051 if (rt->cur_transport_priv) {
2052 if (rt->transport == RTSP_TRANSPORT_RDT) {
2053 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2054 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2055 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2056 } else if (CONFIG_RTPDEC && rt->ts) {
2057 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2059 rt->recvbuf_pos += ret;
2060 ret = rt->recvbuf_pos < rt->recvbuf_len;
2065 rt->cur_transport_priv = NULL;
2067 } else if (ret == 1) {
2070 rt->cur_transport_priv = NULL;
2074 if (rt->transport == RTSP_TRANSPORT_RTP) {
2076 int64_t first_queue_time = 0;
2077 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2078 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2082 queue_time = ff_rtp_queued_packet_time(rtpctx);
2083 if (queue_time && (queue_time - first_queue_time < 0 ||
2084 !first_queue_time)) {
2085 first_queue_time = queue_time;
2086 first_queue_st = rt->rtsp_streams[i];
2089 if (first_queue_time) {
2090 wait_end = first_queue_time + s->max_delay;
2093 first_queue_st = NULL;
2097 /* read next RTP packet */
2099 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2101 return AVERROR(ENOMEM);
2104 switch(rt->lower_transport) {
2106 #if CONFIG_RTSP_DEMUXER
2107 case RTSP_LOWER_TRANSPORT_TCP:
2108 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2111 case RTSP_LOWER_TRANSPORT_UDP:
2112 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2113 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2114 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2115 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2117 case RTSP_LOWER_TRANSPORT_CUSTOM:
2118 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2119 wait_end && wait_end < av_gettime_relative())
2120 len = AVERROR(EAGAIN);
2122 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2123 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2124 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2125 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2128 if (len == AVERROR(EAGAIN) && first_queue_st &&
2129 rt->transport == RTSP_TRANSPORT_RTP) {
2130 av_log(s, AV_LOG_WARNING,
2131 "max delay reached. need to consume packet\n");
2132 rtsp_st = first_queue_st;
2133 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2140 if (rt->transport == RTSP_TRANSPORT_RDT) {
2141 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2142 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2143 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2144 if (rtsp_st->feedback) {
2145 AVIOContext *pb = NULL;
2146 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2148 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2151 /* Either bad packet, or a RTCP packet. Check if the
2152 * first_rtcp_ntp_time field was initialized. */
2153 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2154 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2155 /* first_rtcp_ntp_time has been initialized for this stream,
2156 * copy the same value to all other uninitialized streams,
2157 * in order to map their timestamp origin to the same ntp time
2160 AVStream *st = NULL;
2161 if (rtsp_st->stream_index >= 0)
2162 st = s->streams[rtsp_st->stream_index];
2163 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2164 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2165 AVStream *st2 = NULL;
2166 if (rt->rtsp_streams[i]->stream_index >= 0)
2167 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2168 if (rtpctx2 && st && st2 &&
2169 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2170 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2171 rtpctx2->rtcp_ts_offset = av_rescale_q(
2172 rtpctx->rtcp_ts_offset, st->time_base,
2177 if (ret == -RTCP_BYE) {
2180 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2181 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2183 if (rt->nb_byes == rt->nb_rtsp_streams)
2187 } else if (CONFIG_RTPDEC && rt->ts) {
2188 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2191 rt->recvbuf_len = len;
2192 rt->recvbuf_pos = ret;
2193 rt->cur_transport_priv = rt->ts;
2200 return AVERROR_INVALIDDATA;
2206 /* more packets may follow, so we save the RTP context */
2207 rt->cur_transport_priv = rtsp_st->transport_priv;
2211 #endif /* CONFIG_RTPDEC */
2213 #if CONFIG_SDP_DEMUXER
2214 static int sdp_probe(AVProbeData *p1)
2216 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2218 /* we look for a line beginning "c=IN IP" */
2219 while (p < p_end && *p != '\0') {
2220 if (p + sizeof("c=IN IP") - 1 < p_end &&
2221 av_strstart(p, "c=IN IP", NULL))
2222 return AVPROBE_SCORE_EXTENSION;
2224 while (p < p_end - 1 && *p != '\n') p++;
2233 static void append_source_addrs(char *buf, int size, const char *name,
2234 int count, struct RTSPSource **addrs)
2239 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2240 for (i = 1; i < count; i++)
2241 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2244 static int sdp_read_header(AVFormatContext *s)
2246 RTSPState *rt = s->priv_data;
2247 RTSPStream *rtsp_st;
2252 if (!ff_network_init())
2253 return AVERROR(EIO);
2255 if (!rt->protocols) {
2256 rt->protocols = ffurl_get_protocols(s->protocol_whitelist,
2257 s->protocol_blacklist);
2259 return AVERROR(ENOMEM);
2262 if (s->max_delay < 0) /* Not set by the caller */
2263 s->max_delay = DEFAULT_REORDERING_DELAY;
2264 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2265 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2267 /* read the whole sdp file */
2268 /* XXX: better loading */
2269 content = av_malloc(SDP_MAX_SIZE);
2271 return AVERROR(ENOMEM);
2272 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2275 return AVERROR_INVALIDDATA;
2277 content[size] ='\0';
2279 err = ff_sdp_parse(s, content);
2283 /* open each RTP stream */
2284 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2286 rtsp_st = rt->rtsp_streams[i];
2288 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2289 AVDictionary *opts = map_to_opts(rt);
2291 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2292 sizeof(rtsp_st->sdp_ip),
2293 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2295 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2297 av_dict_free(&opts);
2300 ff_url_join(url, sizeof(url), "rtp", NULL,
2301 namebuf, rtsp_st->sdp_port,
2302 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2303 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2304 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2305 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2307 append_source_addrs(url, sizeof(url), "sources",
2308 rtsp_st->nb_include_source_addrs,
2309 rtsp_st->include_source_addrs);
2310 append_source_addrs(url, sizeof(url), "block",
2311 rtsp_st->nb_exclude_source_addrs,
2312 rtsp_st->exclude_source_addrs);
2313 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2314 &s->interrupt_callback, &opts, rt->protocols, NULL);
2316 av_dict_free(&opts);
2319 err = AVERROR_INVALIDDATA;
2323 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2328 ff_rtsp_close_streams(s);
2333 static int sdp_read_close(AVFormatContext *s)
2335 ff_rtsp_close_streams(s);
2340 static const AVClass sdp_demuxer_class = {
2341 .class_name = "SDP demuxer",
2342 .item_name = av_default_item_name,
2343 .option = sdp_options,
2344 .version = LIBAVUTIL_VERSION_INT,
2347 AVInputFormat ff_sdp_demuxer = {
2349 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2350 .priv_data_size = sizeof(RTSPState),
2351 .read_probe = sdp_probe,
2352 .read_header = sdp_read_header,
2353 .read_packet = ff_rtsp_fetch_packet,
2354 .read_close = sdp_read_close,
2355 .priv_class = &sdp_demuxer_class,
2357 #endif /* CONFIG_SDP_DEMUXER */
2359 #if CONFIG_RTP_DEMUXER
2360 static int rtp_probe(AVProbeData *p)
2362 if (av_strstart(p->filename, "rtp:", NULL))
2363 return AVPROBE_SCORE_MAX;
2367 static int rtp_read_header(AVFormatContext *s)
2369 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2370 char host[500], sdp[500];
2372 URLContext* in = NULL;
2374 AVCodecParameters *par = NULL;
2375 struct sockaddr_storage addr;
2377 socklen_t addrlen = sizeof(addr);
2378 RTSPState *rt = s->priv_data;
2380 if (!ff_network_init())
2381 return AVERROR(EIO);
2383 if (!rt->protocols) {
2384 rt->protocols = ffurl_get_protocols(s->protocol_whitelist,
2385 s->protocol_blacklist);
2387 return AVERROR(ENOMEM);
2390 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2391 &s->interrupt_callback, NULL, rt->protocols, NULL);
2396 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2397 if (ret == AVERROR(EAGAIN))
2402 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2406 if ((recvbuf[0] & 0xc0) != 0x80) {
2407 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2412 if (RTP_PT_IS_RTCP(recvbuf[1]))
2415 payload_type = recvbuf[1] & 0x7f;
2418 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2422 par = avcodec_parameters_alloc();
2424 ret = AVERROR(ENOMEM);
2428 if (ff_rtp_get_codec_info(par, payload_type)) {
2429 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2430 "without an SDP file describing it\n",
2434 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2435 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2436 "properly you need an SDP file "
2440 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2441 NULL, 0, s->filename);
2443 snprintf(sdp, sizeof(sdp),
2444 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2445 addr.ss_family == AF_INET ? 4 : 6, host,
2446 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2447 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2448 port, payload_type);
2449 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2450 avcodec_parameters_free(&par);
2452 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2455 /* sdp_read_header initializes this again */
2458 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2460 ret = sdp_read_header(s);
2465 avcodec_parameters_free(&par);
2472 static const AVClass rtp_demuxer_class = {
2473 .class_name = "RTP demuxer",
2474 .item_name = av_default_item_name,
2475 .option = rtp_options,
2476 .version = LIBAVUTIL_VERSION_INT,
2479 AVInputFormat ff_rtp_demuxer = {
2481 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2482 .priv_data_size = sizeof(RTSPState),
2483 .read_probe = rtp_probe,
2484 .read_header = rtp_read_header,
2485 .read_packet = ff_rtsp_fetch_packet,
2486 .read_close = sdp_read_close,
2487 .flags = AVFMT_NOFILE,
2488 .priv_class = &rtp_demuxer_class,
2490 #endif /* CONFIG_RTP_DEMUXER */