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"
31 #include "avio_internal.h"
38 #include "os_support.h"
44 #include "rtpdec_formats.h"
45 #include "rtpenc_chain.h"
51 /* Timeout values for socket poll, in ms,
52 * and read_packet(), in seconds */
53 #define POLL_TIMEOUT_MS 100
54 #define READ_PACKET_TIMEOUT_S 10
55 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
56 #define SDP_MAX_SIZE 16384
57 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
58 #define DEFAULT_REORDERING_DELAY 100000
60 #define OFFSET(x) offsetof(RTSPState, x)
61 #define DEC AV_OPT_FLAG_DECODING_PARAM
62 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 #define RTSP_FLAG_OPTS(name, longname) \
65 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
66 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
67 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 const AVOption ff_rtsp_options[] = {
76 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
77 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
78 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
79 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
80 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
81 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
82 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
83 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
84 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
85 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
86 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
87 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {-1}, INT_MIN, INT_MAX, DEC },
91 static const AVOption sdp_options[] = {
92 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
97 static const AVOption rtp_options[] = {
98 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
102 static void get_word_until_chars(char *buf, int buf_size,
103 const char *sep, const char **pp)
109 p += strspn(p, SPACE_CHARS);
111 while (!strchr(sep, *p) && *p != '\0') {
112 if ((q - buf) < buf_size - 1)
121 static void get_word_sep(char *buf, int buf_size, const char *sep,
124 if (**pp == '/') (*pp)++;
125 get_word_until_chars(buf, buf_size, sep, pp);
128 static void get_word(char *buf, int buf_size, const char **pp)
130 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
133 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
135 * Used for seeking in the rtp stream.
137 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
141 p += strspn(p, SPACE_CHARS);
142 if (!av_stristart(p, "npt=", &p))
145 *start = AV_NOPTS_VALUE;
146 *end = AV_NOPTS_VALUE;
148 get_word_sep(buf, sizeof(buf), "-", &p);
149 av_parse_time(start, buf, 1);
152 get_word_sep(buf, sizeof(buf), "-", &p);
153 av_parse_time(end, buf, 1);
155 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
156 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
159 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
161 struct addrinfo hints = { 0 }, *ai = NULL;
162 hints.ai_flags = AI_NUMERICHOST;
163 if (getaddrinfo(buf, NULL, &hints, &ai))
165 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
171 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
172 RTSPStream *rtsp_st, AVCodecContext *codec)
176 codec->codec_id = handler->codec_id;
177 rtsp_st->dynamic_handler = handler;
178 if (handler->alloc) {
179 rtsp_st->dynamic_protocol_context = handler->alloc();
180 if (!rtsp_st->dynamic_protocol_context)
181 rtsp_st->dynamic_handler = NULL;
185 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
186 static int sdp_parse_rtpmap(AVFormatContext *s,
187 AVStream *st, RTSPStream *rtsp_st,
188 int payload_type, const char *p)
190 AVCodecContext *codec = st->codec;
196 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
197 * see if we can handle this kind of payload.
198 * The space should normally not be there but some Real streams or
199 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
200 * have a trailing space. */
201 get_word_sep(buf, sizeof(buf), "/ ", &p);
202 if (payload_type < RTP_PT_PRIVATE) {
203 /* We are in a standard case
204 * (from http://www.iana.org/assignments/rtp-parameters). */
205 /* search into AVRtpPayloadTypes[] */
206 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
209 if (codec->codec_id == CODEC_ID_NONE) {
210 RTPDynamicProtocolHandler *handler =
211 ff_rtp_handler_find_by_name(buf, codec->codec_type);
212 init_rtp_handler(handler, rtsp_st, codec);
213 /* If no dynamic handler was found, check with the list of standard
214 * allocated types, if such a stream for some reason happens to
215 * use a private payload type. This isn't handled in rtpdec.c, since
216 * the format name from the rtpmap line never is passed into rtpdec. */
217 if (!rtsp_st->dynamic_handler)
218 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
221 c = avcodec_find_decoder(codec->codec_id);
227 get_word_sep(buf, sizeof(buf), "/", &p);
229 switch (codec->codec_type) {
230 case AVMEDIA_TYPE_AUDIO:
231 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
232 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
233 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
235 codec->sample_rate = i;
236 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
237 get_word_sep(buf, sizeof(buf), "/", &p);
241 // TODO: there is a bug here; if it is a mono stream, and
242 // less than 22000Hz, faad upconverts to stereo and twice
243 // the frequency. No problem, but the sample rate is being
244 // set here by the sdp line. Patch on its way. (rdm)
246 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
248 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
251 case AVMEDIA_TYPE_VIDEO:
252 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
254 avpriv_set_pts_info(st, 32, 1, i);
259 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
260 rtsp_st->dynamic_handler->init(s, st->index,
261 rtsp_st->dynamic_protocol_context);
265 /* parse the attribute line from the fmtp a line of an sdp response. This
266 * is broken out as a function because it is used in rtp_h264.c, which is
268 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
269 char *value, int value_size)
271 *p += strspn(*p, SPACE_CHARS);
273 get_word_sep(attr, attr_size, "=", p);
276 get_word_sep(value, value_size, ";", p);
284 typedef struct SDPParseState {
286 struct sockaddr_storage default_ip;
288 int skip_media; ///< set if an unknown m= line occurs
291 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
292 int letter, const char *buf)
294 RTSPState *rt = s->priv_data;
295 char buf1[64], st_type[64];
297 enum AVMediaType codec_type;
301 struct sockaddr_storage sdp_ip;
304 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
307 if (s1->skip_media && letter != 'm')
311 get_word(buf1, sizeof(buf1), &p);
312 if (strcmp(buf1, "IN") != 0)
314 get_word(buf1, sizeof(buf1), &p);
315 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
317 get_word_sep(buf1, sizeof(buf1), "/", &p);
318 if (get_sockaddr(buf1, &sdp_ip))
323 get_word_sep(buf1, sizeof(buf1), "/", &p);
326 if (s->nb_streams == 0) {
327 s1->default_ip = sdp_ip;
328 s1->default_ttl = ttl;
330 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
331 rtsp_st->sdp_ip = sdp_ip;
332 rtsp_st->sdp_ttl = ttl;
336 av_dict_set(&s->metadata, "title", p, 0);
339 if (s->nb_streams == 0) {
340 av_dict_set(&s->metadata, "comment", p, 0);
347 codec_type = AVMEDIA_TYPE_UNKNOWN;
348 get_word(st_type, sizeof(st_type), &p);
349 if (!strcmp(st_type, "audio")) {
350 codec_type = AVMEDIA_TYPE_AUDIO;
351 } else if (!strcmp(st_type, "video")) {
352 codec_type = AVMEDIA_TYPE_VIDEO;
353 } else if (!strcmp(st_type, "application")) {
354 codec_type = AVMEDIA_TYPE_DATA;
356 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
360 rtsp_st = av_mallocz(sizeof(RTSPStream));
363 rtsp_st->stream_index = -1;
364 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
366 rtsp_st->sdp_ip = s1->default_ip;
367 rtsp_st->sdp_ttl = s1->default_ttl;
369 get_word(buf1, sizeof(buf1), &p); /* port */
370 rtsp_st->sdp_port = atoi(buf1);
372 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
374 /* XXX: handle list of formats */
375 get_word(buf1, sizeof(buf1), &p); /* format list */
376 rtsp_st->sdp_payload_type = atoi(buf1);
378 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
379 /* no corresponding stream */
380 } else if (rt->server_type == RTSP_SERVER_WMS &&
381 codec_type == AVMEDIA_TYPE_DATA) {
382 /* RTX stream, a stream that carries all the other actual
383 * audio/video streams. Don't expose this to the callers. */
385 st = avformat_new_stream(s, NULL);
388 st->id = rt->nb_rtsp_streams - 1;
389 rtsp_st->stream_index = st->index;
390 st->codec->codec_type = codec_type;
391 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
392 RTPDynamicProtocolHandler *handler;
393 /* if standard payload type, we can find the codec right now */
394 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
395 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
396 st->codec->sample_rate > 0)
397 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
398 /* Even static payload types may need a custom depacketizer */
399 handler = ff_rtp_handler_find_by_id(
400 rtsp_st->sdp_payload_type, st->codec->codec_type);
401 init_rtp_handler(handler, rtsp_st, st->codec);
402 if (handler && handler->init)
403 handler->init(s, st->index,
404 rtsp_st->dynamic_protocol_context);
407 /* put a default control url */
408 av_strlcpy(rtsp_st->control_url, rt->control_uri,
409 sizeof(rtsp_st->control_url));
412 if (av_strstart(p, "control:", &p)) {
413 if (s->nb_streams == 0) {
414 if (!strncmp(p, "rtsp://", 7))
415 av_strlcpy(rt->control_uri, p,
416 sizeof(rt->control_uri));
419 /* get the control url */
420 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
422 /* XXX: may need to add full url resolution */
423 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
425 if (proto[0] == '\0') {
426 /* relative control URL */
427 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
428 av_strlcat(rtsp_st->control_url, "/",
429 sizeof(rtsp_st->control_url));
430 av_strlcat(rtsp_st->control_url, p,
431 sizeof(rtsp_st->control_url));
433 av_strlcpy(rtsp_st->control_url, p,
434 sizeof(rtsp_st->control_url));
436 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
437 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
438 get_word(buf1, sizeof(buf1), &p);
439 payload_type = atoi(buf1);
440 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
441 if (rtsp_st->stream_index >= 0) {
442 st = s->streams[rtsp_st->stream_index];
443 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
445 } else if (av_strstart(p, "fmtp:", &p) ||
446 av_strstart(p, "framesize:", &p)) {
447 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
448 // let dynamic protocol handlers have a stab at the line.
449 get_word(buf1, sizeof(buf1), &p);
450 payload_type = atoi(buf1);
451 for (i = 0; i < rt->nb_rtsp_streams; i++) {
452 rtsp_st = rt->rtsp_streams[i];
453 if (rtsp_st->sdp_payload_type == payload_type &&
454 rtsp_st->dynamic_handler &&
455 rtsp_st->dynamic_handler->parse_sdp_a_line)
456 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
457 rtsp_st->dynamic_protocol_context, buf);
459 } else if (av_strstart(p, "range:", &p)) {
462 // this is so that seeking on a streamed file can work.
463 rtsp_parse_range_npt(p, &start, &end);
464 s->start_time = start;
465 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
466 s->duration = (end == AV_NOPTS_VALUE) ?
467 AV_NOPTS_VALUE : end - start;
468 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
470 rt->transport = RTSP_TRANSPORT_RDT;
471 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
473 st = s->streams[s->nb_streams - 1];
474 st->codec->sample_rate = atoi(p);
476 if (rt->server_type == RTSP_SERVER_WMS)
477 ff_wms_parse_sdp_a_line(s, p);
478 if (s->nb_streams > 0) {
479 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
481 if (rt->server_type == RTSP_SERVER_REAL)
482 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
484 if (rtsp_st->dynamic_handler &&
485 rtsp_st->dynamic_handler->parse_sdp_a_line)
486 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
487 rtsp_st->stream_index,
488 rtsp_st->dynamic_protocol_context, buf);
495 int ff_sdp_parse(AVFormatContext *s, const char *content)
497 RTSPState *rt = s->priv_data;
500 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
501 * contain long SDP lines containing complete ASF Headers (several
502 * kB) or arrays of MDPR (RM stream descriptor) headers plus
503 * "rulebooks" describing their properties. Therefore, the SDP line
506 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
507 * in rtpdec_xiph.c. */
509 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
513 p += strspn(p, SPACE_CHARS);
521 /* get the content */
523 while (*p != '\n' && *p != '\r' && *p != '\0') {
524 if ((q - buf) < sizeof(buf) - 1)
529 sdp_parse_line(s, s1, letter, buf);
531 while (*p != '\n' && *p != '\0')
536 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
537 if (!rt->p) return AVERROR(ENOMEM);
540 #endif /* CONFIG_RTPDEC */
542 void ff_rtsp_undo_setup(AVFormatContext *s)
544 RTSPState *rt = s->priv_data;
547 for (i = 0; i < rt->nb_rtsp_streams; i++) {
548 RTSPStream *rtsp_st = rt->rtsp_streams[i];
551 if (rtsp_st->transport_priv) {
553 AVFormatContext *rtpctx = rtsp_st->transport_priv;
554 av_write_trailer(rtpctx);
555 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
557 avio_close_dyn_buf(rtpctx->pb, &ptr);
560 avio_close(rtpctx->pb);
562 avformat_free_context(rtpctx);
563 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
564 ff_rdt_parse_close(rtsp_st->transport_priv);
565 else if (CONFIG_RTPDEC)
566 ff_rtp_parse_close(rtsp_st->transport_priv);
568 rtsp_st->transport_priv = NULL;
569 if (rtsp_st->rtp_handle)
570 ffurl_close(rtsp_st->rtp_handle);
571 rtsp_st->rtp_handle = NULL;
575 /* close and free RTSP streams */
576 void ff_rtsp_close_streams(AVFormatContext *s)
578 RTSPState *rt = s->priv_data;
582 ff_rtsp_undo_setup(s);
583 for (i = 0; i < rt->nb_rtsp_streams; i++) {
584 rtsp_st = rt->rtsp_streams[i];
586 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
587 rtsp_st->dynamic_handler->free(
588 rtsp_st->dynamic_protocol_context);
592 av_free(rt->rtsp_streams);
594 avformat_close_input(&rt->asf_ctx);
597 av_free(rt->recvbuf);
600 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
602 RTSPState *rt = s->priv_data;
605 /* open the RTP context */
606 if (rtsp_st->stream_index >= 0)
607 st = s->streams[rtsp_st->stream_index];
609 s->ctx_flags |= AVFMTCTX_NOHEADER;
611 if (s->oformat && CONFIG_RTSP_MUXER) {
612 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
614 RTSP_TCP_MAX_PACKET_SIZE);
615 /* Ownership of rtp_handle is passed to the rtp mux context */
616 rtsp_st->rtp_handle = NULL;
619 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
620 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
621 rtsp_st->dynamic_protocol_context,
622 rtsp_st->dynamic_handler);
623 else if (CONFIG_RTPDEC)
624 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
625 rtsp_st->sdp_payload_type,
626 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
627 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
629 if (!rtsp_st->transport_priv) {
630 return AVERROR(ENOMEM);
631 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
632 if (rtsp_st->dynamic_handler) {
633 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
634 rtsp_st->dynamic_protocol_context,
635 rtsp_st->dynamic_handler);
642 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
643 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
650 q += strspn(q, SPACE_CHARS);
651 v = strtol(q, &p, 10);
655 v = strtol(p, &p, 10);
664 /* XXX: only one transport specification is parsed */
665 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
667 char transport_protocol[16];
669 char lower_transport[16];
671 RTSPTransportField *th;
674 reply->nb_transports = 0;
677 p += strspn(p, SPACE_CHARS);
681 th = &reply->transports[reply->nb_transports];
683 get_word_sep(transport_protocol, sizeof(transport_protocol),
685 if (!av_strcasecmp (transport_protocol, "rtp")) {
686 get_word_sep(profile, sizeof(profile), "/;,", &p);
687 lower_transport[0] = '\0';
688 /* rtp/avp/<protocol> */
690 get_word_sep(lower_transport, sizeof(lower_transport),
693 th->transport = RTSP_TRANSPORT_RTP;
694 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
695 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
696 /* x-pn-tng/<protocol> */
697 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
699 th->transport = RTSP_TRANSPORT_RDT;
701 if (!av_strcasecmp(lower_transport, "TCP"))
702 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
704 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
708 /* get each parameter */
709 while (*p != '\0' && *p != ',') {
710 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
711 if (!strcmp(parameter, "port")) {
714 rtsp_parse_range(&th->port_min, &th->port_max, &p);
716 } else if (!strcmp(parameter, "client_port")) {
719 rtsp_parse_range(&th->client_port_min,
720 &th->client_port_max, &p);
722 } else if (!strcmp(parameter, "server_port")) {
725 rtsp_parse_range(&th->server_port_min,
726 &th->server_port_max, &p);
728 } else if (!strcmp(parameter, "interleaved")) {
731 rtsp_parse_range(&th->interleaved_min,
732 &th->interleaved_max, &p);
734 } else if (!strcmp(parameter, "multicast")) {
735 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
736 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
737 } else if (!strcmp(parameter, "ttl")) {
740 th->ttl = strtol(p, (char **)&p, 10);
742 } else if (!strcmp(parameter, "destination")) {
745 get_word_sep(buf, sizeof(buf), ";,", &p);
746 get_sockaddr(buf, &th->destination);
748 } else if (!strcmp(parameter, "source")) {
751 get_word_sep(buf, sizeof(buf), ";,", &p);
752 av_strlcpy(th->source, buf, sizeof(th->source));
754 } else if (!strcmp(parameter, "mode")) {
757 get_word_sep(buf, sizeof(buf), ";, ", &p);
758 if (!strcmp(buf, "record") ||
759 !strcmp(buf, "receive"))
764 while (*p != ';' && *p != '\0' && *p != ',')
772 reply->nb_transports++;
776 static void handle_rtp_info(RTSPState *rt, const char *url,
777 uint32_t seq, uint32_t rtptime)
780 if (!rtptime || !url[0])
782 if (rt->transport != RTSP_TRANSPORT_RTP)
784 for (i = 0; i < rt->nb_rtsp_streams; i++) {
785 RTSPStream *rtsp_st = rt->rtsp_streams[i];
786 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
789 if (!strcmp(rtsp_st->control_url, url)) {
790 rtpctx->base_timestamp = rtptime;
796 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
799 char key[20], value[1024], url[1024] = "";
800 uint32_t seq = 0, rtptime = 0;
803 p += strspn(p, SPACE_CHARS);
806 get_word_sep(key, sizeof(key), "=", &p);
810 get_word_sep(value, sizeof(value), ";, ", &p);
812 if (!strcmp(key, "url"))
813 av_strlcpy(url, value, sizeof(url));
814 else if (!strcmp(key, "seq"))
815 seq = strtoul(value, NULL, 10);
816 else if (!strcmp(key, "rtptime"))
817 rtptime = strtoul(value, NULL, 10);
819 handle_rtp_info(rt, url, seq, rtptime);
828 handle_rtp_info(rt, url, seq, rtptime);
831 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
832 RTSPState *rt, const char *method)
836 /* NOTE: we do case independent match for broken servers */
838 if (av_stristart(p, "Session:", &p)) {
840 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
841 if (av_stristart(p, ";timeout=", &p) &&
842 (t = strtol(p, NULL, 10)) > 0) {
845 } else if (av_stristart(p, "Content-Length:", &p)) {
846 reply->content_length = strtol(p, NULL, 10);
847 } else if (av_stristart(p, "Transport:", &p)) {
848 rtsp_parse_transport(reply, p);
849 } else if (av_stristart(p, "CSeq:", &p)) {
850 reply->seq = strtol(p, NULL, 10);
851 } else if (av_stristart(p, "Range:", &p)) {
852 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
853 } else if (av_stristart(p, "RealChallenge1:", &p)) {
854 p += strspn(p, SPACE_CHARS);
855 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
856 } else if (av_stristart(p, "Server:", &p)) {
857 p += strspn(p, SPACE_CHARS);
858 av_strlcpy(reply->server, p, sizeof(reply->server));
859 } else if (av_stristart(p, "Notice:", &p) ||
860 av_stristart(p, "X-Notice:", &p)) {
861 reply->notice = strtol(p, NULL, 10);
862 } else if (av_stristart(p, "Location:", &p)) {
863 p += strspn(p, SPACE_CHARS);
864 av_strlcpy(reply->location, p , sizeof(reply->location));
865 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
866 p += strspn(p, SPACE_CHARS);
867 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
868 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
869 p += strspn(p, SPACE_CHARS);
870 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
871 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
872 p += strspn(p, SPACE_CHARS);
873 if (method && !strcmp(method, "DESCRIBE"))
874 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
875 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
876 p += strspn(p, SPACE_CHARS);
877 if (method && !strcmp(method, "PLAY"))
878 rtsp_parse_rtp_info(rt, p);
879 } else if (av_stristart(p, "Public:", &p) && rt) {
880 if (strstr(p, "GET_PARAMETER") &&
881 method && !strcmp(method, "OPTIONS"))
882 rt->get_parameter_supported = 1;
883 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
884 p += strspn(p, SPACE_CHARS);
885 rt->accept_dynamic_rate = atoi(p);
886 } else if (av_stristart(p, "Content-Type:", &p)) {
887 p += strspn(p, SPACE_CHARS);
888 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
892 /* skip a RTP/TCP interleaved packet */
893 void ff_rtsp_skip_packet(AVFormatContext *s)
895 RTSPState *rt = s->priv_data;
899 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
902 len = AV_RB16(buf + 1);
904 av_dlog(s, "skipping RTP packet len=%d\n", len);
909 if (len1 > sizeof(buf))
911 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
918 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
919 unsigned char **content_ptr,
920 int return_on_interleaved_data, const char *method)
922 RTSPState *rt = s->priv_data;
923 char buf[4096], buf1[1024], *q;
926 int ret, content_length, line_count = 0, request = 0;
927 unsigned char *content = NULL;
933 memset(reply, 0, sizeof(*reply));
935 /* parse reply (XXX: use buffers) */
936 rt->last_reply[0] = '\0';
940 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
941 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
947 /* XXX: only parse it if first char on line ? */
948 if (return_on_interleaved_data) {
951 ff_rtsp_skip_packet(s);
952 } else if (ch != '\r') {
953 if ((q - buf) < sizeof(buf) - 1)
959 av_dlog(s, "line='%s'\n", buf);
961 /* test if last line */
965 if (line_count == 0) {
967 get_word(buf1, sizeof(buf1), &p);
968 if (!strncmp(buf1, "RTSP/", 5)) {
969 get_word(buf1, sizeof(buf1), &p);
970 reply->status_code = atoi(buf1);
971 av_strlcpy(reply->reason, p, sizeof(reply->reason));
973 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
974 get_word(buf1, sizeof(buf1), &p); // object
978 ff_rtsp_parse_line(reply, p, rt, method);
979 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
980 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
985 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
986 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
988 content_length = reply->content_length;
989 if (content_length > 0) {
990 /* leave some room for a trailing '\0' (useful for simple parsing) */
991 content = av_malloc(content_length + 1);
992 ffurl_read_complete(rt->rtsp_hd, content, content_length);
993 content[content_length] = '\0';
996 *content_ptr = content;
1002 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1003 const char* ptr = buf;
1005 if (!strcmp(reply->reason, "OPTIONS")) {
1006 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1008 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1009 if (reply->session_id[0])
1010 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1013 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1015 av_strlcat(buf, "\r\n", sizeof(buf));
1017 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1018 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1021 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1023 rt->last_cmd_time = av_gettime();
1024 /* Even if the request from the server had data, it is not the data
1025 * that the caller wants or expects. The memory could also be leaked
1026 * if the actual following reply has content data. */
1028 av_freep(content_ptr);
1029 /* If method is set, this is called from ff_rtsp_send_cmd,
1030 * where a reply to exactly this request is awaited. For
1031 * callers from within packet receiving, we just want to
1032 * return to the caller and go back to receiving packets. */
1038 if (rt->seq != reply->seq) {
1039 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1040 rt->seq, reply->seq);
1044 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1045 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1046 reply->notice == 2306 /* Continuous Feed Terminated */) {
1047 rt->state = RTSP_STATE_IDLE;
1048 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1049 return AVERROR(EIO); /* data or server error */
1050 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1051 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1052 return AVERROR(EPERM);
1058 * Send a command to the RTSP server without waiting for the reply.
1060 * @param s RTSP (de)muxer context
1061 * @param method the method for the request
1062 * @param url the target url for the request
1063 * @param headers extra header lines to include in the request
1064 * @param send_content if non-null, the data to send as request body content
1065 * @param send_content_length the length of the send_content data, or 0 if
1066 * send_content is null
1068 * @return zero if success, nonzero otherwise
1070 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1071 const char *method, const char *url,
1072 const char *headers,
1073 const unsigned char *send_content,
1074 int send_content_length)
1076 RTSPState *rt = s->priv_data;
1077 char buf[4096], *out_buf;
1078 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1080 /* Add in RTSP headers */
1083 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1085 av_strlcat(buf, headers, sizeof(buf));
1086 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1087 if (rt->session_id[0] != '\0' && (!headers ||
1088 !strstr(headers, "\nIf-Match:"))) {
1089 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1092 char *str = ff_http_auth_create_response(&rt->auth_state,
1093 rt->auth, url, method);
1095 av_strlcat(buf, str, sizeof(buf));
1098 if (send_content_length > 0 && send_content)
1099 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1100 av_strlcat(buf, "\r\n", sizeof(buf));
1102 /* base64 encode rtsp if tunneling */
1103 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1104 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1105 out_buf = base64buf;
1108 av_dlog(s, "Sending:\n%s--\n", buf);
1110 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1111 if (send_content_length > 0 && send_content) {
1112 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1113 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1114 "with content data not supported\n");
1115 return AVERROR_PATCHWELCOME;
1117 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1119 rt->last_cmd_time = av_gettime();
1124 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1125 const char *url, const char *headers)
1127 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1130 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1131 const char *headers, RTSPMessageHeader *reply,
1132 unsigned char **content_ptr)
1134 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1135 content_ptr, NULL, 0);
1138 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1139 const char *method, const char *url,
1141 RTSPMessageHeader *reply,
1142 unsigned char **content_ptr,
1143 const unsigned char *send_content,
1144 int send_content_length)
1146 RTSPState *rt = s->priv_data;
1147 HTTPAuthType cur_auth_type;
1148 int ret, attempts = 0;
1151 cur_auth_type = rt->auth_state.auth_type;
1152 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1154 send_content_length)))
1157 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1161 if (reply->status_code == 401 &&
1162 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1163 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1166 if (reply->status_code > 400){
1167 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1171 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1177 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1178 int lower_transport, const char *real_challenge)
1180 RTSPState *rt = s->priv_data;
1181 int rtx = 0, j, i, err, interleave = 0, port_off;
1182 RTSPStream *rtsp_st;
1183 RTSPMessageHeader reply1, *reply = &reply1;
1185 const char *trans_pref;
1187 if (rt->transport == RTSP_TRANSPORT_RDT)
1188 trans_pref = "x-pn-tng";
1190 trans_pref = "RTP/AVP";
1192 /* default timeout: 1 minute */
1195 /* for each stream, make the setup request */
1196 /* XXX: we assume the same server is used for the control of each
1199 /* Choose a random starting offset within the first half of the
1200 * port range, to allow for a number of ports to try even if the offset
1201 * happens to be at the end of the random range. */
1202 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1203 /* even random offset */
1204 port_off -= port_off & 0x01;
1206 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1207 char transport[2048];
1210 * WMS serves all UDP data over a single connection, the RTX, which
1211 * isn't necessarily the first in the SDP but has to be the first
1212 * to be set up, else the second/third SETUP will fail with a 461.
1214 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1215 rt->server_type == RTSP_SERVER_WMS) {
1218 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1219 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1221 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1225 if (rtx == rt->nb_rtsp_streams)
1226 return -1; /* no RTX found */
1227 rtsp_st = rt->rtsp_streams[rtx];
1229 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1231 rtsp_st = rt->rtsp_streams[i];
1234 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1237 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1238 port = reply->transports[0].client_port_min;
1242 /* first try in specified port range */
1243 while (j <= rt->rtp_port_max) {
1244 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1245 "?localport=%d", j);
1246 /* we will use two ports per rtp stream (rtp and rtcp) */
1248 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1249 &s->interrupt_callback, NULL))
1253 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1258 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1260 snprintf(transport, sizeof(transport) - 1,
1261 "%s/UDP;", trans_pref);
1262 if (rt->server_type != RTSP_SERVER_REAL)
1263 av_strlcat(transport, "unicast;", sizeof(transport));
1264 av_strlcatf(transport, sizeof(transport),
1265 "client_port=%d", port);
1266 if (rt->transport == RTSP_TRANSPORT_RTP &&
1267 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1268 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1272 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1273 /* For WMS streams, the application streams are only used for
1274 * UDP. When trying to set it up for TCP streams, the server
1275 * will return an error. Therefore, we skip those streams. */
1276 if (rt->server_type == RTSP_SERVER_WMS &&
1277 (rtsp_st->stream_index < 0 ||
1278 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1281 snprintf(transport, sizeof(transport) - 1,
1282 "%s/TCP;", trans_pref);
1283 if (rt->transport != RTSP_TRANSPORT_RDT)
1284 av_strlcat(transport, "unicast;", sizeof(transport));
1285 av_strlcatf(transport, sizeof(transport),
1286 "interleaved=%d-%d",
1287 interleave, interleave + 1);
1291 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1292 snprintf(transport, sizeof(transport) - 1,
1293 "%s/UDP;multicast", trans_pref);
1296 av_strlcat(transport, ";mode=record", sizeof(transport));
1297 } else if (rt->server_type == RTSP_SERVER_REAL ||
1298 rt->server_type == RTSP_SERVER_WMS)
1299 av_strlcat(transport, ";mode=play", sizeof(transport));
1300 snprintf(cmd, sizeof(cmd),
1301 "Transport: %s\r\n",
1303 if (rt->accept_dynamic_rate)
1304 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1305 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1306 char real_res[41], real_csum[9];
1307 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1309 av_strlcatf(cmd, sizeof(cmd),
1311 "RealChallenge2: %s, sd=%s\r\n",
1312 rt->session_id, real_res, real_csum);
1314 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1315 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1318 } else if (reply->status_code != RTSP_STATUS_OK ||
1319 reply->nb_transports != 1) {
1320 err = AVERROR_INVALIDDATA;
1324 /* XXX: same protocol for all streams is required */
1326 if (reply->transports[0].lower_transport != rt->lower_transport ||
1327 reply->transports[0].transport != rt->transport) {
1328 err = AVERROR_INVALIDDATA;
1332 rt->lower_transport = reply->transports[0].lower_transport;
1333 rt->transport = reply->transports[0].transport;
1336 /* Fail if the server responded with another lower transport mode
1337 * than what we requested. */
1338 if (reply->transports[0].lower_transport != lower_transport) {
1339 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1340 err = AVERROR_INVALIDDATA;
1344 switch(reply->transports[0].lower_transport) {
1345 case RTSP_LOWER_TRANSPORT_TCP:
1346 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1347 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1350 case RTSP_LOWER_TRANSPORT_UDP: {
1351 char url[1024], options[30] = "";
1353 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1354 av_strlcpy(options, "?connect=1", sizeof(options));
1355 /* Use source address if specified */
1356 if (reply->transports[0].source[0]) {
1357 ff_url_join(url, sizeof(url), "rtp", NULL,
1358 reply->transports[0].source,
1359 reply->transports[0].server_port_min, "%s", options);
1361 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1362 reply->transports[0].server_port_min, "%s", options);
1364 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1365 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1366 err = AVERROR_INVALIDDATA;
1369 /* Try to initialize the connection state in a
1370 * potential NAT router by sending dummy packets.
1371 * RTP/RTCP dummy packets are used for RDT, too.
1373 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1375 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1378 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1379 char url[1024], namebuf[50], optbuf[20] = "";
1380 struct sockaddr_storage addr;
1383 if (reply->transports[0].destination.ss_family) {
1384 addr = reply->transports[0].destination;
1385 port = reply->transports[0].port_min;
1386 ttl = reply->transports[0].ttl;
1388 addr = rtsp_st->sdp_ip;
1389 port = rtsp_st->sdp_port;
1390 ttl = rtsp_st->sdp_ttl;
1393 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1394 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1395 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1396 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1397 port, "%s", optbuf);
1398 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1399 &s->interrupt_callback, NULL) < 0) {
1400 err = AVERROR_INVALIDDATA;
1407 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1411 if (rt->nb_rtsp_streams && reply->timeout > 0)
1412 rt->timeout = reply->timeout;
1414 if (rt->server_type == RTSP_SERVER_REAL)
1415 rt->need_subscription = 1;
1420 ff_rtsp_undo_setup(s);
1424 void ff_rtsp_close_connections(AVFormatContext *s)
1426 RTSPState *rt = s->priv_data;
1427 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1428 ffurl_close(rt->rtsp_hd);
1429 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1432 int ff_rtsp_connect(AVFormatContext *s)
1434 RTSPState *rt = s->priv_data;
1435 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1436 int port, err, tcp_fd;
1437 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1438 int lower_transport_mask = 0;
1439 char real_challenge[64] = "";
1440 struct sockaddr_storage peer;
1441 socklen_t peer_len = sizeof(peer);
1443 if (rt->rtp_port_max < rt->rtp_port_min) {
1444 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1445 "than min port %d\n", rt->rtp_port_max,
1447 return AVERROR(EINVAL);
1450 if (!ff_network_init())
1451 return AVERROR(EIO);
1453 if (s->max_delay < 0) /* Not set by the caller */
1454 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1456 rt->control_transport = RTSP_MODE_PLAIN;
1457 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1458 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1459 rt->control_transport = RTSP_MODE_TUNNEL;
1461 /* Only pass through valid flags from here */
1462 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1465 lower_transport_mask = rt->lower_transport_mask;
1466 /* extract hostname and port */
1467 av_url_split(NULL, 0, auth, sizeof(auth),
1468 host, sizeof(host), &port, path, sizeof(path), s->filename);
1470 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1473 port = RTSP_DEFAULT_PORT;
1475 if (!lower_transport_mask)
1476 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1479 /* Only UDP or TCP - UDP multicast isn't supported. */
1480 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1481 (1 << RTSP_LOWER_TRANSPORT_TCP);
1482 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1483 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1484 "only UDP and TCP are supported for output.\n");
1485 err = AVERROR(EINVAL);
1490 /* Construct the URI used in request; this is similar to s->filename,
1491 * but with authentication credentials removed and RTSP specific options
1493 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1494 host, port, "%s", path);
1496 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1497 /* set up initial handshake for tunneling */
1498 char httpname[1024];
1499 char sessioncookie[17];
1502 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1503 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1504 av_get_random_seed(), av_get_random_seed());
1507 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1508 &s->interrupt_callback) < 0) {
1513 /* generate GET headers */
1514 snprintf(headers, sizeof(headers),
1515 "x-sessioncookie: %s\r\n"
1516 "Accept: application/x-rtsp-tunnelled\r\n"
1517 "Pragma: no-cache\r\n"
1518 "Cache-Control: no-cache\r\n",
1520 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1522 /* complete the connection */
1523 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1529 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1530 &s->interrupt_callback) < 0 ) {
1535 /* generate POST headers */
1536 snprintf(headers, sizeof(headers),
1537 "x-sessioncookie: %s\r\n"
1538 "Content-Type: application/x-rtsp-tunnelled\r\n"
1539 "Pragma: no-cache\r\n"
1540 "Cache-Control: no-cache\r\n"
1541 "Content-Length: 32767\r\n"
1542 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1544 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1545 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1547 /* Initialize the authentication state for the POST session. The HTTP
1548 * protocol implementation doesn't properly handle multi-pass
1549 * authentication for POST requests, since it would require one of
1551 * - implementing Expect: 100-continue, which many HTTP servers
1552 * don't support anyway, even less the RTSP servers that do HTTP
1554 * - sending the whole POST data until getting a 401 reply specifying
1555 * what authentication method to use, then resending all that data
1556 * - waiting for potential 401 replies directly after sending the
1557 * POST header (waiting for some unspecified time)
1558 * Therefore, we copy the full auth state, which works for both basic
1559 * and digest. (For digest, we would have to synchronize the nonce
1560 * count variable between the two sessions, if we'd do more requests
1561 * with the original session, though.)
1563 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1565 /* complete the connection */
1566 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1571 /* open the tcp connection */
1572 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1573 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1574 &s->interrupt_callback, NULL) < 0) {
1578 rt->rtsp_hd_out = rt->rtsp_hd;
1582 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1583 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1584 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1585 NULL, 0, NI_NUMERICHOST);
1588 /* request options supported by the server; this also detects server
1590 for (rt->server_type = RTSP_SERVER_RTP;;) {
1592 if (rt->server_type == RTSP_SERVER_REAL)
1595 * The following entries are required for proper
1596 * streaming from a Realmedia server. They are
1597 * interdependent in some way although we currently
1598 * don't quite understand how. Values were copied
1599 * from mplayer SVN r23589.
1600 * ClientChallenge is a 16-byte ID in hex
1601 * CompanyID is a 16-byte ID in base64
1603 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1604 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1605 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1606 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1608 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1609 if (reply->status_code != RTSP_STATUS_OK) {
1610 err = AVERROR_INVALIDDATA;
1614 /* detect server type if not standard-compliant RTP */
1615 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1616 rt->server_type = RTSP_SERVER_REAL;
1618 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1619 rt->server_type = RTSP_SERVER_WMS;
1620 } else if (rt->server_type == RTSP_SERVER_REAL)
1621 strcpy(real_challenge, reply->real_challenge);
1625 if (s->iformat && CONFIG_RTSP_DEMUXER)
1626 err = ff_rtsp_setup_input_streams(s, reply);
1627 else if (CONFIG_RTSP_MUXER)
1628 err = ff_rtsp_setup_output_streams(s, host);
1633 int lower_transport = ff_log2_tab[lower_transport_mask &
1634 ~(lower_transport_mask - 1)];
1636 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1637 rt->server_type == RTSP_SERVER_REAL ?
1638 real_challenge : NULL);
1641 lower_transport_mask &= ~(1 << lower_transport);
1642 if (lower_transport_mask == 0 && err == 1) {
1643 err = AVERROR(EPROTONOSUPPORT);
1648 rt->lower_transport_mask = lower_transport_mask;
1649 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1650 rt->state = RTSP_STATE_IDLE;
1651 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1654 ff_rtsp_close_streams(s);
1655 ff_rtsp_close_connections(s);
1656 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1657 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1658 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1666 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1669 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1670 uint8_t *buf, int buf_size, int64_t wait_end)
1672 RTSPState *rt = s->priv_data;
1673 RTSPStream *rtsp_st;
1674 int n, i, ret, tcp_fd, timeout_cnt = 0;
1676 struct pollfd *p = rt->p;
1679 if (ff_check_interrupt(&s->interrupt_callback))
1680 return AVERROR_EXIT;
1681 if (wait_end && wait_end - av_gettime() < 0)
1682 return AVERROR(EAGAIN);
1685 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1686 p[max_p].fd = tcp_fd;
1687 p[max_p++].events = POLLIN;
1691 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1692 rtsp_st = rt->rtsp_streams[i];
1693 if (rtsp_st->rtp_handle) {
1694 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1695 p[max_p++].events = POLLIN;
1696 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1697 p[max_p++].events = POLLIN;
1700 n = poll(p, max_p, POLL_TIMEOUT_MS);
1702 int j = 1 - (tcp_fd == -1);
1704 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1705 rtsp_st = rt->rtsp_streams[i];
1706 if (rtsp_st->rtp_handle) {
1707 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1708 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1710 *prtsp_st = rtsp_st;
1717 #if CONFIG_RTSP_DEMUXER
1718 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1719 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1720 if (rt->state == RTSP_STATE_STREAMING) {
1721 if (!ff_rtsp_parse_streaming_commands(s))
1724 av_log(s, AV_LOG_WARNING,
1725 "Unable to answer to TEARDOWN\n");
1729 RTSPMessageHeader reply;
1730 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1733 /* XXX: parse message */
1734 if (rt->state != RTSP_STATE_STREAMING)
1739 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1740 return AVERROR(ETIMEDOUT);
1741 } else if (n < 0 && errno != EINTR)
1742 return AVERROR(errno);
1746 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1748 RTSPState *rt = s->priv_data;
1750 RTSPStream *rtsp_st, *first_queue_st = NULL;
1751 int64_t wait_end = 0;
1753 if (rt->nb_byes == rt->nb_rtsp_streams)
1756 /* get next frames from the same RTP packet */
1757 if (rt->cur_transport_priv) {
1758 if (rt->transport == RTSP_TRANSPORT_RDT) {
1759 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1761 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1763 rt->cur_transport_priv = NULL;
1765 } else if (ret == 1) {
1768 rt->cur_transport_priv = NULL;
1771 if (rt->transport == RTSP_TRANSPORT_RTP) {
1773 int64_t first_queue_time = 0;
1774 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1775 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1779 queue_time = ff_rtp_queued_packet_time(rtpctx);
1780 if (queue_time && (queue_time - first_queue_time < 0 ||
1781 !first_queue_time)) {
1782 first_queue_time = queue_time;
1783 first_queue_st = rt->rtsp_streams[i];
1786 if (first_queue_time)
1787 wait_end = first_queue_time + s->max_delay;
1790 /* read next RTP packet */
1793 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1795 return AVERROR(ENOMEM);
1798 switch(rt->lower_transport) {
1800 #if CONFIG_RTSP_DEMUXER
1801 case RTSP_LOWER_TRANSPORT_TCP:
1802 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1805 case RTSP_LOWER_TRANSPORT_UDP:
1806 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1807 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1808 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1809 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1812 if (len == AVERROR(EAGAIN) && first_queue_st &&
1813 rt->transport == RTSP_TRANSPORT_RTP) {
1814 rtsp_st = first_queue_st;
1815 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1822 if (rt->transport == RTSP_TRANSPORT_RDT) {
1823 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1825 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1827 /* Either bad packet, or a RTCP packet. Check if the
1828 * first_rtcp_ntp_time field was initialized. */
1829 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1830 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1831 /* first_rtcp_ntp_time has been initialized for this stream,
1832 * copy the same value to all other uninitialized streams,
1833 * in order to map their timestamp origin to the same ntp time
1836 AVStream *st = NULL;
1837 if (rtsp_st->stream_index >= 0)
1838 st = s->streams[rtsp_st->stream_index];
1839 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1840 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1841 AVStream *st2 = NULL;
1842 if (rt->rtsp_streams[i]->stream_index >= 0)
1843 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1844 if (rtpctx2 && st && st2 &&
1845 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1846 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1847 rtpctx2->rtcp_ts_offset = av_rescale_q(
1848 rtpctx->rtcp_ts_offset, st->time_base,
1853 if (ret == -RTCP_BYE) {
1856 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1857 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1859 if (rt->nb_byes == rt->nb_rtsp_streams)
1868 /* more packets may follow, so we save the RTP context */
1869 rt->cur_transport_priv = rtsp_st->transport_priv;
1873 #endif /* CONFIG_RTPDEC */
1875 #if CONFIG_SDP_DEMUXER
1876 static int sdp_probe(AVProbeData *p1)
1878 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1880 /* we look for a line beginning "c=IN IP" */
1881 while (p < p_end && *p != '\0') {
1882 if (p + sizeof("c=IN IP") - 1 < p_end &&
1883 av_strstart(p, "c=IN IP", NULL))
1884 return AVPROBE_SCORE_MAX / 2;
1886 while (p < p_end - 1 && *p != '\n') p++;
1895 static int sdp_read_header(AVFormatContext *s)
1897 RTSPState *rt = s->priv_data;
1898 RTSPStream *rtsp_st;
1903 if (!ff_network_init())
1904 return AVERROR(EIO);
1906 if (s->max_delay < 0) /* Not set by the caller */
1907 s->max_delay = DEFAULT_REORDERING_DELAY;
1909 /* read the whole sdp file */
1910 /* XXX: better loading */
1911 content = av_malloc(SDP_MAX_SIZE);
1912 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1915 return AVERROR_INVALIDDATA;
1917 content[size] ='\0';
1919 err = ff_sdp_parse(s, content);
1923 /* open each RTP stream */
1924 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1926 rtsp_st = rt->rtsp_streams[i];
1928 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1929 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1930 ff_url_join(url, sizeof(url), "rtp", NULL,
1931 namebuf, rtsp_st->sdp_port,
1932 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1934 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1935 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1936 &s->interrupt_callback, NULL) < 0) {
1937 err = AVERROR_INVALIDDATA;
1940 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1945 ff_rtsp_close_streams(s);
1950 static int sdp_read_close(AVFormatContext *s)
1952 ff_rtsp_close_streams(s);
1957 static const AVClass sdp_demuxer_class = {
1958 .class_name = "SDP demuxer",
1959 .item_name = av_default_item_name,
1960 .option = sdp_options,
1961 .version = LIBAVUTIL_VERSION_INT,
1964 AVInputFormat ff_sdp_demuxer = {
1966 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1967 .priv_data_size = sizeof(RTSPState),
1968 .read_probe = sdp_probe,
1969 .read_header = sdp_read_header,
1970 .read_packet = ff_rtsp_fetch_packet,
1971 .read_close = sdp_read_close,
1972 .priv_class = &sdp_demuxer_class,
1974 #endif /* CONFIG_SDP_DEMUXER */
1976 #if CONFIG_RTP_DEMUXER
1977 static int rtp_probe(AVProbeData *p)
1979 if (av_strstart(p->filename, "rtp:", NULL))
1980 return AVPROBE_SCORE_MAX;
1984 static int rtp_read_header(AVFormatContext *s)
1986 uint8_t recvbuf[1500];
1987 char host[500], sdp[500];
1989 URLContext* in = NULL;
1991 AVCodecContext codec = { 0 };
1992 struct sockaddr_storage addr;
1994 socklen_t addrlen = sizeof(addr);
1995 RTSPState *rt = s->priv_data;
1997 if (!ff_network_init())
1998 return AVERROR(EIO);
2000 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2001 &s->interrupt_callback, NULL);
2006 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2007 if (ret == AVERROR(EAGAIN))
2012 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2016 if ((recvbuf[0] & 0xc0) != 0x80) {
2017 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2022 if (RTP_PT_IS_RTCP(recvbuf[1]))
2025 payload_type = recvbuf[1] & 0x7f;
2028 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2032 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2033 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2034 "without an SDP file describing it\n",
2038 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2039 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2040 "properly you need an SDP file "
2044 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2045 NULL, 0, s->filename);
2047 snprintf(sdp, sizeof(sdp),
2048 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2049 addr.ss_family == AF_INET ? 4 : 6, host,
2050 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2051 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2052 port, payload_type);
2053 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2055 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2058 /* sdp_read_header initializes this again */
2061 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2063 ret = sdp_read_header(s);
2074 static const AVClass rtp_demuxer_class = {
2075 .class_name = "RTP demuxer",
2076 .item_name = av_default_item_name,
2077 .option = rtp_options,
2078 .version = LIBAVUTIL_VERSION_INT,
2081 AVInputFormat ff_rtp_demuxer = {
2083 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2084 .priv_data_size = sizeof(RTSPState),
2085 .read_probe = rtp_probe,
2086 .read_header = rtp_read_header,
2087 .read_packet = ff_rtsp_fetch_packet,
2088 .read_close = sdp_read_close,
2089 .flags = AVFMT_NOFILE,
2090 .priv_class = &rtp_demuxer_class,
2092 #endif /* CONFIG_RTP_DEMUXER */