3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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"
45 #include "rtpdec_formats.h"
46 #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, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
68 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76 const AVOption ff_rtsp_options[] = {
77 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
78 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
79 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
80 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
81 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
82 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
83 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
84 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
85 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
86 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
87 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
88 { "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 },
92 static const AVOption sdp_options[] = {
93 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
94 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
98 static const AVOption rtp_options[] = {
99 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
103 static void get_word_until_chars(char *buf, int buf_size,
104 const char *sep, const char **pp)
110 p += strspn(p, SPACE_CHARS);
112 while (!strchr(sep, *p) && *p != '\0') {
113 if ((q - buf) < buf_size - 1)
122 static void get_word_sep(char *buf, int buf_size, const char *sep,
125 if (**pp == '/') (*pp)++;
126 get_word_until_chars(buf, buf_size, sep, pp);
129 static void get_word(char *buf, int buf_size, const char **pp)
131 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
134 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
136 * Used for seeking in the rtp stream.
138 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
142 p += strspn(p, SPACE_CHARS);
143 if (!av_stristart(p, "npt=", &p))
146 *start = AV_NOPTS_VALUE;
147 *end = AV_NOPTS_VALUE;
149 get_word_sep(buf, sizeof(buf), "-", &p);
150 av_parse_time(start, buf, 1);
153 get_word_sep(buf, sizeof(buf), "-", &p);
154 av_parse_time(end, buf, 1);
156 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
157 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
160 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
162 struct addrinfo hints = { 0 }, *ai = NULL;
163 hints.ai_flags = AI_NUMERICHOST;
164 if (getaddrinfo(buf, NULL, &hints, &ai))
166 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
172 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
173 RTSPStream *rtsp_st, AVCodecContext *codec)
177 codec->codec_id = handler->codec_id;
178 rtsp_st->dynamic_handler = handler;
179 if (handler->alloc) {
180 rtsp_st->dynamic_protocol_context = handler->alloc();
181 if (!rtsp_st->dynamic_protocol_context)
182 rtsp_st->dynamic_handler = NULL;
186 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
187 static int sdp_parse_rtpmap(AVFormatContext *s,
188 AVStream *st, RTSPStream *rtsp_st,
189 int payload_type, const char *p)
191 AVCodecContext *codec = st->codec;
197 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
198 * see if we can handle this kind of payload.
199 * The space should normally not be there but some Real streams or
200 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
201 * have a trailing space. */
202 get_word_sep(buf, sizeof(buf), "/ ", &p);
203 if (payload_type < RTP_PT_PRIVATE) {
204 /* We are in a standard case
205 * (from http://www.iana.org/assignments/rtp-parameters). */
206 /* search into AVRtpPayloadTypes[] */
207 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
210 if (codec->codec_id == AV_CODEC_ID_NONE) {
211 RTPDynamicProtocolHandler *handler =
212 ff_rtp_handler_find_by_name(buf, codec->codec_type);
213 init_rtp_handler(handler, rtsp_st, codec);
214 /* If no dynamic handler was found, check with the list of standard
215 * allocated types, if such a stream for some reason happens to
216 * use a private payload type. This isn't handled in rtpdec.c, since
217 * the format name from the rtpmap line never is passed into rtpdec. */
218 if (!rtsp_st->dynamic_handler)
219 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
222 c = avcodec_find_decoder(codec->codec_id);
228 get_word_sep(buf, sizeof(buf), "/", &p);
230 switch (codec->codec_type) {
231 case AVMEDIA_TYPE_AUDIO:
232 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
233 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
234 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
236 codec->sample_rate = i;
237 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
238 get_word_sep(buf, sizeof(buf), "/", &p);
242 // TODO: there is a bug here; if it is a mono stream, and
243 // less than 22000Hz, faad upconverts to stereo and twice
244 // the frequency. No problem, but the sample rate is being
245 // set here by the sdp line. Patch on its way. (rdm)
247 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
249 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
252 case AVMEDIA_TYPE_VIDEO:
253 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
255 avpriv_set_pts_info(st, 32, 1, i);
260 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
261 rtsp_st->dynamic_handler->init(s, st->index,
262 rtsp_st->dynamic_protocol_context);
266 /* parse the attribute line from the fmtp a line of an sdp response. This
267 * is broken out as a function because it is used in rtp_h264.c, which is
269 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
270 char *value, int value_size)
272 *p += strspn(*p, SPACE_CHARS);
274 get_word_sep(attr, attr_size, "=", p);
277 get_word_sep(value, value_size, ";", p);
285 typedef struct SDPParseState {
287 struct sockaddr_storage default_ip;
289 int skip_media; ///< set if an unknown m= line occurs
292 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
293 int letter, const char *buf)
295 RTSPState *rt = s->priv_data;
296 char buf1[64], st_type[64];
298 enum AVMediaType codec_type;
302 struct sockaddr_storage sdp_ip;
305 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
308 if (s1->skip_media && letter != 'm')
312 get_word(buf1, sizeof(buf1), &p);
313 if (strcmp(buf1, "IN") != 0)
315 get_word(buf1, sizeof(buf1), &p);
316 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
318 get_word_sep(buf1, sizeof(buf1), "/", &p);
319 if (get_sockaddr(buf1, &sdp_ip))
324 get_word_sep(buf1, sizeof(buf1), "/", &p);
327 if (s->nb_streams == 0) {
328 s1->default_ip = sdp_ip;
329 s1->default_ttl = ttl;
331 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
332 rtsp_st->sdp_ip = sdp_ip;
333 rtsp_st->sdp_ttl = ttl;
337 av_dict_set(&s->metadata, "title", p, 0);
340 if (s->nb_streams == 0) {
341 av_dict_set(&s->metadata, "comment", p, 0);
348 codec_type = AVMEDIA_TYPE_UNKNOWN;
349 get_word(st_type, sizeof(st_type), &p);
350 if (!strcmp(st_type, "audio")) {
351 codec_type = AVMEDIA_TYPE_AUDIO;
352 } else if (!strcmp(st_type, "video")) {
353 codec_type = AVMEDIA_TYPE_VIDEO;
354 } else if (!strcmp(st_type, "application")) {
355 codec_type = AVMEDIA_TYPE_DATA;
357 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
361 rtsp_st = av_mallocz(sizeof(RTSPStream));
364 rtsp_st->stream_index = -1;
365 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
367 rtsp_st->sdp_ip = s1->default_ip;
368 rtsp_st->sdp_ttl = s1->default_ttl;
370 get_word(buf1, sizeof(buf1), &p); /* port */
371 rtsp_st->sdp_port = atoi(buf1);
373 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
375 /* XXX: handle list of formats */
376 get_word(buf1, sizeof(buf1), &p); /* format list */
377 rtsp_st->sdp_payload_type = atoi(buf1);
379 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
380 /* no corresponding stream */
381 } else if (rt->server_type == RTSP_SERVER_WMS &&
382 codec_type == AVMEDIA_TYPE_DATA) {
383 /* RTX stream, a stream that carries all the other actual
384 * audio/video streams. Don't expose this to the callers. */
386 st = avformat_new_stream(s, NULL);
389 st->id = rt->nb_rtsp_streams - 1;
390 rtsp_st->stream_index = st->index;
391 st->codec->codec_type = codec_type;
392 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
393 RTPDynamicProtocolHandler *handler;
394 /* if standard payload type, we can find the codec right now */
395 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
396 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
397 st->codec->sample_rate > 0)
398 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
399 /* Even static payload types may need a custom depacketizer */
400 handler = ff_rtp_handler_find_by_id(
401 rtsp_st->sdp_payload_type, st->codec->codec_type);
402 init_rtp_handler(handler, rtsp_st, st->codec);
403 if (handler && handler->init)
404 handler->init(s, st->index,
405 rtsp_st->dynamic_protocol_context);
408 /* put a default control url */
409 av_strlcpy(rtsp_st->control_url, rt->control_uri,
410 sizeof(rtsp_st->control_url));
413 if (av_strstart(p, "control:", &p)) {
414 if (s->nb_streams == 0) {
415 if (!strncmp(p, "rtsp://", 7))
416 av_strlcpy(rt->control_uri, p,
417 sizeof(rt->control_uri));
420 /* get the control url */
421 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
423 /* XXX: may need to add full url resolution */
424 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
426 if (proto[0] == '\0') {
427 /* relative control URL */
428 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
429 av_strlcat(rtsp_st->control_url, "/",
430 sizeof(rtsp_st->control_url));
431 av_strlcat(rtsp_st->control_url, p,
432 sizeof(rtsp_st->control_url));
434 av_strlcpy(rtsp_st->control_url, p,
435 sizeof(rtsp_st->control_url));
437 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
438 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
439 get_word(buf1, sizeof(buf1), &p);
440 payload_type = atoi(buf1);
441 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
442 if (rtsp_st->stream_index >= 0) {
443 st = s->streams[rtsp_st->stream_index];
444 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
446 } else if (av_strstart(p, "fmtp:", &p) ||
447 av_strstart(p, "framesize:", &p)) {
448 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
449 // let dynamic protocol handlers have a stab at the line.
450 get_word(buf1, sizeof(buf1), &p);
451 payload_type = atoi(buf1);
452 for (i = 0; i < rt->nb_rtsp_streams; i++) {
453 rtsp_st = rt->rtsp_streams[i];
454 if (rtsp_st->sdp_payload_type == payload_type &&
455 rtsp_st->dynamic_handler &&
456 rtsp_st->dynamic_handler->parse_sdp_a_line)
457 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
458 rtsp_st->dynamic_protocol_context, buf);
460 } else if (av_strstart(p, "range:", &p)) {
463 // this is so that seeking on a streamed file can work.
464 rtsp_parse_range_npt(p, &start, &end);
465 s->start_time = start;
466 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
467 s->duration = (end == AV_NOPTS_VALUE) ?
468 AV_NOPTS_VALUE : end - start;
469 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
471 rt->transport = RTSP_TRANSPORT_RDT;
472 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
474 st = s->streams[s->nb_streams - 1];
475 st->codec->sample_rate = atoi(p);
477 if (rt->server_type == RTSP_SERVER_WMS)
478 ff_wms_parse_sdp_a_line(s, p);
479 if (s->nb_streams > 0) {
480 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
482 if (rt->server_type == RTSP_SERVER_REAL)
483 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
485 if (rtsp_st->dynamic_handler &&
486 rtsp_st->dynamic_handler->parse_sdp_a_line)
487 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
488 rtsp_st->stream_index,
489 rtsp_st->dynamic_protocol_context, buf);
496 int ff_sdp_parse(AVFormatContext *s, const char *content)
498 RTSPState *rt = s->priv_data;
501 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
502 * contain long SDP lines containing complete ASF Headers (several
503 * kB) or arrays of MDPR (RM stream descriptor) headers plus
504 * "rulebooks" describing their properties. Therefore, the SDP line
507 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
508 * in rtpdec_xiph.c. */
510 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
514 p += strspn(p, SPACE_CHARS);
522 /* get the content */
524 while (*p != '\n' && *p != '\r' && *p != '\0') {
525 if ((q - buf) < sizeof(buf) - 1)
530 sdp_parse_line(s, s1, letter, buf);
532 while (*p != '\n' && *p != '\0')
537 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
538 if (!rt->p) return AVERROR(ENOMEM);
541 #endif /* CONFIG_RTPDEC */
543 void ff_rtsp_undo_setup(AVFormatContext *s)
545 RTSPState *rt = s->priv_data;
548 for (i = 0; i < rt->nb_rtsp_streams; i++) {
549 RTSPStream *rtsp_st = rt->rtsp_streams[i];
552 if (rtsp_st->transport_priv) {
554 AVFormatContext *rtpctx = rtsp_st->transport_priv;
555 av_write_trailer(rtpctx);
556 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
558 avio_close_dyn_buf(rtpctx->pb, &ptr);
561 avio_close(rtpctx->pb);
563 avformat_free_context(rtpctx);
564 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
565 ff_rdt_parse_close(rtsp_st->transport_priv);
566 else if (CONFIG_RTPDEC)
567 ff_rtp_parse_close(rtsp_st->transport_priv);
569 rtsp_st->transport_priv = NULL;
570 if (rtsp_st->rtp_handle)
571 ffurl_close(rtsp_st->rtp_handle);
572 rtsp_st->rtp_handle = NULL;
576 /* close and free RTSP streams */
577 void ff_rtsp_close_streams(AVFormatContext *s)
579 RTSPState *rt = s->priv_data;
583 ff_rtsp_undo_setup(s);
584 for (i = 0; i < rt->nb_rtsp_streams; i++) {
585 rtsp_st = rt->rtsp_streams[i];
587 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
588 rtsp_st->dynamic_handler->free(
589 rtsp_st->dynamic_protocol_context);
593 av_free(rt->rtsp_streams);
595 avformat_close_input(&rt->asf_ctx);
598 av_free(rt->recvbuf);
601 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
603 RTSPState *rt = s->priv_data;
606 /* open the RTP context */
607 if (rtsp_st->stream_index >= 0)
608 st = s->streams[rtsp_st->stream_index];
610 s->ctx_flags |= AVFMTCTX_NOHEADER;
612 if (s->oformat && CONFIG_RTSP_MUXER) {
613 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
615 RTSP_TCP_MAX_PACKET_SIZE);
616 /* Ownership of rtp_handle is passed to the rtp mux context */
617 rtsp_st->rtp_handle = NULL;
620 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
621 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
622 rtsp_st->dynamic_protocol_context,
623 rtsp_st->dynamic_handler);
624 else if (CONFIG_RTPDEC)
625 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
626 rtsp_st->sdp_payload_type,
627 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
628 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
630 if (!rtsp_st->transport_priv) {
631 return AVERROR(ENOMEM);
632 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
633 if (rtsp_st->dynamic_handler) {
634 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
635 rtsp_st->dynamic_protocol_context,
636 rtsp_st->dynamic_handler);
643 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
644 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
651 q += strspn(q, SPACE_CHARS);
652 v = strtol(q, &p, 10);
656 v = strtol(p, &p, 10);
665 /* XXX: only one transport specification is parsed */
666 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
668 char transport_protocol[16];
670 char lower_transport[16];
672 RTSPTransportField *th;
675 reply->nb_transports = 0;
678 p += strspn(p, SPACE_CHARS);
682 th = &reply->transports[reply->nb_transports];
684 get_word_sep(transport_protocol, sizeof(transport_protocol),
686 if (!av_strcasecmp (transport_protocol, "rtp")) {
687 get_word_sep(profile, sizeof(profile), "/;,", &p);
688 lower_transport[0] = '\0';
689 /* rtp/avp/<protocol> */
691 get_word_sep(lower_transport, sizeof(lower_transport),
694 th->transport = RTSP_TRANSPORT_RTP;
695 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
696 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
697 /* x-pn-tng/<protocol> */
698 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
700 th->transport = RTSP_TRANSPORT_RDT;
702 if (!av_strcasecmp(lower_transport, "TCP"))
703 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
705 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
709 /* get each parameter */
710 while (*p != '\0' && *p != ',') {
711 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
712 if (!strcmp(parameter, "port")) {
715 rtsp_parse_range(&th->port_min, &th->port_max, &p);
717 } else if (!strcmp(parameter, "client_port")) {
720 rtsp_parse_range(&th->client_port_min,
721 &th->client_port_max, &p);
723 } else if (!strcmp(parameter, "server_port")) {
726 rtsp_parse_range(&th->server_port_min,
727 &th->server_port_max, &p);
729 } else if (!strcmp(parameter, "interleaved")) {
732 rtsp_parse_range(&th->interleaved_min,
733 &th->interleaved_max, &p);
735 } else if (!strcmp(parameter, "multicast")) {
736 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
737 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
738 } else if (!strcmp(parameter, "ttl")) {
741 th->ttl = strtol(p, (char **)&p, 10);
743 } else if (!strcmp(parameter, "destination")) {
746 get_word_sep(buf, sizeof(buf), ";,", &p);
747 get_sockaddr(buf, &th->destination);
749 } else if (!strcmp(parameter, "source")) {
752 get_word_sep(buf, sizeof(buf), ";,", &p);
753 av_strlcpy(th->source, buf, sizeof(th->source));
755 } else if (!strcmp(parameter, "mode")) {
758 get_word_sep(buf, sizeof(buf), ";, ", &p);
759 if (!strcmp(buf, "record") ||
760 !strcmp(buf, "receive"))
765 while (*p != ';' && *p != '\0' && *p != ',')
773 reply->nb_transports++;
777 static void handle_rtp_info(RTSPState *rt, const char *url,
778 uint32_t seq, uint32_t rtptime)
781 if (!rtptime || !url[0])
783 if (rt->transport != RTSP_TRANSPORT_RTP)
785 for (i = 0; i < rt->nb_rtsp_streams; i++) {
786 RTSPStream *rtsp_st = rt->rtsp_streams[i];
787 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
790 if (!strcmp(rtsp_st->control_url, url)) {
791 rtpctx->base_timestamp = rtptime;
797 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
800 char key[20], value[1024], url[1024] = "";
801 uint32_t seq = 0, rtptime = 0;
804 p += strspn(p, SPACE_CHARS);
807 get_word_sep(key, sizeof(key), "=", &p);
811 get_word_sep(value, sizeof(value), ";, ", &p);
813 if (!strcmp(key, "url"))
814 av_strlcpy(url, value, sizeof(url));
815 else if (!strcmp(key, "seq"))
816 seq = strtoul(value, NULL, 10);
817 else if (!strcmp(key, "rtptime"))
818 rtptime = strtoul(value, NULL, 10);
820 handle_rtp_info(rt, url, seq, rtptime);
829 handle_rtp_info(rt, url, seq, rtptime);
832 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
833 RTSPState *rt, const char *method)
837 /* NOTE: we do case independent match for broken servers */
839 if (av_stristart(p, "Session:", &p)) {
841 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
842 if (av_stristart(p, ";timeout=", &p) &&
843 (t = strtol(p, NULL, 10)) > 0) {
846 } else if (av_stristart(p, "Content-Length:", &p)) {
847 reply->content_length = strtol(p, NULL, 10);
848 } else if (av_stristart(p, "Transport:", &p)) {
849 rtsp_parse_transport(reply, p);
850 } else if (av_stristart(p, "CSeq:", &p)) {
851 reply->seq = strtol(p, NULL, 10);
852 } else if (av_stristart(p, "Range:", &p)) {
853 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
854 } else if (av_stristart(p, "RealChallenge1:", &p)) {
855 p += strspn(p, SPACE_CHARS);
856 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
857 } else if (av_stristart(p, "Server:", &p)) {
858 p += strspn(p, SPACE_CHARS);
859 av_strlcpy(reply->server, p, sizeof(reply->server));
860 } else if (av_stristart(p, "Notice:", &p) ||
861 av_stristart(p, "X-Notice:", &p)) {
862 reply->notice = strtol(p, NULL, 10);
863 } else if (av_stristart(p, "Location:", &p)) {
864 p += strspn(p, SPACE_CHARS);
865 av_strlcpy(reply->location, p , sizeof(reply->location));
866 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
867 p += strspn(p, SPACE_CHARS);
868 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
869 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
870 p += strspn(p, SPACE_CHARS);
871 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
872 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
873 p += strspn(p, SPACE_CHARS);
874 if (method && !strcmp(method, "DESCRIBE"))
875 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
876 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
877 p += strspn(p, SPACE_CHARS);
878 if (method && !strcmp(method, "PLAY"))
879 rtsp_parse_rtp_info(rt, p);
880 } else if (av_stristart(p, "Public:", &p) && rt) {
881 if (strstr(p, "GET_PARAMETER") &&
882 method && !strcmp(method, "OPTIONS"))
883 rt->get_parameter_supported = 1;
884 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
885 p += strspn(p, SPACE_CHARS);
886 rt->accept_dynamic_rate = atoi(p);
887 } else if (av_stristart(p, "Content-Type:", &p)) {
888 p += strspn(p, SPACE_CHARS);
889 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
893 /* skip a RTP/TCP interleaved packet */
894 void ff_rtsp_skip_packet(AVFormatContext *s)
896 RTSPState *rt = s->priv_data;
900 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
903 len = AV_RB16(buf + 1);
905 av_dlog(s, "skipping RTP packet len=%d\n", len);
910 if (len1 > sizeof(buf))
912 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
919 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
920 unsigned char **content_ptr,
921 int return_on_interleaved_data, const char *method)
923 RTSPState *rt = s->priv_data;
924 char buf[4096], buf1[1024], *q;
927 int ret, content_length, line_count = 0, request = 0;
928 unsigned char *content = NULL;
934 memset(reply, 0, sizeof(*reply));
936 /* parse reply (XXX: use buffers) */
937 rt->last_reply[0] = '\0';
941 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
942 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
948 /* XXX: only parse it if first char on line ? */
949 if (return_on_interleaved_data) {
952 ff_rtsp_skip_packet(s);
953 } else if (ch != '\r') {
954 if ((q - buf) < sizeof(buf) - 1)
960 av_dlog(s, "line='%s'\n", buf);
962 /* test if last line */
966 if (line_count == 0) {
968 get_word(buf1, sizeof(buf1), &p);
969 if (!strncmp(buf1, "RTSP/", 5)) {
970 get_word(buf1, sizeof(buf1), &p);
971 reply->status_code = atoi(buf1);
972 av_strlcpy(reply->reason, p, sizeof(reply->reason));
974 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
975 get_word(buf1, sizeof(buf1), &p); // object
979 ff_rtsp_parse_line(reply, p, rt, method);
980 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
981 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
986 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
987 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
989 content_length = reply->content_length;
990 if (content_length > 0) {
991 /* leave some room for a trailing '\0' (useful for simple parsing) */
992 content = av_malloc(content_length + 1);
993 ffurl_read_complete(rt->rtsp_hd, content, content_length);
994 content[content_length] = '\0';
997 *content_ptr = content;
1003 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1004 const char* ptr = buf;
1006 if (!strcmp(reply->reason, "OPTIONS")) {
1007 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1009 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1010 if (reply->session_id[0])
1011 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1014 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1016 av_strlcat(buf, "\r\n", sizeof(buf));
1018 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1019 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1022 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1024 rt->last_cmd_time = av_gettime();
1025 /* Even if the request from the server had data, it is not the data
1026 * that the caller wants or expects. The memory could also be leaked
1027 * if the actual following reply has content data. */
1029 av_freep(content_ptr);
1030 /* If method is set, this is called from ff_rtsp_send_cmd,
1031 * where a reply to exactly this request is awaited. For
1032 * callers from within packet receiving, we just want to
1033 * return to the caller and go back to receiving packets. */
1039 if (rt->seq != reply->seq) {
1040 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1041 rt->seq, reply->seq);
1045 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1046 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1047 reply->notice == 2306 /* Continuous Feed Terminated */) {
1048 rt->state = RTSP_STATE_IDLE;
1049 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1050 return AVERROR(EIO); /* data or server error */
1051 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1052 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1053 return AVERROR(EPERM);
1059 * Send a command to the RTSP server without waiting for the reply.
1061 * @param s RTSP (de)muxer context
1062 * @param method the method for the request
1063 * @param url the target url for the request
1064 * @param headers extra header lines to include in the request
1065 * @param send_content if non-null, the data to send as request body content
1066 * @param send_content_length the length of the send_content data, or 0 if
1067 * send_content is null
1069 * @return zero if success, nonzero otherwise
1071 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1072 const char *method, const char *url,
1073 const char *headers,
1074 const unsigned char *send_content,
1075 int send_content_length)
1077 RTSPState *rt = s->priv_data;
1078 char buf[4096], *out_buf;
1079 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1081 /* Add in RTSP headers */
1084 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1086 av_strlcat(buf, headers, sizeof(buf));
1087 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1088 if (rt->session_id[0] != '\0' && (!headers ||
1089 !strstr(headers, "\nIf-Match:"))) {
1090 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1093 char *str = ff_http_auth_create_response(&rt->auth_state,
1094 rt->auth, url, method);
1096 av_strlcat(buf, str, sizeof(buf));
1099 if (send_content_length > 0 && send_content)
1100 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1101 av_strlcat(buf, "\r\n", sizeof(buf));
1103 /* base64 encode rtsp if tunneling */
1104 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1105 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1106 out_buf = base64buf;
1109 av_dlog(s, "Sending:\n%s--\n", buf);
1111 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1112 if (send_content_length > 0 && send_content) {
1113 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1114 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1115 "with content data not supported\n");
1116 return AVERROR_PATCHWELCOME;
1118 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1120 rt->last_cmd_time = av_gettime();
1125 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1126 const char *url, const char *headers)
1128 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1131 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1132 const char *headers, RTSPMessageHeader *reply,
1133 unsigned char **content_ptr)
1135 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1136 content_ptr, NULL, 0);
1139 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1140 const char *method, const char *url,
1142 RTSPMessageHeader *reply,
1143 unsigned char **content_ptr,
1144 const unsigned char *send_content,
1145 int send_content_length)
1147 RTSPState *rt = s->priv_data;
1148 HTTPAuthType cur_auth_type;
1149 int ret, attempts = 0;
1152 cur_auth_type = rt->auth_state.auth_type;
1153 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1155 send_content_length)))
1158 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1162 if (reply->status_code == 401 &&
1163 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1164 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1167 if (reply->status_code > 400){
1168 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1172 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1178 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1179 int lower_transport, const char *real_challenge)
1181 RTSPState *rt = s->priv_data;
1182 int rtx = 0, j, i, err, interleave = 0, port_off;
1183 RTSPStream *rtsp_st;
1184 RTSPMessageHeader reply1, *reply = &reply1;
1186 const char *trans_pref;
1188 if (rt->transport == RTSP_TRANSPORT_RDT)
1189 trans_pref = "x-pn-tng";
1191 trans_pref = "RTP/AVP";
1193 /* default timeout: 1 minute */
1196 /* Choose a random starting offset within the first half of the
1197 * port range, to allow for a number of ports to try even if the offset
1198 * happens to be at the end of the random range. */
1199 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1200 /* even random offset */
1201 port_off -= port_off & 0x01;
1203 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1204 char transport[2048];
1207 * WMS serves all UDP data over a single connection, the RTX, which
1208 * isn't necessarily the first in the SDP but has to be the first
1209 * to be set up, else the second/third SETUP will fail with a 461.
1211 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1212 rt->server_type == RTSP_SERVER_WMS) {
1215 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1216 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1218 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1222 if (rtx == rt->nb_rtsp_streams)
1223 return -1; /* no RTX found */
1224 rtsp_st = rt->rtsp_streams[rtx];
1226 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1228 rtsp_st = rt->rtsp_streams[i];
1231 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1234 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1235 port = reply->transports[0].client_port_min;
1239 /* first try in specified port range */
1240 while (j <= rt->rtp_port_max) {
1241 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1242 "?localport=%d", j);
1243 /* we will use two ports per rtp stream (rtp and rtcp) */
1245 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1246 &s->interrupt_callback, NULL))
1249 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1254 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1256 snprintf(transport, sizeof(transport) - 1,
1257 "%s/UDP;", trans_pref);
1258 if (rt->server_type != RTSP_SERVER_REAL)
1259 av_strlcat(transport, "unicast;", sizeof(transport));
1260 av_strlcatf(transport, sizeof(transport),
1261 "client_port=%d", port);
1262 if (rt->transport == RTSP_TRANSPORT_RTP &&
1263 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1264 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1268 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1269 /* For WMS streams, the application streams are only used for
1270 * UDP. When trying to set it up for TCP streams, the server
1271 * will return an error. Therefore, we skip those streams. */
1272 if (rt->server_type == RTSP_SERVER_WMS &&
1273 (rtsp_st->stream_index < 0 ||
1274 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1277 snprintf(transport, sizeof(transport) - 1,
1278 "%s/TCP;", trans_pref);
1279 if (rt->transport != RTSP_TRANSPORT_RDT)
1280 av_strlcat(transport, "unicast;", sizeof(transport));
1281 av_strlcatf(transport, sizeof(transport),
1282 "interleaved=%d-%d",
1283 interleave, interleave + 1);
1287 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1288 snprintf(transport, sizeof(transport) - 1,
1289 "%s/UDP;multicast", trans_pref);
1292 av_strlcat(transport, ";mode=record", sizeof(transport));
1293 } else if (rt->server_type == RTSP_SERVER_REAL ||
1294 rt->server_type == RTSP_SERVER_WMS)
1295 av_strlcat(transport, ";mode=play", sizeof(transport));
1296 snprintf(cmd, sizeof(cmd),
1297 "Transport: %s\r\n",
1299 if (rt->accept_dynamic_rate)
1300 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1301 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1302 char real_res[41], real_csum[9];
1303 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1305 av_strlcatf(cmd, sizeof(cmd),
1307 "RealChallenge2: %s, sd=%s\r\n",
1308 rt->session_id, real_res, real_csum);
1310 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1311 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1314 } else if (reply->status_code != RTSP_STATUS_OK ||
1315 reply->nb_transports != 1) {
1316 err = AVERROR_INVALIDDATA;
1320 /* XXX: same protocol for all streams is required */
1322 if (reply->transports[0].lower_transport != rt->lower_transport ||
1323 reply->transports[0].transport != rt->transport) {
1324 err = AVERROR_INVALIDDATA;
1328 rt->lower_transport = reply->transports[0].lower_transport;
1329 rt->transport = reply->transports[0].transport;
1332 /* Fail if the server responded with another lower transport mode
1333 * than what we requested. */
1334 if (reply->transports[0].lower_transport != lower_transport) {
1335 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1336 err = AVERROR_INVALIDDATA;
1340 switch(reply->transports[0].lower_transport) {
1341 case RTSP_LOWER_TRANSPORT_TCP:
1342 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1343 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1346 case RTSP_LOWER_TRANSPORT_UDP: {
1347 char url[1024], options[30] = "";
1349 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1350 av_strlcpy(options, "?connect=1", sizeof(options));
1351 /* Use source address if specified */
1352 if (reply->transports[0].source[0]) {
1353 ff_url_join(url, sizeof(url), "rtp", NULL,
1354 reply->transports[0].source,
1355 reply->transports[0].server_port_min, "%s", options);
1357 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1358 reply->transports[0].server_port_min, "%s", options);
1360 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1361 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1362 err = AVERROR_INVALIDDATA;
1365 /* Try to initialize the connection state in a
1366 * potential NAT router by sending dummy packets.
1367 * RTP/RTCP dummy packets are used for RDT, too.
1369 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1371 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1374 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1375 char url[1024], namebuf[50], optbuf[20] = "";
1376 struct sockaddr_storage addr;
1379 if (reply->transports[0].destination.ss_family) {
1380 addr = reply->transports[0].destination;
1381 port = reply->transports[0].port_min;
1382 ttl = reply->transports[0].ttl;
1384 addr = rtsp_st->sdp_ip;
1385 port = rtsp_st->sdp_port;
1386 ttl = rtsp_st->sdp_ttl;
1389 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1390 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1391 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1392 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1393 port, "%s", optbuf);
1394 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1395 &s->interrupt_callback, NULL) < 0) {
1396 err = AVERROR_INVALIDDATA;
1403 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1407 if (rt->nb_rtsp_streams && reply->timeout > 0)
1408 rt->timeout = reply->timeout;
1410 if (rt->server_type == RTSP_SERVER_REAL)
1411 rt->need_subscription = 1;
1416 ff_rtsp_undo_setup(s);
1420 void ff_rtsp_close_connections(AVFormatContext *s)
1422 RTSPState *rt = s->priv_data;
1423 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1424 ffurl_close(rt->rtsp_hd);
1425 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1428 int ff_rtsp_connect(AVFormatContext *s)
1430 RTSPState *rt = s->priv_data;
1431 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1432 int port, err, tcp_fd;
1433 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1434 int lower_transport_mask = 0;
1435 char real_challenge[64] = "";
1436 struct sockaddr_storage peer;
1437 socklen_t peer_len = sizeof(peer);
1439 if (rt->rtp_port_max < rt->rtp_port_min) {
1440 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1441 "than min port %d\n", rt->rtp_port_max,
1443 return AVERROR(EINVAL);
1446 if (!ff_network_init())
1447 return AVERROR(EIO);
1449 if (s->max_delay < 0) /* Not set by the caller */
1450 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1452 rt->control_transport = RTSP_MODE_PLAIN;
1453 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1454 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1455 rt->control_transport = RTSP_MODE_TUNNEL;
1457 /* Only pass through valid flags from here */
1458 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1461 lower_transport_mask = rt->lower_transport_mask;
1462 /* extract hostname and port */
1463 av_url_split(NULL, 0, auth, sizeof(auth),
1464 host, sizeof(host), &port, path, sizeof(path), s->filename);
1466 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1469 port = RTSP_DEFAULT_PORT;
1471 if (!lower_transport_mask)
1472 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1475 /* Only UDP or TCP - UDP multicast isn't supported. */
1476 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1477 (1 << RTSP_LOWER_TRANSPORT_TCP);
1478 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1479 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1480 "only UDP and TCP are supported for output.\n");
1481 err = AVERROR(EINVAL);
1486 /* Construct the URI used in request; this is similar to s->filename,
1487 * but with authentication credentials removed and RTSP specific options
1489 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1490 host, port, "%s", path);
1492 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1493 /* set up initial handshake for tunneling */
1494 char httpname[1024];
1495 char sessioncookie[17];
1498 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1499 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1500 av_get_random_seed(), av_get_random_seed());
1503 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1504 &s->interrupt_callback) < 0) {
1509 /* generate GET headers */
1510 snprintf(headers, sizeof(headers),
1511 "x-sessioncookie: %s\r\n"
1512 "Accept: application/x-rtsp-tunnelled\r\n"
1513 "Pragma: no-cache\r\n"
1514 "Cache-Control: no-cache\r\n",
1516 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1518 /* complete the connection */
1519 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1525 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1526 &s->interrupt_callback) < 0 ) {
1531 /* generate POST headers */
1532 snprintf(headers, sizeof(headers),
1533 "x-sessioncookie: %s\r\n"
1534 "Content-Type: application/x-rtsp-tunnelled\r\n"
1535 "Pragma: no-cache\r\n"
1536 "Cache-Control: no-cache\r\n"
1537 "Content-Length: 32767\r\n"
1538 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1540 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1541 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1543 /* Initialize the authentication state for the POST session. The HTTP
1544 * protocol implementation doesn't properly handle multi-pass
1545 * authentication for POST requests, since it would require one of
1547 * - implementing Expect: 100-continue, which many HTTP servers
1548 * don't support anyway, even less the RTSP servers that do HTTP
1550 * - sending the whole POST data until getting a 401 reply specifying
1551 * what authentication method to use, then resending all that data
1552 * - waiting for potential 401 replies directly after sending the
1553 * POST header (waiting for some unspecified time)
1554 * Therefore, we copy the full auth state, which works for both basic
1555 * and digest. (For digest, we would have to synchronize the nonce
1556 * count variable between the two sessions, if we'd do more requests
1557 * with the original session, though.)
1559 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1561 /* complete the connection */
1562 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1567 /* open the tcp connection */
1568 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1569 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1570 &s->interrupt_callback, NULL) < 0) {
1574 rt->rtsp_hd_out = rt->rtsp_hd;
1578 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1579 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1580 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1581 NULL, 0, NI_NUMERICHOST);
1584 /* request options supported by the server; this also detects server
1586 for (rt->server_type = RTSP_SERVER_RTP;;) {
1588 if (rt->server_type == RTSP_SERVER_REAL)
1591 * The following entries are required for proper
1592 * streaming from a Realmedia server. They are
1593 * interdependent in some way although we currently
1594 * don't quite understand how. Values were copied
1595 * from mplayer SVN r23589.
1596 * ClientChallenge is a 16-byte ID in hex
1597 * CompanyID is a 16-byte ID in base64
1599 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1600 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1601 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1602 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1604 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1605 if (reply->status_code != RTSP_STATUS_OK) {
1606 err = AVERROR_INVALIDDATA;
1610 /* detect server type if not standard-compliant RTP */
1611 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1612 rt->server_type = RTSP_SERVER_REAL;
1614 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1615 rt->server_type = RTSP_SERVER_WMS;
1616 } else if (rt->server_type == RTSP_SERVER_REAL)
1617 strcpy(real_challenge, reply->real_challenge);
1621 if (s->iformat && CONFIG_RTSP_DEMUXER)
1622 err = ff_rtsp_setup_input_streams(s, reply);
1623 else if (CONFIG_RTSP_MUXER)
1624 err = ff_rtsp_setup_output_streams(s, host);
1629 int lower_transport = ff_log2_tab[lower_transport_mask &
1630 ~(lower_transport_mask - 1)];
1632 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1633 rt->server_type == RTSP_SERVER_REAL ?
1634 real_challenge : NULL);
1637 lower_transport_mask &= ~(1 << lower_transport);
1638 if (lower_transport_mask == 0 && err == 1) {
1639 err = AVERROR(EPROTONOSUPPORT);
1644 rt->lower_transport_mask = lower_transport_mask;
1645 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1646 rt->state = RTSP_STATE_IDLE;
1647 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1650 ff_rtsp_close_streams(s);
1651 ff_rtsp_close_connections(s);
1652 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1653 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1654 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1662 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1665 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1666 uint8_t *buf, int buf_size, int64_t wait_end)
1668 RTSPState *rt = s->priv_data;
1669 RTSPStream *rtsp_st;
1670 int n, i, ret, tcp_fd, timeout_cnt = 0;
1672 struct pollfd *p = rt->p;
1675 if (ff_check_interrupt(&s->interrupt_callback))
1676 return AVERROR_EXIT;
1677 if (wait_end && wait_end - av_gettime() < 0)
1678 return AVERROR(EAGAIN);
1681 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1682 p[max_p].fd = tcp_fd;
1683 p[max_p++].events = POLLIN;
1687 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1688 rtsp_st = rt->rtsp_streams[i];
1689 if (rtsp_st->rtp_handle) {
1690 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1691 p[max_p++].events = POLLIN;
1692 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1693 p[max_p++].events = POLLIN;
1696 n = poll(p, max_p, POLL_TIMEOUT_MS);
1698 int j = 1 - (tcp_fd == -1);
1700 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1701 rtsp_st = rt->rtsp_streams[i];
1702 if (rtsp_st->rtp_handle) {
1703 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1704 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1706 *prtsp_st = rtsp_st;
1713 #if CONFIG_RTSP_DEMUXER
1714 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1715 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1716 if (rt->state == RTSP_STATE_STREAMING) {
1717 if (!ff_rtsp_parse_streaming_commands(s))
1720 av_log(s, AV_LOG_WARNING,
1721 "Unable to answer to TEARDOWN\n");
1725 RTSPMessageHeader reply;
1726 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1729 /* XXX: parse message */
1730 if (rt->state != RTSP_STATE_STREAMING)
1735 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1736 return AVERROR(ETIMEDOUT);
1737 } else if (n < 0 && errno != EINTR)
1738 return AVERROR(errno);
1742 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1744 RTSPState *rt = s->priv_data;
1746 RTSPStream *rtsp_st, *first_queue_st = NULL;
1747 int64_t wait_end = 0;
1749 if (rt->nb_byes == rt->nb_rtsp_streams)
1752 /* get next frames from the same RTP packet */
1753 if (rt->cur_transport_priv) {
1754 if (rt->transport == RTSP_TRANSPORT_RDT) {
1755 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1757 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1759 rt->cur_transport_priv = NULL;
1761 } else if (ret == 1) {
1764 rt->cur_transport_priv = NULL;
1767 if (rt->transport == RTSP_TRANSPORT_RTP) {
1769 int64_t first_queue_time = 0;
1770 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1771 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1775 queue_time = ff_rtp_queued_packet_time(rtpctx);
1776 if (queue_time && (queue_time - first_queue_time < 0 ||
1777 !first_queue_time)) {
1778 first_queue_time = queue_time;
1779 first_queue_st = rt->rtsp_streams[i];
1782 if (first_queue_time)
1783 wait_end = first_queue_time + s->max_delay;
1786 /* read next RTP packet */
1789 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1791 return AVERROR(ENOMEM);
1794 switch(rt->lower_transport) {
1796 #if CONFIG_RTSP_DEMUXER
1797 case RTSP_LOWER_TRANSPORT_TCP:
1798 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1801 case RTSP_LOWER_TRANSPORT_UDP:
1802 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1803 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1804 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1805 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1808 if (len == AVERROR(EAGAIN) && first_queue_st &&
1809 rt->transport == RTSP_TRANSPORT_RTP) {
1810 rtsp_st = first_queue_st;
1811 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1818 if (rt->transport == RTSP_TRANSPORT_RDT) {
1819 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1821 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1823 /* Either bad packet, or a RTCP packet. Check if the
1824 * first_rtcp_ntp_time field was initialized. */
1825 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1826 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1827 /* first_rtcp_ntp_time has been initialized for this stream,
1828 * copy the same value to all other uninitialized streams,
1829 * in order to map their timestamp origin to the same ntp time
1832 AVStream *st = NULL;
1833 if (rtsp_st->stream_index >= 0)
1834 st = s->streams[rtsp_st->stream_index];
1835 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1836 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1837 AVStream *st2 = NULL;
1838 if (rt->rtsp_streams[i]->stream_index >= 0)
1839 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1840 if (rtpctx2 && st && st2 &&
1841 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1842 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1843 rtpctx2->rtcp_ts_offset = av_rescale_q(
1844 rtpctx->rtcp_ts_offset, st->time_base,
1849 if (ret == -RTCP_BYE) {
1852 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1853 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1855 if (rt->nb_byes == rt->nb_rtsp_streams)
1864 /* more packets may follow, so we save the RTP context */
1865 rt->cur_transport_priv = rtsp_st->transport_priv;
1869 #endif /* CONFIG_RTPDEC */
1871 #if CONFIG_SDP_DEMUXER
1872 static int sdp_probe(AVProbeData *p1)
1874 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1876 /* we look for a line beginning "c=IN IP" */
1877 while (p < p_end && *p != '\0') {
1878 if (p + sizeof("c=IN IP") - 1 < p_end &&
1879 av_strstart(p, "c=IN IP", NULL))
1880 return AVPROBE_SCORE_MAX / 2;
1882 while (p < p_end - 1 && *p != '\n') p++;
1891 static int sdp_read_header(AVFormatContext *s)
1893 RTSPState *rt = s->priv_data;
1894 RTSPStream *rtsp_st;
1899 if (!ff_network_init())
1900 return AVERROR(EIO);
1902 if (s->max_delay < 0) /* Not set by the caller */
1903 s->max_delay = DEFAULT_REORDERING_DELAY;
1905 /* read the whole sdp file */
1906 /* XXX: better loading */
1907 content = av_malloc(SDP_MAX_SIZE);
1908 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1911 return AVERROR_INVALIDDATA;
1913 content[size] ='\0';
1915 err = ff_sdp_parse(s, content);
1919 /* open each RTP stream */
1920 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1922 rtsp_st = rt->rtsp_streams[i];
1924 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1925 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1926 ff_url_join(url, sizeof(url), "rtp", NULL,
1927 namebuf, rtsp_st->sdp_port,
1928 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1930 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1931 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1932 &s->interrupt_callback, NULL) < 0) {
1933 err = AVERROR_INVALIDDATA;
1936 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1941 ff_rtsp_close_streams(s);
1946 static int sdp_read_close(AVFormatContext *s)
1948 ff_rtsp_close_streams(s);
1953 static const AVClass sdp_demuxer_class = {
1954 .class_name = "SDP demuxer",
1955 .item_name = av_default_item_name,
1956 .option = sdp_options,
1957 .version = LIBAVUTIL_VERSION_INT,
1960 AVInputFormat ff_sdp_demuxer = {
1962 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1963 .priv_data_size = sizeof(RTSPState),
1964 .read_probe = sdp_probe,
1965 .read_header = sdp_read_header,
1966 .read_packet = ff_rtsp_fetch_packet,
1967 .read_close = sdp_read_close,
1968 .priv_class = &sdp_demuxer_class,
1970 #endif /* CONFIG_SDP_DEMUXER */
1972 #if CONFIG_RTP_DEMUXER
1973 static int rtp_probe(AVProbeData *p)
1975 if (av_strstart(p->filename, "rtp:", NULL))
1976 return AVPROBE_SCORE_MAX;
1980 static int rtp_read_header(AVFormatContext *s)
1982 uint8_t recvbuf[1500];
1983 char host[500], sdp[500];
1985 URLContext* in = NULL;
1987 AVCodecContext codec = { 0 };
1988 struct sockaddr_storage addr;
1990 socklen_t addrlen = sizeof(addr);
1991 RTSPState *rt = s->priv_data;
1993 if (!ff_network_init())
1994 return AVERROR(EIO);
1996 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1997 &s->interrupt_callback, NULL);
2002 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2003 if (ret == AVERROR(EAGAIN))
2008 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2012 if ((recvbuf[0] & 0xc0) != 0x80) {
2013 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2018 if (RTP_PT_IS_RTCP(recvbuf[1]))
2021 payload_type = recvbuf[1] & 0x7f;
2024 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2028 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2029 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2030 "without an SDP file describing it\n",
2034 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2035 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2036 "properly you need an SDP file "
2040 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2041 NULL, 0, s->filename);
2043 snprintf(sdp, sizeof(sdp),
2044 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2045 addr.ss_family == AF_INET ? 4 : 6, host,
2046 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2047 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2048 port, payload_type);
2049 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2051 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2054 /* sdp_read_header initializes this again */
2057 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2059 ret = sdp_read_header(s);
2070 static const AVClass rtp_demuxer_class = {
2071 .class_name = "RTP demuxer",
2072 .item_name = av_default_item_name,
2073 .option = rtp_options,
2074 .version = LIBAVUTIL_VERSION_INT,
2077 AVInputFormat ff_rtp_demuxer = {
2079 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2080 .priv_data_size = sizeof(RTSPState),
2081 .read_probe = rtp_probe,
2082 .read_header = rtp_read_header,
2083 .read_packet = ff_rtsp_fetch_packet,
2084 .read_close = sdp_read_close,
2085 .flags = AVFMT_NOFILE,
2086 .priv_class = &rtp_demuxer_class,
2088 #endif /* CONFIG_RTP_DEMUXER */