3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define RTSP_FLAG_OPTS(name, longname) \
66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 #define RTSP_REORDERING_OPTS() \
76 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
78 const AVOption ff_rtsp_options[] = {
79 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
80 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
81 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
82 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
85 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
86 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
87 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
88 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
89 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
90 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
91 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
92 RTSP_REORDERING_OPTS(),
96 static const AVOption sdp_options[] = {
97 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
98 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
99 { "rtcp_to_source", "Send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
100 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
101 RTSP_REORDERING_OPTS(),
105 static const AVOption rtp_options[] = {
106 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
107 RTSP_REORDERING_OPTS(),
111 static void get_word_until_chars(char *buf, int buf_size,
112 const char *sep, const char **pp)
118 p += strspn(p, SPACE_CHARS);
120 while (!strchr(sep, *p) && *p != '\0') {
121 if ((q - buf) < buf_size - 1)
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
133 if (**pp == '/') (*pp)++;
134 get_word_until_chars(buf, buf_size, sep, pp);
137 static void get_word(char *buf, int buf_size, const char **pp)
139 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
142 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
144 * Used for seeking in the rtp stream.
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
150 p += strspn(p, SPACE_CHARS);
151 if (!av_stristart(p, "npt=", &p))
154 *start = AV_NOPTS_VALUE;
155 *end = AV_NOPTS_VALUE;
157 get_word_sep(buf, sizeof(buf), "-", &p);
158 av_parse_time(start, buf, 1);
161 get_word_sep(buf, sizeof(buf), "-", &p);
162 av_parse_time(end, buf, 1);
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
168 struct addrinfo hints = { 0 }, *ai = NULL;
169 hints.ai_flags = AI_NUMERICHOST;
170 if (getaddrinfo(buf, NULL, &hints, &ai))
172 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179 RTSPStream *rtsp_st, AVStream *st)
181 AVCodecContext *codec = st ? st->codec : NULL;
185 codec->codec_id = handler->codec_id;
186 rtsp_st->dynamic_handler = handler;
188 st->need_parsing = handler->need_parsing;
189 if (handler->priv_data_size) {
190 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
191 if (!rtsp_st->dynamic_protocol_context)
192 rtsp_st->dynamic_handler = NULL;
196 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
199 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
200 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
201 rtsp_st->dynamic_protocol_context);
203 if (rtsp_st->dynamic_protocol_context) {
204 if (rtsp_st->dynamic_handler->close)
205 rtsp_st->dynamic_handler->close(
206 rtsp_st->dynamic_protocol_context);
207 av_free(rtsp_st->dynamic_protocol_context);
209 rtsp_st->dynamic_protocol_context = NULL;
210 rtsp_st->dynamic_handler = NULL;
215 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
216 static int sdp_parse_rtpmap(AVFormatContext *s,
217 AVStream *st, RTSPStream *rtsp_st,
218 int payload_type, const char *p)
220 AVCodecContext *codec = st->codec;
226 /* See if we can handle this kind of payload.
227 * The space should normally not be there but some Real streams or
228 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
229 * have a trailing space. */
230 get_word_sep(buf, sizeof(buf), "/ ", &p);
231 if (payload_type < RTP_PT_PRIVATE) {
232 /* We are in a standard case
233 * (from http://www.iana.org/assignments/rtp-parameters). */
234 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
237 if (codec->codec_id == AV_CODEC_ID_NONE) {
238 RTPDynamicProtocolHandler *handler =
239 ff_rtp_handler_find_by_name(buf, codec->codec_type);
240 init_rtp_handler(handler, rtsp_st, st);
241 /* If no dynamic handler was found, check with the list of standard
242 * allocated types, if such a stream for some reason happens to
243 * use a private payload type. This isn't handled in rtpdec.c, since
244 * the format name from the rtpmap line never is passed into rtpdec. */
245 if (!rtsp_st->dynamic_handler)
246 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
249 c = avcodec_find_decoder(codec->codec_id);
255 get_word_sep(buf, sizeof(buf), "/", &p);
257 switch (codec->codec_type) {
258 case AVMEDIA_TYPE_AUDIO:
259 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
260 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
261 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
263 codec->sample_rate = i;
264 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
265 get_word_sep(buf, sizeof(buf), "/", &p);
270 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
272 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
275 case AVMEDIA_TYPE_VIDEO:
276 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
278 avpriv_set_pts_info(st, 32, 1, i);
283 finalize_rtp_handler_init(s, rtsp_st, st);
287 /* parse the attribute line from the fmtp a line of an sdp response. This
288 * is broken out as a function because it is used in rtp_h264.c, which is
290 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
291 char *value, int value_size)
293 *p += strspn(*p, SPACE_CHARS);
295 get_word_sep(attr, attr_size, "=", p);
298 get_word_sep(value, value_size, ";", p);
306 typedef struct SDPParseState {
308 struct sockaddr_storage default_ip;
310 int skip_media; ///< set if an unknown m= line occurs
311 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
312 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
313 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
314 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
317 char delayed_fmtp[2048];
320 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
321 struct RTSPSource ***dest, int *dest_count)
323 RTSPSource *rtsp_src, *rtsp_src2;
325 for (i = 0; i < count; i++) {
327 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
330 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
331 dynarray_add(dest, dest_count, rtsp_src2);
335 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
336 int payload_type, const char *line)
340 for (i = 0; i < rt->nb_rtsp_streams; i++) {
341 RTSPStream *rtsp_st = rt->rtsp_streams[i];
342 if (rtsp_st->sdp_payload_type == payload_type &&
343 rtsp_st->dynamic_handler &&
344 rtsp_st->dynamic_handler->parse_sdp_a_line) {
345 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
346 rtsp_st->dynamic_protocol_context, line);
351 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
352 int letter, const char *buf)
354 RTSPState *rt = s->priv_data;
355 char buf1[64], st_type[64];
357 enum AVMediaType codec_type;
361 RTSPSource *rtsp_src;
362 struct sockaddr_storage sdp_ip;
365 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
368 if (s1->skip_media && letter != 'm')
372 get_word(buf1, sizeof(buf1), &p);
373 if (strcmp(buf1, "IN") != 0)
375 get_word(buf1, sizeof(buf1), &p);
376 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
378 get_word_sep(buf1, sizeof(buf1), "/", &p);
379 if (get_sockaddr(buf1, &sdp_ip))
384 get_word_sep(buf1, sizeof(buf1), "/", &p);
387 if (s->nb_streams == 0) {
388 s1->default_ip = sdp_ip;
389 s1->default_ttl = ttl;
391 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
392 rtsp_st->sdp_ip = sdp_ip;
393 rtsp_st->sdp_ttl = ttl;
397 av_dict_set(&s->metadata, "title", p, 0);
400 if (s->nb_streams == 0) {
401 av_dict_set(&s->metadata, "comment", p, 0);
410 codec_type = AVMEDIA_TYPE_UNKNOWN;
411 get_word(st_type, sizeof(st_type), &p);
412 if (!strcmp(st_type, "audio")) {
413 codec_type = AVMEDIA_TYPE_AUDIO;
414 } else if (!strcmp(st_type, "video")) {
415 codec_type = AVMEDIA_TYPE_VIDEO;
416 } else if (!strcmp(st_type, "application") || !strcmp(st_type, "text")) {
417 codec_type = AVMEDIA_TYPE_DATA;
419 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
423 rtsp_st = av_mallocz(sizeof(RTSPStream));
426 rtsp_st->stream_index = -1;
427 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
429 rtsp_st->sdp_ip = s1->default_ip;
430 rtsp_st->sdp_ttl = s1->default_ttl;
432 copy_default_source_addrs(s1->default_include_source_addrs,
433 s1->nb_default_include_source_addrs,
434 &rtsp_st->include_source_addrs,
435 &rtsp_st->nb_include_source_addrs);
436 copy_default_source_addrs(s1->default_exclude_source_addrs,
437 s1->nb_default_exclude_source_addrs,
438 &rtsp_st->exclude_source_addrs,
439 &rtsp_st->nb_exclude_source_addrs);
441 get_word(buf1, sizeof(buf1), &p); /* port */
442 rtsp_st->sdp_port = atoi(buf1);
444 get_word(buf1, sizeof(buf1), &p); /* protocol */
445 if (!strcmp(buf1, "udp"))
446 rt->transport = RTSP_TRANSPORT_RAW;
447 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
448 rtsp_st->feedback = 1;
450 /* XXX: handle list of formats */
451 get_word(buf1, sizeof(buf1), &p); /* format list */
452 rtsp_st->sdp_payload_type = atoi(buf1);
454 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
455 /* no corresponding stream */
456 if (rt->transport == RTSP_TRANSPORT_RAW) {
457 if (CONFIG_RTPDEC && !rt->ts)
458 rt->ts = ff_mpegts_parse_open(s);
460 RTPDynamicProtocolHandler *handler;
461 handler = ff_rtp_handler_find_by_id(
462 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
463 init_rtp_handler(handler, rtsp_st, NULL);
464 finalize_rtp_handler_init(s, rtsp_st, NULL);
466 } else if (rt->server_type == RTSP_SERVER_WMS &&
467 codec_type == AVMEDIA_TYPE_DATA) {
468 /* RTX stream, a stream that carries all the other actual
469 * audio/video streams. Don't expose this to the callers. */
471 st = avformat_new_stream(s, NULL);
474 st->id = rt->nb_rtsp_streams - 1;
475 rtsp_st->stream_index = st->index;
476 st->codec->codec_type = codec_type;
477 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
478 RTPDynamicProtocolHandler *handler;
479 /* if standard payload type, we can find the codec right now */
480 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
481 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
482 st->codec->sample_rate > 0)
483 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
484 /* Even static payload types may need a custom depacketizer */
485 handler = ff_rtp_handler_find_by_id(
486 rtsp_st->sdp_payload_type, st->codec->codec_type);
487 init_rtp_handler(handler, rtsp_st, st);
488 finalize_rtp_handler_init(s, rtsp_st, st);
490 if (rt->default_lang[0])
491 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
493 /* put a default control url */
494 av_strlcpy(rtsp_st->control_url, rt->control_uri,
495 sizeof(rtsp_st->control_url));
498 if (av_strstart(p, "control:", &p)) {
499 if (s->nb_streams == 0) {
500 if (!strncmp(p, "rtsp://", 7))
501 av_strlcpy(rt->control_uri, p,
502 sizeof(rt->control_uri));
505 /* get the control url */
506 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
508 /* XXX: may need to add full url resolution */
509 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
511 if (proto[0] == '\0') {
512 /* relative control URL */
513 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
514 av_strlcat(rtsp_st->control_url, "/",
515 sizeof(rtsp_st->control_url));
516 av_strlcat(rtsp_st->control_url, p,
517 sizeof(rtsp_st->control_url));
519 av_strlcpy(rtsp_st->control_url, p,
520 sizeof(rtsp_st->control_url));
522 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
523 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
524 get_word(buf1, sizeof(buf1), &p);
525 payload_type = atoi(buf1);
526 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
527 if (rtsp_st->stream_index >= 0) {
528 st = s->streams[rtsp_st->stream_index];
529 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
533 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
535 } else if (av_strstart(p, "fmtp:", &p) ||
536 av_strstart(p, "framesize:", &p)) {
537 // let dynamic protocol handlers have a stab at the line.
538 get_word(buf1, sizeof(buf1), &p);
539 payload_type = atoi(buf1);
540 if (s1->seen_rtpmap) {
541 parse_fmtp(s, rt, payload_type, buf);
544 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
546 } else if (av_strstart(p, "range:", &p)) {
549 // this is so that seeking on a streamed file can work.
550 rtsp_parse_range_npt(p, &start, &end);
551 s->start_time = start;
552 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
553 s->duration = (end == AV_NOPTS_VALUE) ?
554 AV_NOPTS_VALUE : end - start;
555 } else if (av_strstart(p, "lang:", &p)) {
556 if (s->nb_streams > 0) {
557 get_word(buf1, sizeof(buf1), &p);
558 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
559 if (rtsp_st->stream_index >= 0) {
560 st = s->streams[rtsp_st->stream_index];
561 av_dict_set(&st->metadata, "language", buf1, 0);
564 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
565 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
567 rt->transport = RTSP_TRANSPORT_RDT;
568 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
570 st = s->streams[s->nb_streams - 1];
571 st->codec->sample_rate = atoi(p);
572 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
574 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
575 get_word(buf1, sizeof(buf1), &p); // ignore tag
576 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
577 p += strspn(p, SPACE_CHARS);
578 if (av_strstart(p, "inline:", &p))
579 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
580 } else if (av_strstart(p, "source-filter:", &p)) {
582 get_word(buf1, sizeof(buf1), &p);
583 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
585 exclude = !strcmp(buf1, "excl");
587 get_word(buf1, sizeof(buf1), &p);
588 if (strcmp(buf1, "IN") != 0)
590 get_word(buf1, sizeof(buf1), &p);
591 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
593 // not checking that the destination address actually matches or is wildcard
594 get_word(buf1, sizeof(buf1), &p);
597 rtsp_src = av_mallocz(sizeof(*rtsp_src));
600 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
602 if (s->nb_streams == 0) {
603 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
605 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
606 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
609 if (s->nb_streams == 0) {
610 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
612 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
613 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
618 if (rt->server_type == RTSP_SERVER_WMS)
619 ff_wms_parse_sdp_a_line(s, p);
620 if (s->nb_streams > 0) {
621 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
623 if (rt->server_type == RTSP_SERVER_REAL)
624 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
626 if (rtsp_st->dynamic_handler &&
627 rtsp_st->dynamic_handler->parse_sdp_a_line)
628 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
629 rtsp_st->stream_index,
630 rtsp_st->dynamic_protocol_context, buf);
637 int ff_sdp_parse(AVFormatContext *s, const char *content)
639 RTSPState *rt = s->priv_data;
642 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
643 * contain long SDP lines containing complete ASF Headers (several
644 * kB) or arrays of MDPR (RM stream descriptor) headers plus
645 * "rulebooks" describing their properties. Therefore, the SDP line
648 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
649 * in rtpdec_xiph.c. */
651 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
655 p += strspn(p, SPACE_CHARS);
663 /* get the content */
665 while (*p != '\n' && *p != '\r' && *p != '\0') {
666 if ((q - buf) < sizeof(buf) - 1)
671 sdp_parse_line(s, s1, letter, buf);
673 while (*p != '\n' && *p != '\0')
679 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
680 av_free(s1->default_include_source_addrs[i]);
681 av_freep(&s1->default_include_source_addrs);
682 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
683 av_free(s1->default_exclude_source_addrs[i]);
684 av_freep(&s1->default_exclude_source_addrs);
686 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
687 if (!rt->p) return AVERROR(ENOMEM);
690 #endif /* CONFIG_RTPDEC */
692 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
694 RTSPState *rt = s->priv_data;
697 for (i = 0; i < rt->nb_rtsp_streams; i++) {
698 RTSPStream *rtsp_st = rt->rtsp_streams[i];
701 if (rtsp_st->transport_priv) {
703 AVFormatContext *rtpctx = rtsp_st->transport_priv;
704 av_write_trailer(rtpctx);
705 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
706 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
707 ff_rtsp_tcp_write_packet(s, rtsp_st);
708 ffio_free_dyn_buf(&rtpctx->pb);
710 avio_close(rtpctx->pb);
712 avformat_free_context(rtpctx);
713 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
714 ff_rdt_parse_close(rtsp_st->transport_priv);
715 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
716 ff_rtp_parse_close(rtsp_st->transport_priv);
718 rtsp_st->transport_priv = NULL;
719 if (rtsp_st->rtp_handle)
720 ffurl_close(rtsp_st->rtp_handle);
721 rtsp_st->rtp_handle = NULL;
725 /* close and free RTSP streams */
726 void ff_rtsp_close_streams(AVFormatContext *s)
728 RTSPState *rt = s->priv_data;
732 ff_rtsp_undo_setup(s, 0);
733 for (i = 0; i < rt->nb_rtsp_streams; i++) {
734 rtsp_st = rt->rtsp_streams[i];
736 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
737 if (rtsp_st->dynamic_handler->close)
738 rtsp_st->dynamic_handler->close(
739 rtsp_st->dynamic_protocol_context);
740 av_free(rtsp_st->dynamic_protocol_context);
742 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
743 av_free(rtsp_st->include_source_addrs[j]);
744 av_freep(&rtsp_st->include_source_addrs);
745 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
746 av_free(rtsp_st->exclude_source_addrs[j]);
747 av_freep(&rtsp_st->exclude_source_addrs);
752 av_free(rt->rtsp_streams);
754 avformat_close_input(&rt->asf_ctx);
756 if (CONFIG_RTPDEC && rt->ts)
757 ff_mpegts_parse_close(rt->ts);
759 av_free(rt->recvbuf);
762 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
764 RTSPState *rt = s->priv_data;
766 int reordering_queue_size = rt->reordering_queue_size;
767 if (reordering_queue_size < 0) {
768 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
769 reordering_queue_size = 0;
771 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
774 /* open the RTP context */
775 if (rtsp_st->stream_index >= 0)
776 st = s->streams[rtsp_st->stream_index];
778 s->ctx_flags |= AVFMTCTX_NOHEADER;
780 if (CONFIG_RTSP_MUXER && s->oformat) {
781 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
782 s, st, rtsp_st->rtp_handle,
783 RTSP_TCP_MAX_PACKET_SIZE,
784 rtsp_st->stream_index);
785 /* Ownership of rtp_handle is passed to the rtp mux context */
786 rtsp_st->rtp_handle = NULL;
789 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
790 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
791 return 0; // Don't need to open any parser here
792 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
793 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
794 rtsp_st->dynamic_protocol_context,
795 rtsp_st->dynamic_handler);
796 else if (CONFIG_RTPDEC)
797 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
798 rtsp_st->sdp_payload_type,
799 reordering_queue_size);
801 if (!rtsp_st->transport_priv) {
802 return AVERROR(ENOMEM);
803 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
804 if (rtsp_st->dynamic_handler) {
805 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
806 rtsp_st->dynamic_protocol_context,
807 rtsp_st->dynamic_handler);
809 if (rtsp_st->crypto_suite[0])
810 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
811 rtsp_st->crypto_suite,
812 rtsp_st->crypto_params);
818 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
819 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
826 q += strspn(q, SPACE_CHARS);
827 v = strtol(q, &p, 10);
831 v = strtol(p, &p, 10);
840 /* XXX: only one transport specification is parsed */
841 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
843 char transport_protocol[16];
845 char lower_transport[16];
847 RTSPTransportField *th;
850 reply->nb_transports = 0;
853 p += strspn(p, SPACE_CHARS);
857 th = &reply->transports[reply->nb_transports];
859 get_word_sep(transport_protocol, sizeof(transport_protocol),
861 if (!av_strcasecmp (transport_protocol, "rtp")) {
862 get_word_sep(profile, sizeof(profile), "/;,", &p);
863 lower_transport[0] = '\0';
864 /* rtp/avp/<protocol> */
866 get_word_sep(lower_transport, sizeof(lower_transport),
869 th->transport = RTSP_TRANSPORT_RTP;
870 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
871 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
872 /* x-pn-tng/<protocol> */
873 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
875 th->transport = RTSP_TRANSPORT_RDT;
876 } else if (!av_strcasecmp(transport_protocol, "raw")) {
877 get_word_sep(profile, sizeof(profile), "/;,", &p);
878 lower_transport[0] = '\0';
879 /* raw/raw/<protocol> */
881 get_word_sep(lower_transport, sizeof(lower_transport),
884 th->transport = RTSP_TRANSPORT_RAW;
886 if (!av_strcasecmp(lower_transport, "TCP"))
887 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
889 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
893 /* get each parameter */
894 while (*p != '\0' && *p != ',') {
895 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
896 if (!strcmp(parameter, "port")) {
899 rtsp_parse_range(&th->port_min, &th->port_max, &p);
901 } else if (!strcmp(parameter, "client_port")) {
904 rtsp_parse_range(&th->client_port_min,
905 &th->client_port_max, &p);
907 } else if (!strcmp(parameter, "server_port")) {
910 rtsp_parse_range(&th->server_port_min,
911 &th->server_port_max, &p);
913 } else if (!strcmp(parameter, "interleaved")) {
916 rtsp_parse_range(&th->interleaved_min,
917 &th->interleaved_max, &p);
919 } else if (!strcmp(parameter, "multicast")) {
920 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
921 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
922 } else if (!strcmp(parameter, "ttl")) {
926 th->ttl = strtol(p, &end, 10);
929 } else if (!strcmp(parameter, "destination")) {
932 get_word_sep(buf, sizeof(buf), ";,", &p);
933 get_sockaddr(buf, &th->destination);
935 } else if (!strcmp(parameter, "source")) {
938 get_word_sep(buf, sizeof(buf), ";,", &p);
939 av_strlcpy(th->source, buf, sizeof(th->source));
941 } else if (!strcmp(parameter, "mode")) {
944 get_word_sep(buf, sizeof(buf), ";, ", &p);
945 if (!strcmp(buf, "record") ||
946 !strcmp(buf, "receive"))
951 while (*p != ';' && *p != '\0' && *p != ',')
959 reply->nb_transports++;
963 static void handle_rtp_info(RTSPState *rt, const char *url,
964 uint32_t seq, uint32_t rtptime)
967 if (!rtptime || !url[0])
969 if (rt->transport != RTSP_TRANSPORT_RTP)
971 for (i = 0; i < rt->nb_rtsp_streams; i++) {
972 RTSPStream *rtsp_st = rt->rtsp_streams[i];
973 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
976 if (!strcmp(rtsp_st->control_url, url)) {
977 rtpctx->base_timestamp = rtptime;
983 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
986 char key[20], value[1024], url[1024] = "";
987 uint32_t seq = 0, rtptime = 0;
990 p += strspn(p, SPACE_CHARS);
993 get_word_sep(key, sizeof(key), "=", &p);
997 get_word_sep(value, sizeof(value), ";, ", &p);
999 if (!strcmp(key, "url"))
1000 av_strlcpy(url, value, sizeof(url));
1001 else if (!strcmp(key, "seq"))
1002 seq = strtoul(value, NULL, 10);
1003 else if (!strcmp(key, "rtptime"))
1004 rtptime = strtoul(value, NULL, 10);
1006 handle_rtp_info(rt, url, seq, rtptime);
1015 handle_rtp_info(rt, url, seq, rtptime);
1018 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1019 RTSPState *rt, const char *method)
1023 /* NOTE: we do case independent match for broken servers */
1025 if (av_stristart(p, "Session:", &p)) {
1027 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1028 if (av_stristart(p, ";timeout=", &p) &&
1029 (t = strtol(p, NULL, 10)) > 0) {
1032 } else if (av_stristart(p, "Content-Length:", &p)) {
1033 reply->content_length = strtol(p, NULL, 10);
1034 } else if (av_stristart(p, "Transport:", &p)) {
1035 rtsp_parse_transport(reply, p);
1036 } else if (av_stristart(p, "CSeq:", &p)) {
1037 reply->seq = strtol(p, NULL, 10);
1038 } else if (av_stristart(p, "Range:", &p)) {
1039 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1040 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1041 p += strspn(p, SPACE_CHARS);
1042 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1043 } else if (av_stristart(p, "Server:", &p)) {
1044 p += strspn(p, SPACE_CHARS);
1045 av_strlcpy(reply->server, p, sizeof(reply->server));
1046 } else if (av_stristart(p, "Notice:", &p) ||
1047 av_stristart(p, "X-Notice:", &p)) {
1048 reply->notice = strtol(p, NULL, 10);
1049 } else if (av_stristart(p, "Location:", &p)) {
1050 p += strspn(p, SPACE_CHARS);
1051 av_strlcpy(reply->location, p , sizeof(reply->location));
1052 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1053 p += strspn(p, SPACE_CHARS);
1054 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1055 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1056 p += strspn(p, SPACE_CHARS);
1057 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1058 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1059 p += strspn(p, SPACE_CHARS);
1060 if (method && !strcmp(method, "DESCRIBE"))
1061 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1062 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1063 p += strspn(p, SPACE_CHARS);
1064 if (method && !strcmp(method, "PLAY"))
1065 rtsp_parse_rtp_info(rt, p);
1066 } else if (av_stristart(p, "Public:", &p) && rt) {
1067 if (strstr(p, "GET_PARAMETER") &&
1068 method && !strcmp(method, "OPTIONS"))
1069 rt->get_parameter_supported = 1;
1070 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1071 p += strspn(p, SPACE_CHARS);
1072 rt->accept_dynamic_rate = atoi(p);
1073 } else if (av_stristart(p, "Content-Type:", &p)) {
1074 p += strspn(p, SPACE_CHARS);
1075 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1079 /* skip a RTP/TCP interleaved packet */
1080 void ff_rtsp_skip_packet(AVFormatContext *s)
1082 RTSPState *rt = s->priv_data;
1086 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1089 len = AV_RB16(buf + 1);
1091 av_dlog(s, "skipping RTP packet len=%d\n", len);
1096 if (len1 > sizeof(buf))
1098 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1105 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1106 unsigned char **content_ptr,
1107 int return_on_interleaved_data, const char *method)
1109 RTSPState *rt = s->priv_data;
1110 char buf[4096], buf1[1024], *q;
1113 int ret, content_length, line_count = 0, request = 0;
1114 unsigned char *content = NULL;
1120 memset(reply, 0, sizeof(*reply));
1122 /* parse reply (XXX: use buffers) */
1123 rt->last_reply[0] = '\0';
1127 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1128 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1134 /* XXX: only parse it if first char on line ? */
1135 if (return_on_interleaved_data) {
1138 ff_rtsp_skip_packet(s);
1139 } else if (ch != '\r') {
1140 if ((q - buf) < sizeof(buf) - 1)
1146 av_dlog(s, "line='%s'\n", buf);
1148 /* test if last line */
1152 if (line_count == 0) {
1153 /* get reply code */
1154 get_word(buf1, sizeof(buf1), &p);
1155 if (!strncmp(buf1, "RTSP/", 5)) {
1156 get_word(buf1, sizeof(buf1), &p);
1157 reply->status_code = atoi(buf1);
1158 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1160 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1161 get_word(buf1, sizeof(buf1), &p); // object
1165 ff_rtsp_parse_line(reply, p, rt, method);
1166 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1167 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1172 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1173 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1175 content_length = reply->content_length;
1176 if (content_length > 0) {
1177 /* leave some room for a trailing '\0' (useful for simple parsing) */
1178 content = av_malloc(content_length + 1);
1180 return AVERROR(ENOMEM);
1181 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1182 content[content_length] = '\0';
1185 *content_ptr = content;
1191 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1192 const char* ptr = buf;
1194 if (!strcmp(reply->reason, "OPTIONS")) {
1195 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1197 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1198 if (reply->session_id[0])
1199 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1202 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1204 av_strlcat(buf, "\r\n", sizeof(buf));
1206 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1207 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1210 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1212 rt->last_cmd_time = av_gettime_relative();
1213 /* Even if the request from the server had data, it is not the data
1214 * that the caller wants or expects. The memory could also be leaked
1215 * if the actual following reply has content data. */
1217 av_freep(content_ptr);
1218 /* If method is set, this is called from ff_rtsp_send_cmd,
1219 * where a reply to exactly this request is awaited. For
1220 * callers from within packet receiving, we just want to
1221 * return to the caller and go back to receiving packets. */
1227 if (rt->seq != reply->seq) {
1228 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1229 rt->seq, reply->seq);
1233 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1234 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1235 reply->notice == 2306 /* Continuous Feed Terminated */) {
1236 rt->state = RTSP_STATE_IDLE;
1237 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1238 return AVERROR(EIO); /* data or server error */
1239 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1240 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1241 return AVERROR(EPERM);
1247 * Send a command to the RTSP server without waiting for the reply.
1249 * @param s RTSP (de)muxer context
1250 * @param method the method for the request
1251 * @param url the target url for the request
1252 * @param headers extra header lines to include in the request
1253 * @param send_content if non-null, the data to send as request body content
1254 * @param send_content_length the length of the send_content data, or 0 if
1255 * send_content is null
1257 * @return zero if success, nonzero otherwise
1259 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1260 const char *method, const char *url,
1261 const char *headers,
1262 const unsigned char *send_content,
1263 int send_content_length)
1265 RTSPState *rt = s->priv_data;
1266 char buf[4096], *out_buf;
1267 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1269 /* Add in RTSP headers */
1272 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1274 av_strlcat(buf, headers, sizeof(buf));
1275 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1276 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1277 if (rt->session_id[0] != '\0' && (!headers ||
1278 !strstr(headers, "\nIf-Match:"))) {
1279 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1282 char *str = ff_http_auth_create_response(&rt->auth_state,
1283 rt->auth, url, method);
1285 av_strlcat(buf, str, sizeof(buf));
1288 if (send_content_length > 0 && send_content)
1289 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1290 av_strlcat(buf, "\r\n", sizeof(buf));
1292 /* base64 encode rtsp if tunneling */
1293 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1294 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1295 out_buf = base64buf;
1298 av_dlog(s, "Sending:\n%s--\n", buf);
1300 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1301 if (send_content_length > 0 && send_content) {
1302 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1303 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1304 "with content data not supported\n");
1305 return AVERROR_PATCHWELCOME;
1307 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1309 rt->last_cmd_time = av_gettime_relative();
1314 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1315 const char *url, const char *headers)
1317 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1320 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1321 const char *headers, RTSPMessageHeader *reply,
1322 unsigned char **content_ptr)
1324 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1325 content_ptr, NULL, 0);
1328 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1329 const char *method, const char *url,
1331 RTSPMessageHeader *reply,
1332 unsigned char **content_ptr,
1333 const unsigned char *send_content,
1334 int send_content_length)
1336 RTSPState *rt = s->priv_data;
1337 HTTPAuthType cur_auth_type;
1338 int ret, attempts = 0;
1341 cur_auth_type = rt->auth_state.auth_type;
1342 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1344 send_content_length)))
1347 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1351 if (reply->status_code == 401 &&
1352 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1353 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1356 if (reply->status_code > 400){
1357 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1361 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1367 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1368 int lower_transport, const char *real_challenge)
1370 RTSPState *rt = s->priv_data;
1371 int rtx = 0, j, i, err, interleave = 0, port_off;
1372 RTSPStream *rtsp_st;
1373 RTSPMessageHeader reply1, *reply = &reply1;
1375 const char *trans_pref;
1377 if (rt->transport == RTSP_TRANSPORT_RDT)
1378 trans_pref = "x-pn-tng";
1379 else if (rt->transport == RTSP_TRANSPORT_RAW)
1380 trans_pref = "RAW/RAW";
1382 trans_pref = "RTP/AVP";
1384 /* default timeout: 1 minute */
1387 /* for each stream, make the setup request */
1388 /* XXX: we assume the same server is used for the control of each
1391 /* Choose a random starting offset within the first half of the
1392 * port range, to allow for a number of ports to try even if the offset
1393 * happens to be at the end of the random range. */
1394 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1395 /* even random offset */
1396 port_off -= port_off & 0x01;
1398 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1399 char transport[2048];
1402 * WMS serves all UDP data over a single connection, the RTX, which
1403 * isn't necessarily the first in the SDP but has to be the first
1404 * to be set up, else the second/third SETUP will fail with a 461.
1406 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1407 rt->server_type == RTSP_SERVER_WMS) {
1410 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1411 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1413 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1417 if (rtx == rt->nb_rtsp_streams)
1418 return -1; /* no RTX found */
1419 rtsp_st = rt->rtsp_streams[rtx];
1421 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1423 rtsp_st = rt->rtsp_streams[i];
1426 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1429 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1430 port = reply->transports[0].client_port_min;
1434 /* first try in specified port range */
1435 while (j <= rt->rtp_port_max) {
1436 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1437 "?localport=%d", j);
1438 /* we will use two ports per rtp stream (rtp and rtcp) */
1440 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1441 &s->interrupt_callback, NULL))
1445 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1450 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1452 snprintf(transport, sizeof(transport) - 1,
1453 "%s/UDP;", trans_pref);
1454 if (rt->server_type != RTSP_SERVER_REAL)
1455 av_strlcat(transport, "unicast;", sizeof(transport));
1456 av_strlcatf(transport, sizeof(transport),
1457 "client_port=%d", port);
1458 if (rt->transport == RTSP_TRANSPORT_RTP &&
1459 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1460 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1464 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1465 /* For WMS streams, the application streams are only used for
1466 * UDP. When trying to set it up for TCP streams, the server
1467 * will return an error. Therefore, we skip those streams. */
1468 if (rt->server_type == RTSP_SERVER_WMS &&
1469 (rtsp_st->stream_index < 0 ||
1470 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1473 snprintf(transport, sizeof(transport) - 1,
1474 "%s/TCP;", trans_pref);
1475 if (rt->transport != RTSP_TRANSPORT_RDT)
1476 av_strlcat(transport, "unicast;", sizeof(transport));
1477 av_strlcatf(transport, sizeof(transport),
1478 "interleaved=%d-%d",
1479 interleave, interleave + 1);
1483 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1484 snprintf(transport, sizeof(transport) - 1,
1485 "%s/UDP;multicast", trans_pref);
1488 av_strlcat(transport, ";mode=record", sizeof(transport));
1489 } else if (rt->server_type == RTSP_SERVER_REAL ||
1490 rt->server_type == RTSP_SERVER_WMS)
1491 av_strlcat(transport, ";mode=play", sizeof(transport));
1492 snprintf(cmd, sizeof(cmd),
1493 "Transport: %s\r\n",
1495 if (rt->accept_dynamic_rate)
1496 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1497 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1498 char real_res[41], real_csum[9];
1499 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1501 av_strlcatf(cmd, sizeof(cmd),
1503 "RealChallenge2: %s, sd=%s\r\n",
1504 rt->session_id, real_res, real_csum);
1506 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1507 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1510 } else if (reply->status_code != RTSP_STATUS_OK ||
1511 reply->nb_transports != 1) {
1512 err = AVERROR_INVALIDDATA;
1516 /* XXX: same protocol for all streams is required */
1518 if (reply->transports[0].lower_transport != rt->lower_transport ||
1519 reply->transports[0].transport != rt->transport) {
1520 err = AVERROR_INVALIDDATA;
1524 rt->lower_transport = reply->transports[0].lower_transport;
1525 rt->transport = reply->transports[0].transport;
1528 /* Fail if the server responded with another lower transport mode
1529 * than what we requested. */
1530 if (reply->transports[0].lower_transport != lower_transport) {
1531 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1532 err = AVERROR_INVALIDDATA;
1536 switch(reply->transports[0].lower_transport) {
1537 case RTSP_LOWER_TRANSPORT_TCP:
1538 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1539 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1542 case RTSP_LOWER_TRANSPORT_UDP: {
1543 char url[1024], options[30] = "";
1544 const char *peer = host;
1546 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1547 av_strlcpy(options, "?connect=1", sizeof(options));
1548 /* Use source address if specified */
1549 if (reply->transports[0].source[0])
1550 peer = reply->transports[0].source;
1551 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1552 reply->transports[0].server_port_min, "%s", options);
1553 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1554 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1555 err = AVERROR_INVALIDDATA;
1560 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1561 char url[1024], namebuf[50], optbuf[20] = "";
1562 struct sockaddr_storage addr;
1565 if (reply->transports[0].destination.ss_family) {
1566 addr = reply->transports[0].destination;
1567 port = reply->transports[0].port_min;
1568 ttl = reply->transports[0].ttl;
1570 addr = rtsp_st->sdp_ip;
1571 port = rtsp_st->sdp_port;
1572 ttl = rtsp_st->sdp_ttl;
1575 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1576 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1577 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1578 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1579 port, "%s", optbuf);
1580 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1581 &s->interrupt_callback, NULL) < 0) {
1582 err = AVERROR_INVALIDDATA;
1589 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1593 if (rt->nb_rtsp_streams && reply->timeout > 0)
1594 rt->timeout = reply->timeout;
1596 if (rt->server_type == RTSP_SERVER_REAL)
1597 rt->need_subscription = 1;
1602 ff_rtsp_undo_setup(s, 0);
1606 void ff_rtsp_close_connections(AVFormatContext *s)
1608 RTSPState *rt = s->priv_data;
1609 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1610 ffurl_close(rt->rtsp_hd);
1611 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1614 int ff_rtsp_connect(AVFormatContext *s)
1616 RTSPState *rt = s->priv_data;
1617 char proto[128], host[1024], path[1024];
1618 char tcpname[1024], cmd[2048], auth[128];
1619 const char *lower_rtsp_proto = "tcp";
1620 int port, err, tcp_fd;
1621 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1622 int lower_transport_mask = 0;
1623 int default_port = RTSP_DEFAULT_PORT;
1624 char real_challenge[64] = "";
1625 struct sockaddr_storage peer;
1626 socklen_t peer_len = sizeof(peer);
1628 if (rt->rtp_port_max < rt->rtp_port_min) {
1629 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1630 "than min port %d\n", rt->rtp_port_max,
1632 return AVERROR(EINVAL);
1635 if (!ff_network_init())
1636 return AVERROR(EIO);
1638 if (s->max_delay < 0) /* Not set by the caller */
1639 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1641 rt->control_transport = RTSP_MODE_PLAIN;
1642 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1643 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1644 rt->control_transport = RTSP_MODE_TUNNEL;
1646 /* Only pass through valid flags from here */
1647 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1650 /* extract hostname and port */
1651 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1652 host, sizeof(host), &port, path, sizeof(path), s->filename);
1654 if (!strcmp(proto, "rtsps")) {
1655 lower_rtsp_proto = "tls";
1656 default_port = RTSPS_DEFAULT_PORT;
1657 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1661 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1664 port = default_port;
1666 lower_transport_mask = rt->lower_transport_mask;
1668 if (!lower_transport_mask)
1669 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1672 /* Only UDP or TCP - UDP multicast isn't supported. */
1673 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1674 (1 << RTSP_LOWER_TRANSPORT_TCP);
1675 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1676 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1677 "only UDP and TCP are supported for output.\n");
1678 err = AVERROR(EINVAL);
1683 /* Construct the URI used in request; this is similar to s->filename,
1684 * but with authentication credentials removed and RTSP specific options
1686 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1687 host, port, "%s", path);
1689 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1690 /* set up initial handshake for tunneling */
1691 char httpname[1024];
1692 char sessioncookie[17];
1695 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1696 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1697 av_get_random_seed(), av_get_random_seed());
1700 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1701 &s->interrupt_callback) < 0) {
1706 /* generate GET headers */
1707 snprintf(headers, sizeof(headers),
1708 "x-sessioncookie: %s\r\n"
1709 "Accept: application/x-rtsp-tunnelled\r\n"
1710 "Pragma: no-cache\r\n"
1711 "Cache-Control: no-cache\r\n",
1713 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1715 /* complete the connection */
1716 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1722 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1723 &s->interrupt_callback) < 0 ) {
1728 /* generate POST headers */
1729 snprintf(headers, sizeof(headers),
1730 "x-sessioncookie: %s\r\n"
1731 "Content-Type: application/x-rtsp-tunnelled\r\n"
1732 "Pragma: no-cache\r\n"
1733 "Cache-Control: no-cache\r\n"
1734 "Content-Length: 32767\r\n"
1735 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1737 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1738 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1740 /* Initialize the authentication state for the POST session. The HTTP
1741 * protocol implementation doesn't properly handle multi-pass
1742 * authentication for POST requests, since it would require one of
1744 * - implementing Expect: 100-continue, which many HTTP servers
1745 * don't support anyway, even less the RTSP servers that do HTTP
1747 * - sending the whole POST data until getting a 401 reply specifying
1748 * what authentication method to use, then resending all that data
1749 * - waiting for potential 401 replies directly after sending the
1750 * POST header (waiting for some unspecified time)
1751 * Therefore, we copy the full auth state, which works for both basic
1752 * and digest. (For digest, we would have to synchronize the nonce
1753 * count variable between the two sessions, if we'd do more requests
1754 * with the original session, though.)
1756 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1758 /* complete the connection */
1759 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1764 /* open the tcp connection */
1765 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1767 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1768 &s->interrupt_callback, NULL) < 0) {
1772 rt->rtsp_hd_out = rt->rtsp_hd;
1776 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1781 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1782 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1783 NULL, 0, NI_NUMERICHOST);
1786 /* request options supported by the server; this also detects server
1788 for (rt->server_type = RTSP_SERVER_RTP;;) {
1790 if (rt->server_type == RTSP_SERVER_REAL)
1793 * The following entries are required for proper
1794 * streaming from a Realmedia server. They are
1795 * interdependent in some way although we currently
1796 * don't quite understand how. Values were copied
1797 * from mplayer SVN r23589.
1798 * ClientChallenge is a 16-byte ID in hex
1799 * CompanyID is a 16-byte ID in base64
1801 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1802 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1803 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1804 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1806 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1807 if (reply->status_code != RTSP_STATUS_OK) {
1808 err = AVERROR_INVALIDDATA;
1812 /* detect server type if not standard-compliant RTP */
1813 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1814 rt->server_type = RTSP_SERVER_REAL;
1816 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1817 rt->server_type = RTSP_SERVER_WMS;
1818 } else if (rt->server_type == RTSP_SERVER_REAL)
1819 strcpy(real_challenge, reply->real_challenge);
1823 if (CONFIG_RTSP_DEMUXER && s->iformat)
1824 err = ff_rtsp_setup_input_streams(s, reply);
1825 else if (CONFIG_RTSP_MUXER)
1826 err = ff_rtsp_setup_output_streams(s, host);
1831 int lower_transport = ff_log2_tab[lower_transport_mask &
1832 ~(lower_transport_mask - 1)];
1834 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1835 rt->server_type == RTSP_SERVER_REAL ?
1836 real_challenge : NULL);
1839 lower_transport_mask &= ~(1 << lower_transport);
1840 if (lower_transport_mask == 0 && err == 1) {
1841 err = AVERROR(EPROTONOSUPPORT);
1846 rt->lower_transport_mask = lower_transport_mask;
1847 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1848 rt->state = RTSP_STATE_IDLE;
1849 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1852 ff_rtsp_close_streams(s);
1853 ff_rtsp_close_connections(s);
1854 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1855 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1856 rt->session_id[0] = '\0';
1857 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1865 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1868 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1869 uint8_t *buf, int buf_size, int64_t wait_end)
1871 RTSPState *rt = s->priv_data;
1872 RTSPStream *rtsp_st;
1873 int n, i, ret, tcp_fd, timeout_cnt = 0;
1875 struct pollfd *p = rt->p;
1876 int *fds = NULL, fdsnum, fdsidx;
1879 if (ff_check_interrupt(&s->interrupt_callback))
1880 return AVERROR_EXIT;
1881 if (wait_end && wait_end - av_gettime_relative() < 0)
1882 return AVERROR(EAGAIN);
1885 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1886 p[max_p].fd = tcp_fd;
1887 p[max_p++].events = POLLIN;
1891 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1892 rtsp_st = rt->rtsp_streams[i];
1893 if (rtsp_st->rtp_handle) {
1894 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1896 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1900 av_log(s, AV_LOG_ERROR,
1901 "Number of fds %d not supported\n", fdsnum);
1902 return AVERROR_INVALIDDATA;
1904 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1905 p[max_p].fd = fds[fdsidx];
1906 p[max_p++].events = POLLIN;
1911 n = poll(p, max_p, POLL_TIMEOUT_MS);
1913 int j = 1 - (tcp_fd == -1);
1915 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1916 rtsp_st = rt->rtsp_streams[i];
1917 if (rtsp_st->rtp_handle) {
1918 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1919 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1921 *prtsp_st = rtsp_st;
1928 #if CONFIG_RTSP_DEMUXER
1929 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1930 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1931 if (rt->state == RTSP_STATE_STREAMING) {
1932 if (!ff_rtsp_parse_streaming_commands(s))
1935 av_log(s, AV_LOG_WARNING,
1936 "Unable to answer to TEARDOWN\n");
1940 RTSPMessageHeader reply;
1941 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1944 /* XXX: parse message */
1945 if (rt->state != RTSP_STATE_STREAMING)
1950 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1951 return AVERROR(ETIMEDOUT);
1952 } else if (n < 0 && errno != EINTR)
1953 return AVERROR(errno);
1957 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1958 const uint8_t *buf, int len)
1960 RTSPState *rt = s->priv_data;
1964 if (rt->nb_rtsp_streams == 1) {
1965 *rtsp_st = rt->rtsp_streams[0];
1968 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1969 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1971 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1972 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1975 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1976 *rtsp_st = rt->rtsp_streams[i];
1983 av_log(s, AV_LOG_WARNING,
1984 "Unable to pick stream for packet - SSRC not known for "
1986 return AVERROR(EAGAIN);
1989 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1990 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1991 *rtsp_st = rt->rtsp_streams[i];
1997 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1998 return AVERROR(EAGAIN);
2001 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2003 RTSPState *rt = s->priv_data;
2005 RTSPStream *rtsp_st, *first_queue_st = NULL;
2006 int64_t wait_end = 0;
2008 if (rt->nb_byes == rt->nb_rtsp_streams)
2011 /* get next frames from the same RTP packet */
2012 if (rt->cur_transport_priv) {
2013 if (rt->transport == RTSP_TRANSPORT_RDT) {
2014 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2015 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2016 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2017 } else if (CONFIG_RTPDEC && rt->ts) {
2018 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2020 rt->recvbuf_pos += ret;
2021 ret = rt->recvbuf_pos < rt->recvbuf_len;
2026 rt->cur_transport_priv = NULL;
2028 } else if (ret == 1) {
2031 rt->cur_transport_priv = NULL;
2035 if (rt->transport == RTSP_TRANSPORT_RTP) {
2037 int64_t first_queue_time = 0;
2038 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2039 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2043 queue_time = ff_rtp_queued_packet_time(rtpctx);
2044 if (queue_time && (queue_time - first_queue_time < 0 ||
2045 !first_queue_time)) {
2046 first_queue_time = queue_time;
2047 first_queue_st = rt->rtsp_streams[i];
2050 if (first_queue_time) {
2051 wait_end = first_queue_time + s->max_delay;
2054 first_queue_st = NULL;
2058 /* read next RTP packet */
2060 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2062 return AVERROR(ENOMEM);
2065 switch(rt->lower_transport) {
2067 #if CONFIG_RTSP_DEMUXER
2068 case RTSP_LOWER_TRANSPORT_TCP:
2069 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2072 case RTSP_LOWER_TRANSPORT_UDP:
2073 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2074 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2075 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2076 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2078 case RTSP_LOWER_TRANSPORT_CUSTOM:
2079 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2080 wait_end && wait_end < av_gettime_relative())
2081 len = AVERROR(EAGAIN);
2083 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2084 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2085 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2086 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2089 if (len == AVERROR(EAGAIN) && first_queue_st &&
2090 rt->transport == RTSP_TRANSPORT_RTP) {
2091 rtsp_st = first_queue_st;
2092 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2099 if (rt->transport == RTSP_TRANSPORT_RDT) {
2100 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2101 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2102 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2103 if (rtsp_st->feedback) {
2104 AVIOContext *pb = NULL;
2105 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2107 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2110 /* Either bad packet, or a RTCP packet. Check if the
2111 * first_rtcp_ntp_time field was initialized. */
2112 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2113 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2114 /* first_rtcp_ntp_time has been initialized for this stream,
2115 * copy the same value to all other uninitialized streams,
2116 * in order to map their timestamp origin to the same ntp time
2119 AVStream *st = NULL;
2120 if (rtsp_st->stream_index >= 0)
2121 st = s->streams[rtsp_st->stream_index];
2122 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2123 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2124 AVStream *st2 = NULL;
2125 if (rt->rtsp_streams[i]->stream_index >= 0)
2126 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2127 if (rtpctx2 && st && st2 &&
2128 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2129 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2130 rtpctx2->rtcp_ts_offset = av_rescale_q(
2131 rtpctx->rtcp_ts_offset, st->time_base,
2136 if (ret == -RTCP_BYE) {
2139 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2140 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2142 if (rt->nb_byes == rt->nb_rtsp_streams)
2146 } else if (CONFIG_RTPDEC && rt->ts) {
2147 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2150 rt->recvbuf_len = len;
2151 rt->recvbuf_pos = ret;
2152 rt->cur_transport_priv = rt->ts;
2159 return AVERROR_INVALIDDATA;
2165 /* more packets may follow, so we save the RTP context */
2166 rt->cur_transport_priv = rtsp_st->transport_priv;
2170 #endif /* CONFIG_RTPDEC */
2172 #if CONFIG_SDP_DEMUXER
2173 static int sdp_probe(AVProbeData *p1)
2175 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2177 /* we look for a line beginning "c=IN IP" */
2178 while (p < p_end && *p != '\0') {
2179 if (p + sizeof("c=IN IP") - 1 < p_end &&
2180 av_strstart(p, "c=IN IP", NULL))
2181 return AVPROBE_SCORE_EXTENSION;
2183 while (p < p_end - 1 && *p != '\n') p++;
2192 static void append_source_addrs(char *buf, int size, const char *name,
2193 int count, struct RTSPSource **addrs)
2198 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2199 for (i = 1; i < count; i++)
2200 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2203 static int sdp_read_header(AVFormatContext *s)
2205 RTSPState *rt = s->priv_data;
2206 RTSPStream *rtsp_st;
2211 if (!ff_network_init())
2212 return AVERROR(EIO);
2214 if (s->max_delay < 0) /* Not set by the caller */
2215 s->max_delay = DEFAULT_REORDERING_DELAY;
2216 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2217 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2219 /* read the whole sdp file */
2220 /* XXX: better loading */
2221 content = av_malloc(SDP_MAX_SIZE);
2222 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2225 return AVERROR_INVALIDDATA;
2227 content[size] ='\0';
2229 err = ff_sdp_parse(s, content);
2233 /* open each RTP stream */
2234 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2236 rtsp_st = rt->rtsp_streams[i];
2238 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2239 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2240 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2241 ff_url_join(url, sizeof(url), "rtp", NULL,
2242 namebuf, rtsp_st->sdp_port,
2243 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2244 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2245 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2246 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2248 append_source_addrs(url, sizeof(url), "sources",
2249 rtsp_st->nb_include_source_addrs,
2250 rtsp_st->include_source_addrs);
2251 append_source_addrs(url, sizeof(url), "block",
2252 rtsp_st->nb_exclude_source_addrs,
2253 rtsp_st->exclude_source_addrs);
2254 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2255 &s->interrupt_callback, NULL) < 0) {
2256 err = AVERROR_INVALIDDATA;
2260 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2265 ff_rtsp_close_streams(s);
2270 static int sdp_read_close(AVFormatContext *s)
2272 ff_rtsp_close_streams(s);
2277 static const AVClass sdp_demuxer_class = {
2278 .class_name = "SDP demuxer",
2279 .item_name = av_default_item_name,
2280 .option = sdp_options,
2281 .version = LIBAVUTIL_VERSION_INT,
2284 AVInputFormat ff_sdp_demuxer = {
2286 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2287 .priv_data_size = sizeof(RTSPState),
2288 .read_probe = sdp_probe,
2289 .read_header = sdp_read_header,
2290 .read_packet = ff_rtsp_fetch_packet,
2291 .read_close = sdp_read_close,
2292 .priv_class = &sdp_demuxer_class,
2294 #endif /* CONFIG_SDP_DEMUXER */
2296 #if CONFIG_RTP_DEMUXER
2297 static int rtp_probe(AVProbeData *p)
2299 if (av_strstart(p->filename, "rtp:", NULL))
2300 return AVPROBE_SCORE_MAX;
2304 static int rtp_read_header(AVFormatContext *s)
2306 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2307 char host[500], sdp[500];
2309 URLContext* in = NULL;
2311 AVCodecContext codec = { 0 };
2312 struct sockaddr_storage addr;
2314 socklen_t addrlen = sizeof(addr);
2315 RTSPState *rt = s->priv_data;
2317 if (!ff_network_init())
2318 return AVERROR(EIO);
2320 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2321 &s->interrupt_callback, NULL);
2326 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2327 if (ret == AVERROR(EAGAIN))
2332 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2336 if ((recvbuf[0] & 0xc0) != 0x80) {
2337 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2342 if (RTP_PT_IS_RTCP(recvbuf[1]))
2345 payload_type = recvbuf[1] & 0x7f;
2348 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2352 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2353 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2354 "without an SDP file describing it\n",
2358 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2359 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2360 "properly you need an SDP file "
2364 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2365 NULL, 0, s->filename);
2367 snprintf(sdp, sizeof(sdp),
2368 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2369 addr.ss_family == AF_INET ? 4 : 6, host,
2370 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2371 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2372 port, payload_type);
2373 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2375 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2378 /* sdp_read_header initializes this again */
2381 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2383 ret = sdp_read_header(s);
2394 static const AVClass rtp_demuxer_class = {
2395 .class_name = "RTP demuxer",
2396 .item_name = av_default_item_name,
2397 .option = rtp_options,
2398 .version = LIBAVUTIL_VERSION_INT,
2401 AVInputFormat ff_rtp_demuxer = {
2403 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2404 .priv_data_size = sizeof(RTSPState),
2405 .read_probe = rtp_probe,
2406 .read_header = rtp_read_header,
2407 .read_packet = ff_rtsp_fetch_packet,
2408 .read_close = sdp_read_close,
2409 .flags = AVFMT_NOFILE,
2410 .priv_class = &rtp_demuxer_class,
2412 #endif /* CONFIG_RTP_DEMUXER */