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/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "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" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
77 #define COMMON_OPTS() \
78 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
79 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
82 const AVOption ff_rtsp_options[] = {
83 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
84 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
99 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
103 static const AVOption sdp_options[] = {
104 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106 { "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" },
107 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
112 static const AVOption rtp_options[] = {
113 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
119 static AVDictionary *map_to_opts(RTSPState *rt)
121 AVDictionary *opts = NULL;
124 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125 av_dict_set(&opts, "buffer_size", buf, 0);
130 static void get_word_until_chars(char *buf, int buf_size,
131 const char *sep, const char **pp)
137 p += strspn(p, SPACE_CHARS);
139 while (!strchr(sep, *p) && *p != '\0') {
140 if ((q - buf) < buf_size - 1)
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
152 if (**pp == '/') (*pp)++;
153 get_word_until_chars(buf, buf_size, sep, pp);
156 static void get_word(char *buf, int buf_size, const char **pp)
158 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
163 * Used for seeking in the rtp stream.
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
169 p += strspn(p, SPACE_CHARS);
170 if (!av_stristart(p, "npt=", &p))
173 *start = AV_NOPTS_VALUE;
174 *end = AV_NOPTS_VALUE;
176 get_word_sep(buf, sizeof(buf), "-", &p);
177 if (av_parse_time(start, buf, 1) < 0) {
178 av_log(NULL, AV_LOG_ERROR, "Invalid interval start specification '%s'\n", buf);
183 get_word_sep(buf, sizeof(buf), "-", &p);
184 if (av_parse_time(end, buf, 1) < 0)
185 av_log(NULL, AV_LOG_ERROR, "Invalid interval end specification '%s'\n", buf);
189 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
191 struct addrinfo hints = { 0 }, *ai = NULL;
192 hints.ai_flags = AI_NUMERICHOST;
193 if (getaddrinfo(buf, NULL, &hints, &ai))
195 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
201 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
202 RTSPStream *rtsp_st, AVStream *st)
204 AVCodecContext *codec = st ? st->codec : NULL;
208 codec->codec_id = handler->codec_id;
209 rtsp_st->dynamic_handler = handler;
211 st->need_parsing = handler->need_parsing;
212 if (handler->priv_data_size) {
213 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
214 if (!rtsp_st->dynamic_protocol_context)
215 rtsp_st->dynamic_handler = NULL;
219 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
222 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
223 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
224 rtsp_st->dynamic_protocol_context);
226 if (rtsp_st->dynamic_protocol_context) {
227 if (rtsp_st->dynamic_handler->close)
228 rtsp_st->dynamic_handler->close(
229 rtsp_st->dynamic_protocol_context);
230 av_free(rtsp_st->dynamic_protocol_context);
232 rtsp_st->dynamic_protocol_context = NULL;
233 rtsp_st->dynamic_handler = NULL;
238 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
239 static int sdp_parse_rtpmap(AVFormatContext *s,
240 AVStream *st, RTSPStream *rtsp_st,
241 int payload_type, const char *p)
243 AVCodecContext *codec = st->codec;
249 /* See if we can handle this kind of payload.
250 * The space should normally not be there but some Real streams or
251 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
252 * have a trailing space. */
253 get_word_sep(buf, sizeof(buf), "/ ", &p);
254 if (payload_type < RTP_PT_PRIVATE) {
255 /* We are in a standard case
256 * (from http://www.iana.org/assignments/rtp-parameters). */
257 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
260 if (codec->codec_id == AV_CODEC_ID_NONE) {
261 RTPDynamicProtocolHandler *handler =
262 ff_rtp_handler_find_by_name(buf, codec->codec_type);
263 init_rtp_handler(handler, rtsp_st, st);
264 /* If no dynamic handler was found, check with the list of standard
265 * allocated types, if such a stream for some reason happens to
266 * use a private payload type. This isn't handled in rtpdec.c, since
267 * the format name from the rtpmap line never is passed into rtpdec. */
268 if (!rtsp_st->dynamic_handler)
269 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
272 c = avcodec_find_decoder(codec->codec_id);
278 get_word_sep(buf, sizeof(buf), "/", &p);
280 switch (codec->codec_type) {
281 case AVMEDIA_TYPE_AUDIO:
282 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
283 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
284 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
286 codec->sample_rate = i;
287 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
288 get_word_sep(buf, sizeof(buf), "/", &p);
293 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
295 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
298 case AVMEDIA_TYPE_VIDEO:
299 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
301 avpriv_set_pts_info(st, 32, 1, i);
306 finalize_rtp_handler_init(s, rtsp_st, st);
310 /* parse the attribute line from the fmtp a line of an sdp response. This
311 * is broken out as a function because it is used in rtp_h264.c, which is
313 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
314 char *value, int value_size)
316 *p += strspn(*p, SPACE_CHARS);
318 get_word_sep(attr, attr_size, "=", p);
321 get_word_sep(value, value_size, ";", p);
329 typedef struct SDPParseState {
331 struct sockaddr_storage default_ip;
333 int skip_media; ///< set if an unknown m= line occurs
334 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
335 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
336 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
337 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
340 char delayed_fmtp[2048];
343 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
344 struct RTSPSource ***dest, int *dest_count)
346 RTSPSource *rtsp_src, *rtsp_src2;
348 for (i = 0; i < count; i++) {
350 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
353 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
354 dynarray_add(dest, dest_count, rtsp_src2);
358 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
359 int payload_type, const char *line)
363 for (i = 0; i < rt->nb_rtsp_streams; i++) {
364 RTSPStream *rtsp_st = rt->rtsp_streams[i];
365 if (rtsp_st->sdp_payload_type == payload_type &&
366 rtsp_st->dynamic_handler &&
367 rtsp_st->dynamic_handler->parse_sdp_a_line) {
368 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
369 rtsp_st->dynamic_protocol_context, line);
374 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
375 int letter, const char *buf)
377 RTSPState *rt = s->priv_data;
378 char buf1[64], st_type[64];
380 enum AVMediaType codec_type;
384 RTSPSource *rtsp_src;
385 struct sockaddr_storage sdp_ip;
388 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
391 if (s1->skip_media && letter != 'm')
395 get_word(buf1, sizeof(buf1), &p);
396 if (strcmp(buf1, "IN") != 0)
398 get_word(buf1, sizeof(buf1), &p);
399 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
401 get_word_sep(buf1, sizeof(buf1), "/", &p);
402 if (get_sockaddr(buf1, &sdp_ip))
407 get_word_sep(buf1, sizeof(buf1), "/", &p);
410 if (s->nb_streams == 0) {
411 s1->default_ip = sdp_ip;
412 s1->default_ttl = ttl;
414 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
415 rtsp_st->sdp_ip = sdp_ip;
416 rtsp_st->sdp_ttl = ttl;
420 av_dict_set(&s->metadata, "title", p, 0);
423 if (s->nb_streams == 0) {
424 av_dict_set(&s->metadata, "comment", p, 0);
433 codec_type = AVMEDIA_TYPE_UNKNOWN;
434 get_word(st_type, sizeof(st_type), &p);
435 if (!strcmp(st_type, "audio")) {
436 codec_type = AVMEDIA_TYPE_AUDIO;
437 } else if (!strcmp(st_type, "video")) {
438 codec_type = AVMEDIA_TYPE_VIDEO;
439 } else if (!strcmp(st_type, "application")) {
440 codec_type = AVMEDIA_TYPE_DATA;
441 } else if (!strcmp(st_type, "text")) {
442 codec_type = AVMEDIA_TYPE_SUBTITLE;
444 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
448 rtsp_st = av_mallocz(sizeof(RTSPStream));
451 rtsp_st->stream_index = -1;
452 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
454 rtsp_st->sdp_ip = s1->default_ip;
455 rtsp_st->sdp_ttl = s1->default_ttl;
457 copy_default_source_addrs(s1->default_include_source_addrs,
458 s1->nb_default_include_source_addrs,
459 &rtsp_st->include_source_addrs,
460 &rtsp_st->nb_include_source_addrs);
461 copy_default_source_addrs(s1->default_exclude_source_addrs,
462 s1->nb_default_exclude_source_addrs,
463 &rtsp_st->exclude_source_addrs,
464 &rtsp_st->nb_exclude_source_addrs);
466 get_word(buf1, sizeof(buf1), &p); /* port */
467 rtsp_st->sdp_port = atoi(buf1);
469 get_word(buf1, sizeof(buf1), &p); /* protocol */
470 if (!strcmp(buf1, "udp"))
471 rt->transport = RTSP_TRANSPORT_RAW;
472 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
473 rtsp_st->feedback = 1;
475 /* XXX: handle list of formats */
476 get_word(buf1, sizeof(buf1), &p); /* format list */
477 rtsp_st->sdp_payload_type = atoi(buf1);
479 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
480 /* no corresponding stream */
481 if (rt->transport == RTSP_TRANSPORT_RAW) {
482 if (CONFIG_RTPDEC && !rt->ts)
483 rt->ts = avpriv_mpegts_parse_open(s);
485 RTPDynamicProtocolHandler *handler;
486 handler = ff_rtp_handler_find_by_id(
487 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
488 init_rtp_handler(handler, rtsp_st, NULL);
489 finalize_rtp_handler_init(s, rtsp_st, NULL);
491 } else if (rt->server_type == RTSP_SERVER_WMS &&
492 codec_type == AVMEDIA_TYPE_DATA) {
493 /* RTX stream, a stream that carries all the other actual
494 * audio/video streams. Don't expose this to the callers. */
496 st = avformat_new_stream(s, NULL);
499 st->id = rt->nb_rtsp_streams - 1;
500 rtsp_st->stream_index = st->index;
501 st->codec->codec_type = codec_type;
502 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
503 RTPDynamicProtocolHandler *handler;
504 /* if standard payload type, we can find the codec right now */
505 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
506 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
507 st->codec->sample_rate > 0)
508 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
509 /* Even static payload types may need a custom depacketizer */
510 handler = ff_rtp_handler_find_by_id(
511 rtsp_st->sdp_payload_type, st->codec->codec_type);
512 init_rtp_handler(handler, rtsp_st, st);
513 finalize_rtp_handler_init(s, rtsp_st, st);
515 if (rt->default_lang[0])
516 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
518 /* put a default control url */
519 av_strlcpy(rtsp_st->control_url, rt->control_uri,
520 sizeof(rtsp_st->control_url));
523 if (av_strstart(p, "control:", &p)) {
524 if (s->nb_streams == 0) {
525 if (!strncmp(p, "rtsp://", 7))
526 av_strlcpy(rt->control_uri, p,
527 sizeof(rt->control_uri));
530 /* get the control url */
531 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
533 /* XXX: may need to add full url resolution */
534 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
536 if (proto[0] == '\0') {
537 /* relative control URL */
538 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
539 av_strlcat(rtsp_st->control_url, "/",
540 sizeof(rtsp_st->control_url));
541 av_strlcat(rtsp_st->control_url, p,
542 sizeof(rtsp_st->control_url));
544 av_strlcpy(rtsp_st->control_url, p,
545 sizeof(rtsp_st->control_url));
547 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
548 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
549 get_word(buf1, sizeof(buf1), &p);
550 payload_type = atoi(buf1);
551 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
552 if (rtsp_st->stream_index >= 0) {
553 st = s->streams[rtsp_st->stream_index];
554 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
558 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
560 } else if (av_strstart(p, "fmtp:", &p) ||
561 av_strstart(p, "framesize:", &p)) {
562 // let dynamic protocol handlers have a stab at the line.
563 get_word(buf1, sizeof(buf1), &p);
564 payload_type = atoi(buf1);
565 if (s1->seen_rtpmap) {
566 parse_fmtp(s, rt, payload_type, buf);
569 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
571 } else if (av_strstart(p, "range:", &p)) {
574 // this is so that seeking on a streamed file can work.
575 rtsp_parse_range_npt(p, &start, &end);
576 s->start_time = start;
577 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
578 s->duration = (end == AV_NOPTS_VALUE) ?
579 AV_NOPTS_VALUE : end - start;
580 } else if (av_strstart(p, "lang:", &p)) {
581 if (s->nb_streams > 0) {
582 get_word(buf1, sizeof(buf1), &p);
583 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
584 if (rtsp_st->stream_index >= 0) {
585 st = s->streams[rtsp_st->stream_index];
586 av_dict_set(&st->metadata, "language", buf1, 0);
589 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
590 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
592 rt->transport = RTSP_TRANSPORT_RDT;
593 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
595 st = s->streams[s->nb_streams - 1];
596 st->codec->sample_rate = atoi(p);
597 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
599 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
600 get_word(buf1, sizeof(buf1), &p); // ignore tag
601 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
602 p += strspn(p, SPACE_CHARS);
603 if (av_strstart(p, "inline:", &p))
604 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
605 } else if (av_strstart(p, "source-filter:", &p)) {
607 get_word(buf1, sizeof(buf1), &p);
608 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
610 exclude = !strcmp(buf1, "excl");
612 get_word(buf1, sizeof(buf1), &p);
613 if (strcmp(buf1, "IN") != 0)
615 get_word(buf1, sizeof(buf1), &p);
616 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
618 // not checking that the destination address actually matches or is wildcard
619 get_word(buf1, sizeof(buf1), &p);
622 rtsp_src = av_mallocz(sizeof(*rtsp_src));
625 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
627 if (s->nb_streams == 0) {
628 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
630 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
631 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
634 if (s->nb_streams == 0) {
635 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
637 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
638 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
643 if (rt->server_type == RTSP_SERVER_WMS)
644 ff_wms_parse_sdp_a_line(s, p);
645 if (s->nb_streams > 0) {
646 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
648 if (rt->server_type == RTSP_SERVER_REAL)
649 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
651 if (rtsp_st->dynamic_handler &&
652 rtsp_st->dynamic_handler->parse_sdp_a_line)
653 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
654 rtsp_st->stream_index,
655 rtsp_st->dynamic_protocol_context, buf);
662 int ff_sdp_parse(AVFormatContext *s, const char *content)
664 RTSPState *rt = s->priv_data;
667 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
668 * contain long SDP lines containing complete ASF Headers (several
669 * kB) or arrays of MDPR (RM stream descriptor) headers plus
670 * "rulebooks" describing their properties. Therefore, the SDP line
673 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
674 * in rtpdec_xiph.c. */
676 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
680 p += strspn(p, SPACE_CHARS);
688 /* get the content */
690 while (*p != '\n' && *p != '\r' && *p != '\0') {
691 if ((q - buf) < sizeof(buf) - 1)
696 sdp_parse_line(s, s1, letter, buf);
698 while (*p != '\n' && *p != '\0')
704 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
705 av_freep(&s1->default_include_source_addrs[i]);
706 av_freep(&s1->default_include_source_addrs);
707 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
708 av_freep(&s1->default_exclude_source_addrs[i]);
709 av_freep(&s1->default_exclude_source_addrs);
711 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
712 if (!rt->p) return AVERROR(ENOMEM);
715 #endif /* CONFIG_RTPDEC */
717 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
719 RTSPState *rt = s->priv_data;
722 for (i = 0; i < rt->nb_rtsp_streams; i++) {
723 RTSPStream *rtsp_st = rt->rtsp_streams[i];
726 if (rtsp_st->transport_priv) {
728 AVFormatContext *rtpctx = rtsp_st->transport_priv;
729 av_write_trailer(rtpctx);
730 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
731 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
732 ff_rtsp_tcp_write_packet(s, rtsp_st);
733 ffio_free_dyn_buf(&rtpctx->pb);
735 avio_closep(&rtpctx->pb);
737 avformat_free_context(rtpctx);
738 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
739 ff_rdt_parse_close(rtsp_st->transport_priv);
740 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
741 ff_rtp_parse_close(rtsp_st->transport_priv);
743 rtsp_st->transport_priv = NULL;
744 if (rtsp_st->rtp_handle)
745 ffurl_close(rtsp_st->rtp_handle);
746 rtsp_st->rtp_handle = NULL;
750 /* close and free RTSP streams */
751 void ff_rtsp_close_streams(AVFormatContext *s)
753 RTSPState *rt = s->priv_data;
757 ff_rtsp_undo_setup(s, 0);
758 for (i = 0; i < rt->nb_rtsp_streams; i++) {
759 rtsp_st = rt->rtsp_streams[i];
761 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
762 if (rtsp_st->dynamic_handler->close)
763 rtsp_st->dynamic_handler->close(
764 rtsp_st->dynamic_protocol_context);
765 av_free(rtsp_st->dynamic_protocol_context);
767 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
768 av_freep(&rtsp_st->include_source_addrs[j]);
769 av_freep(&rtsp_st->include_source_addrs);
770 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
771 av_freep(&rtsp_st->exclude_source_addrs[j]);
772 av_freep(&rtsp_st->exclude_source_addrs);
777 av_freep(&rt->rtsp_streams);
779 avformat_close_input(&rt->asf_ctx);
781 if (CONFIG_RTPDEC && rt->ts)
782 avpriv_mpegts_parse_close(rt->ts);
784 av_freep(&rt->recvbuf);
787 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
789 RTSPState *rt = s->priv_data;
791 int reordering_queue_size = rt->reordering_queue_size;
792 if (reordering_queue_size < 0) {
793 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
794 reordering_queue_size = 0;
796 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
799 /* open the RTP context */
800 if (rtsp_st->stream_index >= 0)
801 st = s->streams[rtsp_st->stream_index];
803 s->ctx_flags |= AVFMTCTX_NOHEADER;
805 if (CONFIG_RTSP_MUXER && s->oformat && st) {
806 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
807 s, st, rtsp_st->rtp_handle,
808 RTSP_TCP_MAX_PACKET_SIZE,
809 rtsp_st->stream_index);
810 /* Ownership of rtp_handle is passed to the rtp mux context */
811 rtsp_st->rtp_handle = NULL;
814 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
815 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
816 return 0; // Don't need to open any parser here
817 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
818 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
819 rtsp_st->dynamic_protocol_context,
820 rtsp_st->dynamic_handler);
821 else if (CONFIG_RTPDEC)
822 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
823 rtsp_st->sdp_payload_type,
824 reordering_queue_size);
826 if (!rtsp_st->transport_priv) {
827 return AVERROR(ENOMEM);
828 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
829 if (rtsp_st->dynamic_handler) {
830 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
831 rtsp_st->dynamic_protocol_context,
832 rtsp_st->dynamic_handler);
834 if (rtsp_st->crypto_suite[0])
835 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
836 rtsp_st->crypto_suite,
837 rtsp_st->crypto_params);
843 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
844 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
851 q += strspn(q, SPACE_CHARS);
852 v = strtol(q, &p, 10);
856 v = strtol(p, &p, 10);
865 /* XXX: only one transport specification is parsed */
866 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
868 char transport_protocol[16];
870 char lower_transport[16];
872 RTSPTransportField *th;
875 reply->nb_transports = 0;
878 p += strspn(p, SPACE_CHARS);
882 th = &reply->transports[reply->nb_transports];
884 get_word_sep(transport_protocol, sizeof(transport_protocol),
886 if (!av_strcasecmp (transport_protocol, "rtp")) {
887 get_word_sep(profile, sizeof(profile), "/;,", &p);
888 lower_transport[0] = '\0';
889 /* rtp/avp/<protocol> */
891 get_word_sep(lower_transport, sizeof(lower_transport),
894 th->transport = RTSP_TRANSPORT_RTP;
895 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
896 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
897 /* x-pn-tng/<protocol> */
898 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
900 th->transport = RTSP_TRANSPORT_RDT;
901 } else if (!av_strcasecmp(transport_protocol, "raw")) {
902 get_word_sep(profile, sizeof(profile), "/;,", &p);
903 lower_transport[0] = '\0';
904 /* raw/raw/<protocol> */
906 get_word_sep(lower_transport, sizeof(lower_transport),
909 th->transport = RTSP_TRANSPORT_RAW;
911 if (!av_strcasecmp(lower_transport, "TCP"))
912 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
914 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
918 /* get each parameter */
919 while (*p != '\0' && *p != ',') {
920 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
921 if (!strcmp(parameter, "port")) {
924 rtsp_parse_range(&th->port_min, &th->port_max, &p);
926 } else if (!strcmp(parameter, "client_port")) {
929 rtsp_parse_range(&th->client_port_min,
930 &th->client_port_max, &p);
932 } else if (!strcmp(parameter, "server_port")) {
935 rtsp_parse_range(&th->server_port_min,
936 &th->server_port_max, &p);
938 } else if (!strcmp(parameter, "interleaved")) {
941 rtsp_parse_range(&th->interleaved_min,
942 &th->interleaved_max, &p);
944 } else if (!strcmp(parameter, "multicast")) {
945 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
946 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
947 } else if (!strcmp(parameter, "ttl")) {
951 th->ttl = strtol(p, &end, 10);
954 } else if (!strcmp(parameter, "destination")) {
957 get_word_sep(buf, sizeof(buf), ";,", &p);
958 get_sockaddr(buf, &th->destination);
960 } else if (!strcmp(parameter, "source")) {
963 get_word_sep(buf, sizeof(buf), ";,", &p);
964 av_strlcpy(th->source, buf, sizeof(th->source));
966 } else if (!strcmp(parameter, "mode")) {
969 get_word_sep(buf, sizeof(buf), ";, ", &p);
970 if (!strcmp(buf, "record") ||
971 !strcmp(buf, "receive"))
976 while (*p != ';' && *p != '\0' && *p != ',')
984 reply->nb_transports++;
988 static void handle_rtp_info(RTSPState *rt, const char *url,
989 uint32_t seq, uint32_t rtptime)
992 if (!rtptime || !url[0])
994 if (rt->transport != RTSP_TRANSPORT_RTP)
996 for (i = 0; i < rt->nb_rtsp_streams; i++) {
997 RTSPStream *rtsp_st = rt->rtsp_streams[i];
998 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1001 if (!strcmp(rtsp_st->control_url, url)) {
1002 rtpctx->base_timestamp = rtptime;
1008 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1011 char key[20], value[1024], url[1024] = "";
1012 uint32_t seq = 0, rtptime = 0;
1015 p += strspn(p, SPACE_CHARS);
1018 get_word_sep(key, sizeof(key), "=", &p);
1022 get_word_sep(value, sizeof(value), ";, ", &p);
1024 if (!strcmp(key, "url"))
1025 av_strlcpy(url, value, sizeof(url));
1026 else if (!strcmp(key, "seq"))
1027 seq = strtoul(value, NULL, 10);
1028 else if (!strcmp(key, "rtptime"))
1029 rtptime = strtoul(value, NULL, 10);
1031 handle_rtp_info(rt, url, seq, rtptime);
1040 handle_rtp_info(rt, url, seq, rtptime);
1043 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1044 RTSPState *rt, const char *method)
1048 /* NOTE: we do case independent match for broken servers */
1050 if (av_stristart(p, "Session:", &p)) {
1052 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1053 if (av_stristart(p, ";timeout=", &p) &&
1054 (t = strtol(p, NULL, 10)) > 0) {
1057 } else if (av_stristart(p, "Content-Length:", &p)) {
1058 reply->content_length = strtol(p, NULL, 10);
1059 } else if (av_stristart(p, "Transport:", &p)) {
1060 rtsp_parse_transport(reply, p);
1061 } else if (av_stristart(p, "CSeq:", &p)) {
1062 reply->seq = strtol(p, NULL, 10);
1063 } else if (av_stristart(p, "Range:", &p)) {
1064 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1065 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1066 p += strspn(p, SPACE_CHARS);
1067 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1068 } else if (av_stristart(p, "Server:", &p)) {
1069 p += strspn(p, SPACE_CHARS);
1070 av_strlcpy(reply->server, p, sizeof(reply->server));
1071 } else if (av_stristart(p, "Notice:", &p) ||
1072 av_stristart(p, "X-Notice:", &p)) {
1073 reply->notice = strtol(p, NULL, 10);
1074 } else if (av_stristart(p, "Location:", &p)) {
1075 p += strspn(p, SPACE_CHARS);
1076 av_strlcpy(reply->location, p , sizeof(reply->location));
1077 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1078 p += strspn(p, SPACE_CHARS);
1079 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1080 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1081 p += strspn(p, SPACE_CHARS);
1082 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1083 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1084 p += strspn(p, SPACE_CHARS);
1085 if (method && !strcmp(method, "DESCRIBE"))
1086 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1087 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1088 p += strspn(p, SPACE_CHARS);
1089 if (method && !strcmp(method, "PLAY"))
1090 rtsp_parse_rtp_info(rt, p);
1091 } else if (av_stristart(p, "Public:", &p) && rt) {
1092 if (strstr(p, "GET_PARAMETER") &&
1093 method && !strcmp(method, "OPTIONS"))
1094 rt->get_parameter_supported = 1;
1095 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1096 p += strspn(p, SPACE_CHARS);
1097 rt->accept_dynamic_rate = atoi(p);
1098 } else if (av_stristart(p, "Content-Type:", &p)) {
1099 p += strspn(p, SPACE_CHARS);
1100 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1104 /* skip a RTP/TCP interleaved packet */
1105 void ff_rtsp_skip_packet(AVFormatContext *s)
1107 RTSPState *rt = s->priv_data;
1111 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1114 len = AV_RB16(buf + 1);
1116 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1121 if (len1 > sizeof(buf))
1123 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1130 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1131 unsigned char **content_ptr,
1132 int return_on_interleaved_data, const char *method)
1134 RTSPState *rt = s->priv_data;
1135 char buf[4096], buf1[1024], *q;
1138 int ret, content_length, line_count = 0, request = 0;
1139 unsigned char *content = NULL;
1145 memset(reply, 0, sizeof(*reply));
1147 /* parse reply (XXX: use buffers) */
1148 rt->last_reply[0] = '\0';
1152 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1153 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1159 /* XXX: only parse it if first char on line ? */
1160 if (return_on_interleaved_data) {
1163 ff_rtsp_skip_packet(s);
1164 } else if (ch != '\r') {
1165 if ((q - buf) < sizeof(buf) - 1)
1171 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1173 /* test if last line */
1177 if (line_count == 0) {
1178 /* get reply code */
1179 get_word(buf1, sizeof(buf1), &p);
1180 if (!strncmp(buf1, "RTSP/", 5)) {
1181 get_word(buf1, sizeof(buf1), &p);
1182 reply->status_code = atoi(buf1);
1183 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1185 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1186 get_word(buf1, sizeof(buf1), &p); // object
1190 ff_rtsp_parse_line(reply, p, rt, method);
1191 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1192 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1197 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1198 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1200 content_length = reply->content_length;
1201 if (content_length > 0) {
1202 /* leave some room for a trailing '\0' (useful for simple parsing) */
1203 content = av_malloc(content_length + 1);
1205 return AVERROR(ENOMEM);
1206 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1207 content[content_length] = '\0';
1210 *content_ptr = content;
1216 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1217 const char* ptr = buf;
1219 if (!strcmp(reply->reason, "OPTIONS")) {
1220 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1222 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1223 if (reply->session_id[0])
1224 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1227 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1229 av_strlcat(buf, "\r\n", sizeof(buf));
1231 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1232 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1235 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1237 rt->last_cmd_time = av_gettime_relative();
1238 /* Even if the request from the server had data, it is not the data
1239 * that the caller wants or expects. The memory could also be leaked
1240 * if the actual following reply has content data. */
1242 av_freep(content_ptr);
1243 /* If method is set, this is called from ff_rtsp_send_cmd,
1244 * where a reply to exactly this request is awaited. For
1245 * callers from within packet receiving, we just want to
1246 * return to the caller and go back to receiving packets. */
1252 if (rt->seq != reply->seq) {
1253 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1254 rt->seq, reply->seq);
1258 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1259 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1260 reply->notice == 2306 /* Continuous Feed Terminated */) {
1261 rt->state = RTSP_STATE_IDLE;
1262 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1263 return AVERROR(EIO); /* data or server error */
1264 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1265 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1266 return AVERROR(EPERM);
1272 * Send a command to the RTSP server without waiting for the reply.
1274 * @param s RTSP (de)muxer context
1275 * @param method the method for the request
1276 * @param url the target url for the request
1277 * @param headers extra header lines to include in the request
1278 * @param send_content if non-null, the data to send as request body content
1279 * @param send_content_length the length of the send_content data, or 0 if
1280 * send_content is null
1282 * @return zero if success, nonzero otherwise
1284 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1285 const char *method, const char *url,
1286 const char *headers,
1287 const unsigned char *send_content,
1288 int send_content_length)
1290 RTSPState *rt = s->priv_data;
1291 char buf[4096], *out_buf;
1292 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1294 /* Add in RTSP headers */
1297 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1299 av_strlcat(buf, headers, sizeof(buf));
1300 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1301 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1302 if (rt->session_id[0] != '\0' && (!headers ||
1303 !strstr(headers, "\nIf-Match:"))) {
1304 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1307 char *str = ff_http_auth_create_response(&rt->auth_state,
1308 rt->auth, url, method);
1310 av_strlcat(buf, str, sizeof(buf));
1313 if (send_content_length > 0 && send_content)
1314 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1315 av_strlcat(buf, "\r\n", sizeof(buf));
1317 /* base64 encode rtsp if tunneling */
1318 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1319 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1320 out_buf = base64buf;
1323 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1325 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1326 if (send_content_length > 0 && send_content) {
1327 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1328 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1329 "with content data not supported\n");
1330 return AVERROR_PATCHWELCOME;
1332 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1334 rt->last_cmd_time = av_gettime_relative();
1339 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1340 const char *url, const char *headers)
1342 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1345 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1346 const char *headers, RTSPMessageHeader *reply,
1347 unsigned char **content_ptr)
1349 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1350 content_ptr, NULL, 0);
1353 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1354 const char *method, const char *url,
1356 RTSPMessageHeader *reply,
1357 unsigned char **content_ptr,
1358 const unsigned char *send_content,
1359 int send_content_length)
1361 RTSPState *rt = s->priv_data;
1362 HTTPAuthType cur_auth_type;
1363 int ret, attempts = 0;
1366 cur_auth_type = rt->auth_state.auth_type;
1367 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1369 send_content_length)))
1372 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1376 if (reply->status_code == 401 &&
1377 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1378 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1381 if (reply->status_code > 400){
1382 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1386 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1392 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1393 int lower_transport, const char *real_challenge)
1395 RTSPState *rt = s->priv_data;
1396 int rtx = 0, j, i, err, interleave = 0, port_off;
1397 RTSPStream *rtsp_st;
1398 RTSPMessageHeader reply1, *reply = &reply1;
1400 const char *trans_pref;
1402 if (rt->transport == RTSP_TRANSPORT_RDT)
1403 trans_pref = "x-pn-tng";
1404 else if (rt->transport == RTSP_TRANSPORT_RAW)
1405 trans_pref = "RAW/RAW";
1407 trans_pref = "RTP/AVP";
1409 /* default timeout: 1 minute */
1412 /* Choose a random starting offset within the first half of the
1413 * port range, to allow for a number of ports to try even if the offset
1414 * happens to be at the end of the random range. */
1415 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1416 /* even random offset */
1417 port_off -= port_off & 0x01;
1419 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1420 char transport[2048];
1423 * WMS serves all UDP data over a single connection, the RTX, which
1424 * isn't necessarily the first in the SDP but has to be the first
1425 * to be set up, else the second/third SETUP will fail with a 461.
1427 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1428 rt->server_type == RTSP_SERVER_WMS) {
1431 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1432 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1434 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1438 if (rtx == rt->nb_rtsp_streams)
1439 return -1; /* no RTX found */
1440 rtsp_st = rt->rtsp_streams[rtx];
1442 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1444 rtsp_st = rt->rtsp_streams[i];
1447 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1450 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1451 port = reply->transports[0].client_port_min;
1455 /* first try in specified port range */
1456 while (j <= rt->rtp_port_max) {
1457 AVDictionary *opts = map_to_opts(rt);
1459 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1460 "?localport=%d", j);
1461 /* we will use two ports per rtp stream (rtp and rtcp) */
1463 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1464 &s->interrupt_callback, &opts);
1466 av_dict_free(&opts);
1471 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1476 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1478 snprintf(transport, sizeof(transport) - 1,
1479 "%s/UDP;", trans_pref);
1480 if (rt->server_type != RTSP_SERVER_REAL)
1481 av_strlcat(transport, "unicast;", sizeof(transport));
1482 av_strlcatf(transport, sizeof(transport),
1483 "client_port=%d", port);
1484 if (rt->transport == RTSP_TRANSPORT_RTP &&
1485 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1486 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1490 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1491 /* For WMS streams, the application streams are only used for
1492 * UDP. When trying to set it up for TCP streams, the server
1493 * will return an error. Therefore, we skip those streams. */
1494 if (rt->server_type == RTSP_SERVER_WMS &&
1495 (rtsp_st->stream_index < 0 ||
1496 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1499 snprintf(transport, sizeof(transport) - 1,
1500 "%s/TCP;", trans_pref);
1501 if (rt->transport != RTSP_TRANSPORT_RDT)
1502 av_strlcat(transport, "unicast;", sizeof(transport));
1503 av_strlcatf(transport, sizeof(transport),
1504 "interleaved=%d-%d",
1505 interleave, interleave + 1);
1509 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1510 snprintf(transport, sizeof(transport) - 1,
1511 "%s/UDP;multicast", trans_pref);
1514 av_strlcat(transport, ";mode=record", sizeof(transport));
1515 } else if (rt->server_type == RTSP_SERVER_REAL ||
1516 rt->server_type == RTSP_SERVER_WMS)
1517 av_strlcat(transport, ";mode=play", sizeof(transport));
1518 snprintf(cmd, sizeof(cmd),
1519 "Transport: %s\r\n",
1521 if (rt->accept_dynamic_rate)
1522 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1523 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1524 char real_res[41], real_csum[9];
1525 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1527 av_strlcatf(cmd, sizeof(cmd),
1529 "RealChallenge2: %s, sd=%s\r\n",
1530 rt->session_id, real_res, real_csum);
1532 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1533 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1536 } else if (reply->status_code != RTSP_STATUS_OK ||
1537 reply->nb_transports != 1) {
1538 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1542 /* XXX: same protocol for all streams is required */
1544 if (reply->transports[0].lower_transport != rt->lower_transport ||
1545 reply->transports[0].transport != rt->transport) {
1546 err = AVERROR_INVALIDDATA;
1550 rt->lower_transport = reply->transports[0].lower_transport;
1551 rt->transport = reply->transports[0].transport;
1554 /* Fail if the server responded with another lower transport mode
1555 * than what we requested. */
1556 if (reply->transports[0].lower_transport != lower_transport) {
1557 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1558 err = AVERROR_INVALIDDATA;
1562 switch(reply->transports[0].lower_transport) {
1563 case RTSP_LOWER_TRANSPORT_TCP:
1564 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1565 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1568 case RTSP_LOWER_TRANSPORT_UDP: {
1569 char url[1024], options[30] = "";
1570 const char *peer = host;
1572 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1573 av_strlcpy(options, "?connect=1", sizeof(options));
1574 /* Use source address if specified */
1575 if (reply->transports[0].source[0])
1576 peer = reply->transports[0].source;
1577 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1578 reply->transports[0].server_port_min, "%s", options);
1579 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1580 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1581 err = AVERROR_INVALIDDATA;
1586 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1587 char url[1024], namebuf[50], optbuf[20] = "";
1588 struct sockaddr_storage addr;
1591 if (reply->transports[0].destination.ss_family) {
1592 addr = reply->transports[0].destination;
1593 port = reply->transports[0].port_min;
1594 ttl = reply->transports[0].ttl;
1596 addr = rtsp_st->sdp_ip;
1597 port = rtsp_st->sdp_port;
1598 ttl = rtsp_st->sdp_ttl;
1601 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1602 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1603 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1604 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1605 port, "%s", optbuf);
1606 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1607 &s->interrupt_callback, NULL) < 0) {
1608 err = AVERROR_INVALIDDATA;
1615 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1619 if (rt->nb_rtsp_streams && reply->timeout > 0)
1620 rt->timeout = reply->timeout;
1622 if (rt->server_type == RTSP_SERVER_REAL)
1623 rt->need_subscription = 1;
1628 ff_rtsp_undo_setup(s, 0);
1632 void ff_rtsp_close_connections(AVFormatContext *s)
1634 RTSPState *rt = s->priv_data;
1635 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1636 ffurl_close(rt->rtsp_hd);
1637 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1640 int ff_rtsp_connect(AVFormatContext *s)
1642 RTSPState *rt = s->priv_data;
1643 char proto[128], host[1024], path[1024];
1644 char tcpname[1024], cmd[2048], auth[128];
1645 const char *lower_rtsp_proto = "tcp";
1646 int port, err, tcp_fd;
1647 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1648 int lower_transport_mask = 0;
1649 int default_port = RTSP_DEFAULT_PORT;
1650 char real_challenge[64] = "";
1651 struct sockaddr_storage peer;
1652 socklen_t peer_len = sizeof(peer);
1654 if (rt->rtp_port_max < rt->rtp_port_min) {
1655 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1656 "than min port %d\n", rt->rtp_port_max,
1658 return AVERROR(EINVAL);
1661 if (!ff_network_init())
1662 return AVERROR(EIO);
1664 if (s->max_delay < 0) /* Not set by the caller */
1665 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1667 rt->control_transport = RTSP_MODE_PLAIN;
1668 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1669 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1670 rt->control_transport = RTSP_MODE_TUNNEL;
1672 /* Only pass through valid flags from here */
1673 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1676 /* extract hostname and port */
1677 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1678 host, sizeof(host), &port, path, sizeof(path), s->filename);
1680 if (!strcmp(proto, "rtsps")) {
1681 lower_rtsp_proto = "tls";
1682 default_port = RTSPS_DEFAULT_PORT;
1683 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1687 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1690 port = default_port;
1692 lower_transport_mask = rt->lower_transport_mask;
1694 if (!lower_transport_mask)
1695 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1698 /* Only UDP or TCP - UDP multicast isn't supported. */
1699 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1700 (1 << RTSP_LOWER_TRANSPORT_TCP);
1701 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1702 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1703 "only UDP and TCP are supported for output.\n");
1704 err = AVERROR(EINVAL);
1709 /* Construct the URI used in request; this is similar to s->filename,
1710 * but with authentication credentials removed and RTSP specific options
1712 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1713 host, port, "%s", path);
1715 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1716 /* set up initial handshake for tunneling */
1717 char httpname[1024];
1718 char sessioncookie[17];
1721 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1722 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1723 av_get_random_seed(), av_get_random_seed());
1726 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1727 &s->interrupt_callback) < 0) {
1732 /* generate GET headers */
1733 snprintf(headers, sizeof(headers),
1734 "x-sessioncookie: %s\r\n"
1735 "Accept: application/x-rtsp-tunnelled\r\n"
1736 "Pragma: no-cache\r\n"
1737 "Cache-Control: no-cache\r\n",
1739 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1741 /* complete the connection */
1742 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1748 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1749 &s->interrupt_callback) < 0 ) {
1754 /* generate POST headers */
1755 snprintf(headers, sizeof(headers),
1756 "x-sessioncookie: %s\r\n"
1757 "Content-Type: application/x-rtsp-tunnelled\r\n"
1758 "Pragma: no-cache\r\n"
1759 "Cache-Control: no-cache\r\n"
1760 "Content-Length: 32767\r\n"
1761 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1763 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1764 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1766 /* Initialize the authentication state for the POST session. The HTTP
1767 * protocol implementation doesn't properly handle multi-pass
1768 * authentication for POST requests, since it would require one of
1770 * - implementing Expect: 100-continue, which many HTTP servers
1771 * don't support anyway, even less the RTSP servers that do HTTP
1773 * - sending the whole POST data until getting a 401 reply specifying
1774 * what authentication method to use, then resending all that data
1775 * - waiting for potential 401 replies directly after sending the
1776 * POST header (waiting for some unspecified time)
1777 * Therefore, we copy the full auth state, which works for both basic
1778 * and digest. (For digest, we would have to synchronize the nonce
1779 * count variable between the two sessions, if we'd do more requests
1780 * with the original session, though.)
1782 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1784 /* complete the connection */
1785 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1791 /* open the tcp connection */
1792 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1794 "?timeout=%d", rt->stimeout);
1795 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1796 &s->interrupt_callback, NULL)) < 0) {
1800 rt->rtsp_hd_out = rt->rtsp_hd;
1804 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1809 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1810 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1811 NULL, 0, NI_NUMERICHOST);
1814 /* request options supported by the server; this also detects server
1816 for (rt->server_type = RTSP_SERVER_RTP;;) {
1818 if (rt->server_type == RTSP_SERVER_REAL)
1821 * The following entries are required for proper
1822 * streaming from a Realmedia server. They are
1823 * interdependent in some way although we currently
1824 * don't quite understand how. Values were copied
1825 * from mplayer SVN r23589.
1826 * ClientChallenge is a 16-byte ID in hex
1827 * CompanyID is a 16-byte ID in base64
1829 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1830 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1831 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1832 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1834 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1835 if (reply->status_code != RTSP_STATUS_OK) {
1836 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1840 /* detect server type if not standard-compliant RTP */
1841 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1842 rt->server_type = RTSP_SERVER_REAL;
1844 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1845 rt->server_type = RTSP_SERVER_WMS;
1846 } else if (rt->server_type == RTSP_SERVER_REAL)
1847 strcpy(real_challenge, reply->real_challenge);
1851 if (CONFIG_RTSP_DEMUXER && s->iformat)
1852 err = ff_rtsp_setup_input_streams(s, reply);
1853 else if (CONFIG_RTSP_MUXER)
1854 err = ff_rtsp_setup_output_streams(s, host);
1861 int lower_transport = ff_log2_tab[lower_transport_mask &
1862 ~(lower_transport_mask - 1)];
1864 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1865 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1866 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1868 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1869 rt->server_type == RTSP_SERVER_REAL ?
1870 real_challenge : NULL);
1873 lower_transport_mask &= ~(1 << lower_transport);
1874 if (lower_transport_mask == 0 && err == 1) {
1875 err = AVERROR(EPROTONOSUPPORT);
1880 rt->lower_transport_mask = lower_transport_mask;
1881 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1882 rt->state = RTSP_STATE_IDLE;
1883 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1886 ff_rtsp_close_streams(s);
1887 ff_rtsp_close_connections(s);
1888 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1889 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1890 rt->session_id[0] = '\0';
1891 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1899 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1902 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1903 uint8_t *buf, int buf_size, int64_t wait_end)
1905 RTSPState *rt = s->priv_data;
1906 RTSPStream *rtsp_st;
1907 int n, i, ret, tcp_fd, timeout_cnt = 0;
1909 struct pollfd *p = rt->p;
1910 int *fds = NULL, fdsnum, fdsidx;
1913 if (ff_check_interrupt(&s->interrupt_callback))
1914 return AVERROR_EXIT;
1915 if (wait_end && wait_end - av_gettime_relative() < 0)
1916 return AVERROR(EAGAIN);
1919 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1920 p[max_p].fd = tcp_fd;
1921 p[max_p++].events = POLLIN;
1925 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1926 rtsp_st = rt->rtsp_streams[i];
1927 if (rtsp_st->rtp_handle) {
1928 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1930 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1934 av_log(s, AV_LOG_ERROR,
1935 "Number of fds %d not supported\n", fdsnum);
1936 return AVERROR_INVALIDDATA;
1938 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1939 p[max_p].fd = fds[fdsidx];
1940 p[max_p++].events = POLLIN;
1945 n = poll(p, max_p, POLL_TIMEOUT_MS);
1947 int j = 1 - (tcp_fd == -1);
1949 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1950 rtsp_st = rt->rtsp_streams[i];
1951 if (rtsp_st->rtp_handle) {
1952 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1953 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1955 *prtsp_st = rtsp_st;
1962 #if CONFIG_RTSP_DEMUXER
1963 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1964 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1965 if (rt->state == RTSP_STATE_STREAMING) {
1966 if (!ff_rtsp_parse_streaming_commands(s))
1969 av_log(s, AV_LOG_WARNING,
1970 "Unable to answer to TEARDOWN\n");
1974 RTSPMessageHeader reply;
1975 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1978 /* XXX: parse message */
1979 if (rt->state != RTSP_STATE_STREAMING)
1984 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1985 return AVERROR(ETIMEDOUT);
1986 } else if (n < 0 && errno != EINTR)
1987 return AVERROR(errno);
1991 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1992 const uint8_t *buf, int len)
1994 RTSPState *rt = s->priv_data;
1998 if (rt->nb_rtsp_streams == 1) {
1999 *rtsp_st = rt->rtsp_streams[0];
2002 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2003 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2005 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2006 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2009 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2010 *rtsp_st = rt->rtsp_streams[i];
2017 av_log(s, AV_LOG_WARNING,
2018 "Unable to pick stream for packet - SSRC not known for "
2020 return AVERROR(EAGAIN);
2023 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2024 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2025 *rtsp_st = rt->rtsp_streams[i];
2031 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2032 return AVERROR(EAGAIN);
2035 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2037 RTSPState *rt = s->priv_data;
2039 RTSPStream *rtsp_st, *first_queue_st = NULL;
2040 int64_t wait_end = 0;
2042 if (rt->nb_byes == rt->nb_rtsp_streams)
2045 /* get next frames from the same RTP packet */
2046 if (rt->cur_transport_priv) {
2047 if (rt->transport == RTSP_TRANSPORT_RDT) {
2048 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2049 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2050 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2051 } else if (CONFIG_RTPDEC && rt->ts) {
2052 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2054 rt->recvbuf_pos += ret;
2055 ret = rt->recvbuf_pos < rt->recvbuf_len;
2060 rt->cur_transport_priv = NULL;
2062 } else if (ret == 1) {
2065 rt->cur_transport_priv = NULL;
2069 if (rt->transport == RTSP_TRANSPORT_RTP) {
2071 int64_t first_queue_time = 0;
2072 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2073 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2077 queue_time = ff_rtp_queued_packet_time(rtpctx);
2078 if (queue_time && (queue_time - first_queue_time < 0 ||
2079 !first_queue_time)) {
2080 first_queue_time = queue_time;
2081 first_queue_st = rt->rtsp_streams[i];
2084 if (first_queue_time) {
2085 wait_end = first_queue_time + s->max_delay;
2088 first_queue_st = NULL;
2092 /* read next RTP packet */
2094 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2096 return AVERROR(ENOMEM);
2099 switch(rt->lower_transport) {
2101 #if CONFIG_RTSP_DEMUXER
2102 case RTSP_LOWER_TRANSPORT_TCP:
2103 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2106 case RTSP_LOWER_TRANSPORT_UDP:
2107 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2108 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2109 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2110 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2112 case RTSP_LOWER_TRANSPORT_CUSTOM:
2113 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2114 wait_end && wait_end < av_gettime_relative())
2115 len = AVERROR(EAGAIN);
2117 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2118 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2119 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2120 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2123 if (len == AVERROR(EAGAIN) && first_queue_st &&
2124 rt->transport == RTSP_TRANSPORT_RTP) {
2125 rtsp_st = first_queue_st;
2126 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2133 if (rt->transport == RTSP_TRANSPORT_RDT) {
2134 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2135 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2136 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2137 if (rtsp_st->feedback) {
2138 AVIOContext *pb = NULL;
2139 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2141 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2144 /* Either bad packet, or a RTCP packet. Check if the
2145 * first_rtcp_ntp_time field was initialized. */
2146 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2147 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2148 /* first_rtcp_ntp_time has been initialized for this stream,
2149 * copy the same value to all other uninitialized streams,
2150 * in order to map their timestamp origin to the same ntp time
2153 AVStream *st = NULL;
2154 if (rtsp_st->stream_index >= 0)
2155 st = s->streams[rtsp_st->stream_index];
2156 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2157 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2158 AVStream *st2 = NULL;
2159 if (rt->rtsp_streams[i]->stream_index >= 0)
2160 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2161 if (rtpctx2 && st && st2 &&
2162 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2163 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2164 rtpctx2->rtcp_ts_offset = av_rescale_q(
2165 rtpctx->rtcp_ts_offset, st->time_base,
2169 // Make real NTP start time available in AVFormatContext
2170 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2171 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2173 s->start_time_realtime -=
2174 av_rescale (rtpctx->rtcp_ts_offset,
2175 (uint64_t) rtpctx->st->time_base.num * 1000000,
2176 rtpctx->st->time_base.den);
2180 if (ret == -RTCP_BYE) {
2183 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2184 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2186 if (rt->nb_byes == rt->nb_rtsp_streams)
2190 } else if (CONFIG_RTPDEC && rt->ts) {
2191 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2194 rt->recvbuf_len = len;
2195 rt->recvbuf_pos = ret;
2196 rt->cur_transport_priv = rt->ts;
2203 return AVERROR_INVALIDDATA;
2209 /* more packets may follow, so we save the RTP context */
2210 rt->cur_transport_priv = rtsp_st->transport_priv;
2214 #endif /* CONFIG_RTPDEC */
2216 #if CONFIG_SDP_DEMUXER
2217 static int sdp_probe(AVProbeData *p1)
2219 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2221 /* we look for a line beginning "c=IN IP" */
2222 while (p < p_end && *p != '\0') {
2223 if (sizeof("c=IN IP") - 1 < p_end - p &&
2224 av_strstart(p, "c=IN IP", NULL))
2225 return AVPROBE_SCORE_EXTENSION;
2227 while (p < p_end - 1 && *p != '\n') p++;
2236 static void append_source_addrs(char *buf, int size, const char *name,
2237 int count, struct RTSPSource **addrs)
2242 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2243 for (i = 1; i < count; i++)
2244 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2247 static int sdp_read_header(AVFormatContext *s)
2249 RTSPState *rt = s->priv_data;
2250 RTSPStream *rtsp_st;
2255 if (!ff_network_init())
2256 return AVERROR(EIO);
2258 if (s->max_delay < 0) /* Not set by the caller */
2259 s->max_delay = DEFAULT_REORDERING_DELAY;
2260 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2261 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2263 /* read the whole sdp file */
2264 /* XXX: better loading */
2265 content = av_malloc(SDP_MAX_SIZE);
2267 return AVERROR(ENOMEM);
2268 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2271 return AVERROR_INVALIDDATA;
2273 content[size] ='\0';
2275 err = ff_sdp_parse(s, content);
2279 /* open each RTP stream */
2280 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2282 rtsp_st = rt->rtsp_streams[i];
2284 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2285 AVDictionary *opts = map_to_opts(rt);
2287 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2288 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2289 ff_url_join(url, sizeof(url), "rtp", NULL,
2290 namebuf, rtsp_st->sdp_port,
2291 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2292 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2293 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2294 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2296 append_source_addrs(url, sizeof(url), "sources",
2297 rtsp_st->nb_include_source_addrs,
2298 rtsp_st->include_source_addrs);
2299 append_source_addrs(url, sizeof(url), "block",
2300 rtsp_st->nb_exclude_source_addrs,
2301 rtsp_st->exclude_source_addrs);
2302 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2303 &s->interrupt_callback, &opts);
2305 av_dict_free(&opts);
2308 err = AVERROR_INVALIDDATA;
2312 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2317 ff_rtsp_close_streams(s);
2322 static int sdp_read_close(AVFormatContext *s)
2324 ff_rtsp_close_streams(s);
2329 static const AVClass sdp_demuxer_class = {
2330 .class_name = "SDP demuxer",
2331 .item_name = av_default_item_name,
2332 .option = sdp_options,
2333 .version = LIBAVUTIL_VERSION_INT,
2336 AVInputFormat ff_sdp_demuxer = {
2338 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2339 .priv_data_size = sizeof(RTSPState),
2340 .read_probe = sdp_probe,
2341 .read_header = sdp_read_header,
2342 .read_packet = ff_rtsp_fetch_packet,
2343 .read_close = sdp_read_close,
2344 .priv_class = &sdp_demuxer_class,
2346 #endif /* CONFIG_SDP_DEMUXER */
2348 #if CONFIG_RTP_DEMUXER
2349 static int rtp_probe(AVProbeData *p)
2351 if (av_strstart(p->filename, "rtp:", NULL))
2352 return AVPROBE_SCORE_MAX;
2356 static int rtp_read_header(AVFormatContext *s)
2358 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2359 char host[500], sdp[500];
2361 URLContext* in = NULL;
2363 AVCodecContext codec = { 0 };
2364 struct sockaddr_storage addr;
2366 socklen_t addrlen = sizeof(addr);
2367 RTSPState *rt = s->priv_data;
2369 if (!ff_network_init())
2370 return AVERROR(EIO);
2372 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2373 &s->interrupt_callback, NULL);
2378 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2379 if (ret == AVERROR(EAGAIN))
2384 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2388 if ((recvbuf[0] & 0xc0) != 0x80) {
2389 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2394 if (RTP_PT_IS_RTCP(recvbuf[1]))
2397 payload_type = recvbuf[1] & 0x7f;
2400 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2404 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2405 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2406 "without an SDP file describing it\n",
2410 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2411 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2412 "properly you need an SDP file "
2416 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2417 NULL, 0, s->filename);
2419 snprintf(sdp, sizeof(sdp),
2420 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2421 addr.ss_family == AF_INET ? 4 : 6, host,
2422 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2423 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2424 port, payload_type);
2425 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2427 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2430 /* sdp_read_header initializes this again */
2433 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2435 ret = sdp_read_header(s);
2446 static const AVClass rtp_demuxer_class = {
2447 .class_name = "RTP demuxer",
2448 .item_name = av_default_item_name,
2449 .option = rtp_options,
2450 .version = LIBAVUTIL_VERSION_INT,
2453 AVInputFormat ff_rtp_demuxer = {
2455 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2456 .priv_data_size = sizeof(RTSPState),
2457 .read_probe = rtp_probe,
2458 .read_header = rtp_read_header,
2459 .read_packet = ff_rtsp_fetch_packet,
2460 .read_close = sdp_read_close,
2461 .flags = AVFMT_NOFILE,
2462 .priv_class = &rtp_demuxer_class,
2464 #endif /* CONFIG_RTP_DEMUXER */