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_BOOL, {.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)
181 get_word_sep(buf, sizeof(buf), "-", &p);
182 if (av_parse_time(end, buf, 1) < 0)
183 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
187 static int get_sockaddr(AVFormatContext *s,
188 const char *buf, struct sockaddr_storage *sock)
190 struct addrinfo hints = { 0 }, *ai = NULL;
193 hints.ai_flags = AI_NUMERICHOST;
194 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
195 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
200 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
206 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
207 RTSPStream *rtsp_st, AVStream *st)
209 AVCodecParameters *par = st ? st->codecpar : NULL;
213 par->codec_id = handler->codec_id;
214 rtsp_st->dynamic_handler = handler;
216 st->need_parsing = handler->need_parsing;
217 if (handler->priv_data_size) {
218 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
219 if (!rtsp_st->dynamic_protocol_context)
220 rtsp_st->dynamic_handler = NULL;
224 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
227 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
228 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
229 rtsp_st->dynamic_protocol_context);
231 if (rtsp_st->dynamic_protocol_context) {
232 if (rtsp_st->dynamic_handler->close)
233 rtsp_st->dynamic_handler->close(
234 rtsp_st->dynamic_protocol_context);
235 av_free(rtsp_st->dynamic_protocol_context);
237 rtsp_st->dynamic_protocol_context = NULL;
238 rtsp_st->dynamic_handler = NULL;
243 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
244 static int sdp_parse_rtpmap(AVFormatContext *s,
245 AVStream *st, RTSPStream *rtsp_st,
246 int payload_type, const char *p)
248 AVCodecParameters *par = st->codecpar;
251 const AVCodecDescriptor *desc;
254 /* See if we can handle this kind of payload.
255 * The space should normally not be there but some Real streams or
256 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
257 * have a trailing space. */
258 get_word_sep(buf, sizeof(buf), "/ ", &p);
259 if (payload_type < RTP_PT_PRIVATE) {
260 /* We are in a standard case
261 * (from http://www.iana.org/assignments/rtp-parameters). */
262 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
265 if (par->codec_id == AV_CODEC_ID_NONE) {
266 RTPDynamicProtocolHandler *handler =
267 ff_rtp_handler_find_by_name(buf, par->codec_type);
268 init_rtp_handler(handler, rtsp_st, st);
269 /* If no dynamic handler was found, check with the list of standard
270 * allocated types, if such a stream for some reason happens to
271 * use a private payload type. This isn't handled in rtpdec.c, since
272 * the format name from the rtpmap line never is passed into rtpdec. */
273 if (!rtsp_st->dynamic_handler)
274 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
277 desc = avcodec_descriptor_get(par->codec_id);
278 if (desc && desc->name)
283 get_word_sep(buf, sizeof(buf), "/", &p);
285 switch (par->codec_type) {
286 case AVMEDIA_TYPE_AUDIO:
287 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
288 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
289 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
291 par->sample_rate = i;
292 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
293 get_word_sep(buf, sizeof(buf), "/", &p);
298 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
300 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
303 case AVMEDIA_TYPE_VIDEO:
304 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
306 avpriv_set_pts_info(st, 32, 1, i);
311 finalize_rtp_handler_init(s, rtsp_st, st);
315 /* parse the attribute line from the fmtp a line of an sdp response. This
316 * is broken out as a function because it is used in rtp_h264.c, which is
318 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
319 char *value, int value_size)
321 *p += strspn(*p, SPACE_CHARS);
323 get_word_sep(attr, attr_size, "=", p);
326 get_word_sep(value, value_size, ";", p);
334 typedef struct SDPParseState {
336 struct sockaddr_storage default_ip;
338 int skip_media; ///< set if an unknown m= line occurs
339 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
340 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
341 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
342 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
345 char delayed_fmtp[2048];
348 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
349 struct RTSPSource ***dest, int *dest_count)
351 RTSPSource *rtsp_src, *rtsp_src2;
353 for (i = 0; i < count; i++) {
355 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
358 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
359 dynarray_add(dest, dest_count, rtsp_src2);
363 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
364 int payload_type, const char *line)
368 for (i = 0; i < rt->nb_rtsp_streams; i++) {
369 RTSPStream *rtsp_st = rt->rtsp_streams[i];
370 if (rtsp_st->sdp_payload_type == payload_type &&
371 rtsp_st->dynamic_handler &&
372 rtsp_st->dynamic_handler->parse_sdp_a_line) {
373 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
374 rtsp_st->dynamic_protocol_context, line);
379 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
380 int letter, const char *buf)
382 RTSPState *rt = s->priv_data;
383 char buf1[64], st_type[64];
385 enum AVMediaType codec_type;
389 RTSPSource *rtsp_src;
390 struct sockaddr_storage sdp_ip;
393 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
396 if (s1->skip_media && letter != 'm')
400 get_word(buf1, sizeof(buf1), &p);
401 if (strcmp(buf1, "IN") != 0)
403 get_word(buf1, sizeof(buf1), &p);
404 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
406 get_word_sep(buf1, sizeof(buf1), "/", &p);
407 if (get_sockaddr(s, buf1, &sdp_ip))
412 get_word_sep(buf1, sizeof(buf1), "/", &p);
415 if (s->nb_streams == 0) {
416 s1->default_ip = sdp_ip;
417 s1->default_ttl = ttl;
419 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
420 rtsp_st->sdp_ip = sdp_ip;
421 rtsp_st->sdp_ttl = ttl;
425 av_dict_set(&s->metadata, "title", p, 0);
428 if (s->nb_streams == 0) {
429 av_dict_set(&s->metadata, "comment", p, 0);
438 codec_type = AVMEDIA_TYPE_UNKNOWN;
439 get_word(st_type, sizeof(st_type), &p);
440 if (!strcmp(st_type, "audio")) {
441 codec_type = AVMEDIA_TYPE_AUDIO;
442 } else if (!strcmp(st_type, "video")) {
443 codec_type = AVMEDIA_TYPE_VIDEO;
444 } else if (!strcmp(st_type, "application")) {
445 codec_type = AVMEDIA_TYPE_DATA;
446 } else if (!strcmp(st_type, "text")) {
447 codec_type = AVMEDIA_TYPE_SUBTITLE;
449 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
453 rtsp_st = av_mallocz(sizeof(RTSPStream));
456 rtsp_st->stream_index = -1;
457 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
459 rtsp_st->sdp_ip = s1->default_ip;
460 rtsp_st->sdp_ttl = s1->default_ttl;
462 copy_default_source_addrs(s1->default_include_source_addrs,
463 s1->nb_default_include_source_addrs,
464 &rtsp_st->include_source_addrs,
465 &rtsp_st->nb_include_source_addrs);
466 copy_default_source_addrs(s1->default_exclude_source_addrs,
467 s1->nb_default_exclude_source_addrs,
468 &rtsp_st->exclude_source_addrs,
469 &rtsp_st->nb_exclude_source_addrs);
471 get_word(buf1, sizeof(buf1), &p); /* port */
472 rtsp_st->sdp_port = atoi(buf1);
474 get_word(buf1, sizeof(buf1), &p); /* protocol */
475 if (!strcmp(buf1, "udp"))
476 rt->transport = RTSP_TRANSPORT_RAW;
477 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
478 rtsp_st->feedback = 1;
480 /* XXX: handle list of formats */
481 get_word(buf1, sizeof(buf1), &p); /* format list */
482 rtsp_st->sdp_payload_type = atoi(buf1);
484 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
485 /* no corresponding stream */
486 if (rt->transport == RTSP_TRANSPORT_RAW) {
487 if (CONFIG_RTPDEC && !rt->ts)
488 rt->ts = avpriv_mpegts_parse_open(s);
490 RTPDynamicProtocolHandler *handler;
491 handler = ff_rtp_handler_find_by_id(
492 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
493 init_rtp_handler(handler, rtsp_st, NULL);
494 finalize_rtp_handler_init(s, rtsp_st, NULL);
496 } else if (rt->server_type == RTSP_SERVER_WMS &&
497 codec_type == AVMEDIA_TYPE_DATA) {
498 /* RTX stream, a stream that carries all the other actual
499 * audio/video streams. Don't expose this to the callers. */
501 st = avformat_new_stream(s, NULL);
504 st->id = rt->nb_rtsp_streams - 1;
505 rtsp_st->stream_index = st->index;
506 st->codecpar->codec_type = codec_type;
507 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
508 RTPDynamicProtocolHandler *handler;
509 /* if standard payload type, we can find the codec right now */
510 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
511 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
512 st->codecpar->sample_rate > 0)
513 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
514 /* Even static payload types may need a custom depacketizer */
515 handler = ff_rtp_handler_find_by_id(
516 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
517 init_rtp_handler(handler, rtsp_st, st);
518 finalize_rtp_handler_init(s, rtsp_st, st);
520 if (rt->default_lang[0])
521 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
523 /* put a default control url */
524 av_strlcpy(rtsp_st->control_url, rt->control_uri,
525 sizeof(rtsp_st->control_url));
528 if (av_strstart(p, "control:", &p)) {
529 if (s->nb_streams == 0) {
530 if (!strncmp(p, "rtsp://", 7))
531 av_strlcpy(rt->control_uri, p,
532 sizeof(rt->control_uri));
535 /* get the control url */
536 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
538 /* XXX: may need to add full url resolution */
539 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
541 if (proto[0] == '\0') {
542 /* relative control URL */
543 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
544 av_strlcat(rtsp_st->control_url, "/",
545 sizeof(rtsp_st->control_url));
546 av_strlcat(rtsp_st->control_url, p,
547 sizeof(rtsp_st->control_url));
549 av_strlcpy(rtsp_st->control_url, p,
550 sizeof(rtsp_st->control_url));
552 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
553 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
554 get_word(buf1, sizeof(buf1), &p);
555 payload_type = atoi(buf1);
556 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
557 if (rtsp_st->stream_index >= 0) {
558 st = s->streams[rtsp_st->stream_index];
559 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
563 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
565 } else if (av_strstart(p, "fmtp:", &p) ||
566 av_strstart(p, "framesize:", &p)) {
567 // let dynamic protocol handlers have a stab at the line.
568 get_word(buf1, sizeof(buf1), &p);
569 payload_type = atoi(buf1);
570 if (s1->seen_rtpmap) {
571 parse_fmtp(s, rt, payload_type, buf);
574 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
576 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
577 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
578 get_word(buf1, sizeof(buf1), &p);
579 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
580 } else if (av_strstart(p, "range:", &p)) {
583 // this is so that seeking on a streamed file can work.
584 rtsp_parse_range_npt(p, &start, &end);
585 s->start_time = start;
586 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
587 s->duration = (end == AV_NOPTS_VALUE) ?
588 AV_NOPTS_VALUE : end - start;
589 } else if (av_strstart(p, "lang:", &p)) {
590 if (s->nb_streams > 0) {
591 get_word(buf1, sizeof(buf1), &p);
592 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593 if (rtsp_st->stream_index >= 0) {
594 st = s->streams[rtsp_st->stream_index];
595 av_dict_set(&st->metadata, "language", buf1, 0);
598 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
599 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
601 rt->transport = RTSP_TRANSPORT_RDT;
602 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
604 st = s->streams[s->nb_streams - 1];
605 st->codecpar->sample_rate = atoi(p);
606 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
608 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
609 get_word(buf1, sizeof(buf1), &p); // ignore tag
610 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
611 p += strspn(p, SPACE_CHARS);
612 if (av_strstart(p, "inline:", &p))
613 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
614 } else if (av_strstart(p, "source-filter:", &p)) {
616 get_word(buf1, sizeof(buf1), &p);
617 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
619 exclude = !strcmp(buf1, "excl");
621 get_word(buf1, sizeof(buf1), &p);
622 if (strcmp(buf1, "IN") != 0)
624 get_word(buf1, sizeof(buf1), &p);
625 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
627 // not checking that the destination address actually matches or is wildcard
628 get_word(buf1, sizeof(buf1), &p);
631 rtsp_src = av_mallocz(sizeof(*rtsp_src));
634 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
636 if (s->nb_streams == 0) {
637 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
639 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
640 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
643 if (s->nb_streams == 0) {
644 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
646 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
647 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
652 if (rt->server_type == RTSP_SERVER_WMS)
653 ff_wms_parse_sdp_a_line(s, p);
654 if (s->nb_streams > 0) {
655 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
657 if (rt->server_type == RTSP_SERVER_REAL)
658 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
660 if (rtsp_st->dynamic_handler &&
661 rtsp_st->dynamic_handler->parse_sdp_a_line)
662 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
663 rtsp_st->stream_index,
664 rtsp_st->dynamic_protocol_context, buf);
671 int ff_sdp_parse(AVFormatContext *s, const char *content)
673 RTSPState *rt = s->priv_data;
676 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
677 * contain long SDP lines containing complete ASF Headers (several
678 * kB) or arrays of MDPR (RM stream descriptor) headers plus
679 * "rulebooks" describing their properties. Therefore, the SDP line
682 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
683 * in rtpdec_xiph.c. */
685 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
689 p += strspn(p, SPACE_CHARS);
697 /* get the content */
699 while (*p != '\n' && *p != '\r' && *p != '\0') {
700 if ((q - buf) < sizeof(buf) - 1)
705 sdp_parse_line(s, s1, letter, buf);
707 while (*p != '\n' && *p != '\0')
713 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
714 av_freep(&s1->default_include_source_addrs[i]);
715 av_freep(&s1->default_include_source_addrs);
716 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
717 av_freep(&s1->default_exclude_source_addrs[i]);
718 av_freep(&s1->default_exclude_source_addrs);
720 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
721 if (!rt->p) return AVERROR(ENOMEM);
724 #endif /* CONFIG_RTPDEC */
726 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
728 RTSPState *rt = s->priv_data;
731 for (i = 0; i < rt->nb_rtsp_streams; i++) {
732 RTSPStream *rtsp_st = rt->rtsp_streams[i];
735 if (rtsp_st->transport_priv) {
737 AVFormatContext *rtpctx = rtsp_st->transport_priv;
738 av_write_trailer(rtpctx);
739 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
740 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
741 ff_rtsp_tcp_write_packet(s, rtsp_st);
742 ffio_free_dyn_buf(&rtpctx->pb);
744 avio_closep(&rtpctx->pb);
746 avformat_free_context(rtpctx);
747 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
748 ff_rdt_parse_close(rtsp_st->transport_priv);
749 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
750 ff_rtp_parse_close(rtsp_st->transport_priv);
752 rtsp_st->transport_priv = NULL;
753 if (rtsp_st->rtp_handle)
754 ffurl_close(rtsp_st->rtp_handle);
755 rtsp_st->rtp_handle = NULL;
759 /* close and free RTSP streams */
760 void ff_rtsp_close_streams(AVFormatContext *s)
762 RTSPState *rt = s->priv_data;
766 ff_rtsp_undo_setup(s, 0);
767 for (i = 0; i < rt->nb_rtsp_streams; i++) {
768 rtsp_st = rt->rtsp_streams[i];
770 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
771 if (rtsp_st->dynamic_handler->close)
772 rtsp_st->dynamic_handler->close(
773 rtsp_st->dynamic_protocol_context);
774 av_free(rtsp_st->dynamic_protocol_context);
776 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
777 av_freep(&rtsp_st->include_source_addrs[j]);
778 av_freep(&rtsp_st->include_source_addrs);
779 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
780 av_freep(&rtsp_st->exclude_source_addrs[j]);
781 av_freep(&rtsp_st->exclude_source_addrs);
786 av_freep(&rt->rtsp_streams);
788 avformat_close_input(&rt->asf_ctx);
790 if (CONFIG_RTPDEC && rt->ts)
791 avpriv_mpegts_parse_close(rt->ts);
793 av_freep(&rt->recvbuf);
796 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
798 RTSPState *rt = s->priv_data;
800 int reordering_queue_size = rt->reordering_queue_size;
801 if (reordering_queue_size < 0) {
802 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
803 reordering_queue_size = 0;
805 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
808 /* open the RTP context */
809 if (rtsp_st->stream_index >= 0)
810 st = s->streams[rtsp_st->stream_index];
812 s->ctx_flags |= AVFMTCTX_NOHEADER;
814 if (CONFIG_RTSP_MUXER && s->oformat && st) {
815 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
816 s, st, rtsp_st->rtp_handle,
817 RTSP_TCP_MAX_PACKET_SIZE,
818 rtsp_st->stream_index);
819 /* Ownership of rtp_handle is passed to the rtp mux context */
820 rtsp_st->rtp_handle = NULL;
823 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
824 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
825 return 0; // Don't need to open any parser here
826 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
827 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
828 rtsp_st->dynamic_protocol_context,
829 rtsp_st->dynamic_handler);
830 else if (CONFIG_RTPDEC)
831 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
832 rtsp_st->sdp_payload_type,
833 reordering_queue_size);
835 if (!rtsp_st->transport_priv) {
836 return AVERROR(ENOMEM);
837 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
839 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
840 rtpctx->ssrc = rtsp_st->ssrc;
841 if (rtsp_st->dynamic_handler) {
842 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
843 rtsp_st->dynamic_protocol_context,
844 rtsp_st->dynamic_handler);
846 if (rtsp_st->crypto_suite[0])
847 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
848 rtsp_st->crypto_suite,
849 rtsp_st->crypto_params);
855 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
856 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
863 q += strspn(q, SPACE_CHARS);
864 v = strtol(q, &p, 10);
868 v = strtol(p, &p, 10);
877 /* XXX: only one transport specification is parsed */
878 static void rtsp_parse_transport(AVFormatContext *s,
879 RTSPMessageHeader *reply, const char *p)
881 char transport_protocol[16];
883 char lower_transport[16];
885 RTSPTransportField *th;
888 reply->nb_transports = 0;
891 p += strspn(p, SPACE_CHARS);
895 th = &reply->transports[reply->nb_transports];
897 get_word_sep(transport_protocol, sizeof(transport_protocol),
899 if (!av_strcasecmp (transport_protocol, "rtp")) {
900 get_word_sep(profile, sizeof(profile), "/;,", &p);
901 lower_transport[0] = '\0';
902 /* rtp/avp/<protocol> */
904 get_word_sep(lower_transport, sizeof(lower_transport),
907 th->transport = RTSP_TRANSPORT_RTP;
908 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
909 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
910 /* x-pn-tng/<protocol> */
911 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
913 th->transport = RTSP_TRANSPORT_RDT;
914 } else if (!av_strcasecmp(transport_protocol, "raw")) {
915 get_word_sep(profile, sizeof(profile), "/;,", &p);
916 lower_transport[0] = '\0';
917 /* raw/raw/<protocol> */
919 get_word_sep(lower_transport, sizeof(lower_transport),
922 th->transport = RTSP_TRANSPORT_RAW;
924 if (!av_strcasecmp(lower_transport, "TCP"))
925 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
927 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
931 /* get each parameter */
932 while (*p != '\0' && *p != ',') {
933 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
934 if (!strcmp(parameter, "port")) {
937 rtsp_parse_range(&th->port_min, &th->port_max, &p);
939 } else if (!strcmp(parameter, "client_port")) {
942 rtsp_parse_range(&th->client_port_min,
943 &th->client_port_max, &p);
945 } else if (!strcmp(parameter, "server_port")) {
948 rtsp_parse_range(&th->server_port_min,
949 &th->server_port_max, &p);
951 } else if (!strcmp(parameter, "interleaved")) {
954 rtsp_parse_range(&th->interleaved_min,
955 &th->interleaved_max, &p);
957 } else if (!strcmp(parameter, "multicast")) {
958 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
959 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
960 } else if (!strcmp(parameter, "ttl")) {
964 th->ttl = strtol(p, &end, 10);
967 } else if (!strcmp(parameter, "destination")) {
970 get_word_sep(buf, sizeof(buf), ";,", &p);
971 get_sockaddr(s, buf, &th->destination);
973 } else if (!strcmp(parameter, "source")) {
976 get_word_sep(buf, sizeof(buf), ";,", &p);
977 av_strlcpy(th->source, buf, sizeof(th->source));
979 } else if (!strcmp(parameter, "mode")) {
982 get_word_sep(buf, sizeof(buf), ";, ", &p);
983 if (!strcmp(buf, "record") ||
984 !strcmp(buf, "receive"))
989 while (*p != ';' && *p != '\0' && *p != ',')
997 reply->nb_transports++;
998 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1003 static void handle_rtp_info(RTSPState *rt, const char *url,
1004 uint32_t seq, uint32_t rtptime)
1007 if (!rtptime || !url[0])
1009 if (rt->transport != RTSP_TRANSPORT_RTP)
1011 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1012 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1013 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1016 if (!strcmp(rtsp_st->control_url, url)) {
1017 rtpctx->base_timestamp = rtptime;
1023 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1026 char key[20], value[1024], url[1024] = "";
1027 uint32_t seq = 0, rtptime = 0;
1030 p += strspn(p, SPACE_CHARS);
1033 get_word_sep(key, sizeof(key), "=", &p);
1037 get_word_sep(value, sizeof(value), ";, ", &p);
1039 if (!strcmp(key, "url"))
1040 av_strlcpy(url, value, sizeof(url));
1041 else if (!strcmp(key, "seq"))
1042 seq = strtoul(value, NULL, 10);
1043 else if (!strcmp(key, "rtptime"))
1044 rtptime = strtoul(value, NULL, 10);
1046 handle_rtp_info(rt, url, seq, rtptime);
1055 handle_rtp_info(rt, url, seq, rtptime);
1058 void ff_rtsp_parse_line(AVFormatContext *s,
1059 RTSPMessageHeader *reply, const char *buf,
1060 RTSPState *rt, const char *method)
1064 /* NOTE: we do case independent match for broken servers */
1066 if (av_stristart(p, "Session:", &p)) {
1068 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1069 if (av_stristart(p, ";timeout=", &p) &&
1070 (t = strtol(p, NULL, 10)) > 0) {
1073 } else if (av_stristart(p, "Content-Length:", &p)) {
1074 reply->content_length = strtol(p, NULL, 10);
1075 } else if (av_stristart(p, "Transport:", &p)) {
1076 rtsp_parse_transport(s, reply, p);
1077 } else if (av_stristart(p, "CSeq:", &p)) {
1078 reply->seq = strtol(p, NULL, 10);
1079 } else if (av_stristart(p, "Range:", &p)) {
1080 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1081 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1082 p += strspn(p, SPACE_CHARS);
1083 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1084 } else if (av_stristart(p, "Server:", &p)) {
1085 p += strspn(p, SPACE_CHARS);
1086 av_strlcpy(reply->server, p, sizeof(reply->server));
1087 } else if (av_stristart(p, "Notice:", &p) ||
1088 av_stristart(p, "X-Notice:", &p)) {
1089 reply->notice = strtol(p, NULL, 10);
1090 } else if (av_stristart(p, "Location:", &p)) {
1091 p += strspn(p, SPACE_CHARS);
1092 av_strlcpy(reply->location, p , sizeof(reply->location));
1093 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1094 p += strspn(p, SPACE_CHARS);
1095 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1096 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1097 p += strspn(p, SPACE_CHARS);
1098 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1099 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1100 p += strspn(p, SPACE_CHARS);
1101 if (method && !strcmp(method, "DESCRIBE"))
1102 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1103 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1104 p += strspn(p, SPACE_CHARS);
1105 if (method && !strcmp(method, "PLAY"))
1106 rtsp_parse_rtp_info(rt, p);
1107 } else if (av_stristart(p, "Public:", &p) && rt) {
1108 if (strstr(p, "GET_PARAMETER") &&
1109 method && !strcmp(method, "OPTIONS"))
1110 rt->get_parameter_supported = 1;
1111 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1112 p += strspn(p, SPACE_CHARS);
1113 rt->accept_dynamic_rate = atoi(p);
1114 } else if (av_stristart(p, "Content-Type:", &p)) {
1115 p += strspn(p, SPACE_CHARS);
1116 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1120 /* skip a RTP/TCP interleaved packet */
1121 void ff_rtsp_skip_packet(AVFormatContext *s)
1123 RTSPState *rt = s->priv_data;
1127 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1130 len = AV_RB16(buf + 1);
1132 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1137 if (len1 > sizeof(buf))
1139 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1146 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1147 unsigned char **content_ptr,
1148 int return_on_interleaved_data, const char *method)
1150 RTSPState *rt = s->priv_data;
1151 char buf[4096], buf1[1024], *q;
1154 int ret, content_length, line_count = 0, request = 0;
1155 unsigned char *content = NULL;
1161 memset(reply, 0, sizeof(*reply));
1163 /* parse reply (XXX: use buffers) */
1164 rt->last_reply[0] = '\0';
1168 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1169 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1174 if (ch == '$' && q == buf) {
1175 if (return_on_interleaved_data) {
1178 ff_rtsp_skip_packet(s);
1179 } else if (ch != '\r') {
1180 if ((q - buf) < sizeof(buf) - 1)
1186 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1188 /* test if last line */
1192 if (line_count == 0) {
1193 /* get reply code */
1194 get_word(buf1, sizeof(buf1), &p);
1195 if (!strncmp(buf1, "RTSP/", 5)) {
1196 get_word(buf1, sizeof(buf1), &p);
1197 reply->status_code = atoi(buf1);
1198 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1200 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1201 get_word(buf1, sizeof(buf1), &p); // object
1205 ff_rtsp_parse_line(s, reply, p, rt, method);
1206 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1207 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1212 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1213 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1215 content_length = reply->content_length;
1216 if (content_length > 0) {
1217 /* leave some room for a trailing '\0' (useful for simple parsing) */
1218 content = av_malloc(content_length + 1);
1220 return AVERROR(ENOMEM);
1221 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1222 content[content_length] = '\0';
1225 *content_ptr = content;
1231 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1232 const char* ptr = buf;
1234 if (!strcmp(reply->reason, "OPTIONS")) {
1235 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1237 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1238 if (reply->session_id[0])
1239 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1242 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1244 av_strlcat(buf, "\r\n", sizeof(buf));
1246 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1247 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1250 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1252 rt->last_cmd_time = av_gettime_relative();
1253 /* Even if the request from the server had data, it is not the data
1254 * that the caller wants or expects. The memory could also be leaked
1255 * if the actual following reply has content data. */
1257 av_freep(content_ptr);
1258 /* If method is set, this is called from ff_rtsp_send_cmd,
1259 * where a reply to exactly this request is awaited. For
1260 * callers from within packet receiving, we just want to
1261 * return to the caller and go back to receiving packets. */
1267 if (rt->seq != reply->seq) {
1268 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1269 rt->seq, reply->seq);
1273 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1274 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1275 reply->notice == 2306 /* Continuous Feed Terminated */) {
1276 rt->state = RTSP_STATE_IDLE;
1277 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1278 return AVERROR(EIO); /* data or server error */
1279 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1280 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1281 return AVERROR(EPERM);
1287 * Send a command to the RTSP server without waiting for the reply.
1289 * @param s RTSP (de)muxer context
1290 * @param method the method for the request
1291 * @param url the target url for the request
1292 * @param headers extra header lines to include in the request
1293 * @param send_content if non-null, the data to send as request body content
1294 * @param send_content_length the length of the send_content data, or 0 if
1295 * send_content is null
1297 * @return zero if success, nonzero otherwise
1299 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1300 const char *method, const char *url,
1301 const char *headers,
1302 const unsigned char *send_content,
1303 int send_content_length)
1305 RTSPState *rt = s->priv_data;
1306 char buf[4096], *out_buf;
1307 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1309 /* Add in RTSP headers */
1312 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1314 av_strlcat(buf, headers, sizeof(buf));
1315 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1316 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1317 if (rt->session_id[0] != '\0' && (!headers ||
1318 !strstr(headers, "\nIf-Match:"))) {
1319 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1322 char *str = ff_http_auth_create_response(&rt->auth_state,
1323 rt->auth, url, method);
1325 av_strlcat(buf, str, sizeof(buf));
1328 if (send_content_length > 0 && send_content)
1329 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1330 av_strlcat(buf, "\r\n", sizeof(buf));
1332 /* base64 encode rtsp if tunneling */
1333 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1334 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1335 out_buf = base64buf;
1338 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1340 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1341 if (send_content_length > 0 && send_content) {
1342 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1343 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1344 "with content data not supported\n");
1345 return AVERROR_PATCHWELCOME;
1347 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1349 rt->last_cmd_time = av_gettime_relative();
1354 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1355 const char *url, const char *headers)
1357 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1360 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1361 const char *headers, RTSPMessageHeader *reply,
1362 unsigned char **content_ptr)
1364 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1365 content_ptr, NULL, 0);
1368 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1369 const char *method, const char *url,
1371 RTSPMessageHeader *reply,
1372 unsigned char **content_ptr,
1373 const unsigned char *send_content,
1374 int send_content_length)
1376 RTSPState *rt = s->priv_data;
1377 HTTPAuthType cur_auth_type;
1378 int ret, attempts = 0;
1381 cur_auth_type = rt->auth_state.auth_type;
1382 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1384 send_content_length)))
1387 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1391 if (reply->status_code == 401 &&
1392 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1393 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1396 if (reply->status_code > 400){
1397 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1401 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1407 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1408 int lower_transport, const char *real_challenge)
1410 RTSPState *rt = s->priv_data;
1411 int rtx = 0, j, i, err, interleave = 0, port_off;
1412 RTSPStream *rtsp_st;
1413 RTSPMessageHeader reply1, *reply = &reply1;
1415 const char *trans_pref;
1417 if (rt->transport == RTSP_TRANSPORT_RDT)
1418 trans_pref = "x-pn-tng";
1419 else if (rt->transport == RTSP_TRANSPORT_RAW)
1420 trans_pref = "RAW/RAW";
1422 trans_pref = "RTP/AVP";
1424 /* default timeout: 1 minute */
1427 /* Choose a random starting offset within the first half of the
1428 * port range, to allow for a number of ports to try even if the offset
1429 * happens to be at the end of the random range. */
1430 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1431 /* even random offset */
1432 port_off -= port_off & 0x01;
1434 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1435 char transport[2048];
1438 * WMS serves all UDP data over a single connection, the RTX, which
1439 * isn't necessarily the first in the SDP but has to be the first
1440 * to be set up, else the second/third SETUP will fail with a 461.
1442 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1443 rt->server_type == RTSP_SERVER_WMS) {
1446 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1447 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1449 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1453 if (rtx == rt->nb_rtsp_streams)
1454 return -1; /* no RTX found */
1455 rtsp_st = rt->rtsp_streams[rtx];
1457 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1459 rtsp_st = rt->rtsp_streams[i];
1462 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1465 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1466 port = reply->transports[0].client_port_min;
1470 /* first try in specified port range */
1471 while (j <= rt->rtp_port_max) {
1472 AVDictionary *opts = map_to_opts(rt);
1474 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1475 "?localport=%d", j);
1476 /* we will use two ports per rtp stream (rtp and rtcp) */
1478 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1479 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1481 av_dict_free(&opts);
1486 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1491 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1493 snprintf(transport, sizeof(transport) - 1,
1494 "%s/UDP;", trans_pref);
1495 if (rt->server_type != RTSP_SERVER_REAL)
1496 av_strlcat(transport, "unicast;", sizeof(transport));
1497 av_strlcatf(transport, sizeof(transport),
1498 "client_port=%d", port);
1499 if (rt->transport == RTSP_TRANSPORT_RTP &&
1500 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1501 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1505 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1506 /* For WMS streams, the application streams are only used for
1507 * UDP. When trying to set it up for TCP streams, the server
1508 * will return an error. Therefore, we skip those streams. */
1509 if (rt->server_type == RTSP_SERVER_WMS &&
1510 (rtsp_st->stream_index < 0 ||
1511 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1514 snprintf(transport, sizeof(transport) - 1,
1515 "%s/TCP;", trans_pref);
1516 if (rt->transport != RTSP_TRANSPORT_RDT)
1517 av_strlcat(transport, "unicast;", sizeof(transport));
1518 av_strlcatf(transport, sizeof(transport),
1519 "interleaved=%d-%d",
1520 interleave, interleave + 1);
1524 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1525 snprintf(transport, sizeof(transport) - 1,
1526 "%s/UDP;multicast", trans_pref);
1529 av_strlcat(transport, ";mode=record", sizeof(transport));
1530 } else if (rt->server_type == RTSP_SERVER_REAL ||
1531 rt->server_type == RTSP_SERVER_WMS)
1532 av_strlcat(transport, ";mode=play", sizeof(transport));
1533 snprintf(cmd, sizeof(cmd),
1534 "Transport: %s\r\n",
1536 if (rt->accept_dynamic_rate)
1537 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1538 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1539 char real_res[41], real_csum[9];
1540 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1542 av_strlcatf(cmd, sizeof(cmd),
1544 "RealChallenge2: %s, sd=%s\r\n",
1545 rt->session_id, real_res, real_csum);
1547 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1548 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1551 } else if (reply->status_code != RTSP_STATUS_OK ||
1552 reply->nb_transports != 1) {
1553 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1557 /* XXX: same protocol for all streams is required */
1559 if (reply->transports[0].lower_transport != rt->lower_transport ||
1560 reply->transports[0].transport != rt->transport) {
1561 err = AVERROR_INVALIDDATA;
1565 rt->lower_transport = reply->transports[0].lower_transport;
1566 rt->transport = reply->transports[0].transport;
1569 /* Fail if the server responded with another lower transport mode
1570 * than what we requested. */
1571 if (reply->transports[0].lower_transport != lower_transport) {
1572 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1573 err = AVERROR_INVALIDDATA;
1577 switch(reply->transports[0].lower_transport) {
1578 case RTSP_LOWER_TRANSPORT_TCP:
1579 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1580 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1583 case RTSP_LOWER_TRANSPORT_UDP: {
1584 char url[1024], options[30] = "";
1585 const char *peer = host;
1587 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1588 av_strlcpy(options, "?connect=1", sizeof(options));
1589 /* Use source address if specified */
1590 if (reply->transports[0].source[0])
1591 peer = reply->transports[0].source;
1592 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1593 reply->transports[0].server_port_min, "%s", options);
1594 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1595 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1596 err = AVERROR_INVALIDDATA;
1601 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1602 char url[1024], namebuf[50], optbuf[20] = "";
1603 struct sockaddr_storage addr;
1606 if (reply->transports[0].destination.ss_family) {
1607 addr = reply->transports[0].destination;
1608 port = reply->transports[0].port_min;
1609 ttl = reply->transports[0].ttl;
1611 addr = rtsp_st->sdp_ip;
1612 port = rtsp_st->sdp_port;
1613 ttl = rtsp_st->sdp_ttl;
1616 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1617 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1618 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1619 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1620 port, "%s", optbuf);
1621 if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1622 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL) < 0) {
1623 err = AVERROR_INVALIDDATA;
1630 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1634 if (rt->nb_rtsp_streams && reply->timeout > 0)
1635 rt->timeout = reply->timeout;
1637 if (rt->server_type == RTSP_SERVER_REAL)
1638 rt->need_subscription = 1;
1643 ff_rtsp_undo_setup(s, 0);
1647 void ff_rtsp_close_connections(AVFormatContext *s)
1649 RTSPState *rt = s->priv_data;
1650 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1651 ffurl_close(rt->rtsp_hd);
1652 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1655 int ff_rtsp_connect(AVFormatContext *s)
1657 RTSPState *rt = s->priv_data;
1658 char proto[128], host[1024], path[1024];
1659 char tcpname[1024], cmd[2048], auth[128];
1660 const char *lower_rtsp_proto = "tcp";
1661 int port, err, tcp_fd;
1662 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1663 int lower_transport_mask = 0;
1664 int default_port = RTSP_DEFAULT_PORT;
1665 char real_challenge[64] = "";
1666 struct sockaddr_storage peer;
1667 socklen_t peer_len = sizeof(peer);
1669 if (rt->rtp_port_max < rt->rtp_port_min) {
1670 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1671 "than min port %d\n", rt->rtp_port_max,
1673 return AVERROR(EINVAL);
1676 if (!ff_network_init())
1677 return AVERROR(EIO);
1679 if (s->max_delay < 0) /* Not set by the caller */
1680 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1682 rt->control_transport = RTSP_MODE_PLAIN;
1683 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1684 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1685 rt->control_transport = RTSP_MODE_TUNNEL;
1687 /* Only pass through valid flags from here */
1688 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1691 /* extract hostname and port */
1692 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1693 host, sizeof(host), &port, path, sizeof(path), s->filename);
1695 if (!strcmp(proto, "rtsps")) {
1696 lower_rtsp_proto = "tls";
1697 default_port = RTSPS_DEFAULT_PORT;
1698 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1702 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1705 port = default_port;
1707 lower_transport_mask = rt->lower_transport_mask;
1709 if (!lower_transport_mask)
1710 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1713 /* Only UDP or TCP - UDP multicast isn't supported. */
1714 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1715 (1 << RTSP_LOWER_TRANSPORT_TCP);
1716 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1717 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1718 "only UDP and TCP are supported for output.\n");
1719 err = AVERROR(EINVAL);
1724 /* Construct the URI used in request; this is similar to s->filename,
1725 * but with authentication credentials removed and RTSP specific options
1727 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1728 host, port, "%s", path);
1730 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1731 /* set up initial handshake for tunneling */
1732 char httpname[1024];
1733 char sessioncookie[17];
1736 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1737 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1738 av_get_random_seed(), av_get_random_seed());
1741 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1742 &s->interrupt_callback) < 0) {
1747 /* generate GET headers */
1748 snprintf(headers, sizeof(headers),
1749 "x-sessioncookie: %s\r\n"
1750 "Accept: application/x-rtsp-tunnelled\r\n"
1751 "Pragma: no-cache\r\n"
1752 "Cache-Control: no-cache\r\n",
1754 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1756 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1757 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1758 if (!rt->rtsp_hd->protocol_whitelist) {
1759 err = AVERROR(ENOMEM);
1764 /* complete the connection */
1765 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1771 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1772 &s->interrupt_callback) < 0 ) {
1777 /* generate POST headers */
1778 snprintf(headers, sizeof(headers),
1779 "x-sessioncookie: %s\r\n"
1780 "Content-Type: application/x-rtsp-tunnelled\r\n"
1781 "Pragma: no-cache\r\n"
1782 "Cache-Control: no-cache\r\n"
1783 "Content-Length: 32767\r\n"
1784 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1786 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1787 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1789 /* Initialize the authentication state for the POST session. The HTTP
1790 * protocol implementation doesn't properly handle multi-pass
1791 * authentication for POST requests, since it would require one of
1793 * - implementing Expect: 100-continue, which many HTTP servers
1794 * don't support anyway, even less the RTSP servers that do HTTP
1796 * - sending the whole POST data until getting a 401 reply specifying
1797 * what authentication method to use, then resending all that data
1798 * - waiting for potential 401 replies directly after sending the
1799 * POST header (waiting for some unspecified time)
1800 * Therefore, we copy the full auth state, which works for both basic
1801 * and digest. (For digest, we would have to synchronize the nonce
1802 * count variable between the two sessions, if we'd do more requests
1803 * with the original session, though.)
1805 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1807 /* complete the connection */
1808 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1814 /* open the tcp connection */
1815 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1817 "?timeout=%d", rt->stimeout);
1818 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1819 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1823 rt->rtsp_hd_out = rt->rtsp_hd;
1827 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1832 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1833 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1834 NULL, 0, NI_NUMERICHOST);
1837 /* request options supported by the server; this also detects server
1839 for (rt->server_type = RTSP_SERVER_RTP;;) {
1841 if (rt->server_type == RTSP_SERVER_REAL)
1844 * The following entries are required for proper
1845 * streaming from a Realmedia server. They are
1846 * interdependent in some way although we currently
1847 * don't quite understand how. Values were copied
1848 * from mplayer SVN r23589.
1849 * ClientChallenge is a 16-byte ID in hex
1850 * CompanyID is a 16-byte ID in base64
1852 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1853 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1854 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1855 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1857 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1858 if (reply->status_code != RTSP_STATUS_OK) {
1859 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1863 /* detect server type if not standard-compliant RTP */
1864 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1865 rt->server_type = RTSP_SERVER_REAL;
1867 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1868 rt->server_type = RTSP_SERVER_WMS;
1869 } else if (rt->server_type == RTSP_SERVER_REAL)
1870 strcpy(real_challenge, reply->real_challenge);
1874 if (CONFIG_RTSP_DEMUXER && s->iformat)
1875 err = ff_rtsp_setup_input_streams(s, reply);
1876 else if (CONFIG_RTSP_MUXER)
1877 err = ff_rtsp_setup_output_streams(s, host);
1884 int lower_transport = ff_log2_tab[lower_transport_mask &
1885 ~(lower_transport_mask - 1)];
1887 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1888 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1889 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1891 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1892 rt->server_type == RTSP_SERVER_REAL ?
1893 real_challenge : NULL);
1896 lower_transport_mask &= ~(1 << lower_transport);
1897 if (lower_transport_mask == 0 && err == 1) {
1898 err = AVERROR(EPROTONOSUPPORT);
1903 rt->lower_transport_mask = lower_transport_mask;
1904 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1905 rt->state = RTSP_STATE_IDLE;
1906 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1909 ff_rtsp_close_streams(s);
1910 ff_rtsp_close_connections(s);
1911 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1912 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1913 rt->session_id[0] = '\0';
1914 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1922 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1925 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1926 uint8_t *buf, int buf_size, int64_t wait_end)
1928 RTSPState *rt = s->priv_data;
1929 RTSPStream *rtsp_st;
1930 int n, i, ret, tcp_fd, timeout_cnt = 0;
1932 struct pollfd *p = rt->p;
1933 int *fds = NULL, fdsnum, fdsidx;
1936 if (ff_check_interrupt(&s->interrupt_callback))
1937 return AVERROR_EXIT;
1938 if (wait_end && wait_end - av_gettime_relative() < 0)
1939 return AVERROR(EAGAIN);
1942 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1943 p[max_p].fd = tcp_fd;
1944 p[max_p++].events = POLLIN;
1948 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1949 rtsp_st = rt->rtsp_streams[i];
1950 if (rtsp_st->rtp_handle) {
1951 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1953 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1957 av_log(s, AV_LOG_ERROR,
1958 "Number of fds %d not supported\n", fdsnum);
1959 return AVERROR_INVALIDDATA;
1961 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1962 p[max_p].fd = fds[fdsidx];
1963 p[max_p++].events = POLLIN;
1968 n = poll(p, max_p, POLL_TIMEOUT_MS);
1970 int j = 1 - (tcp_fd == -1);
1972 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973 rtsp_st = rt->rtsp_streams[i];
1974 if (rtsp_st->rtp_handle) {
1975 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1976 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1978 *prtsp_st = rtsp_st;
1985 #if CONFIG_RTSP_DEMUXER
1986 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1987 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1988 if (rt->state == RTSP_STATE_STREAMING) {
1989 if (!ff_rtsp_parse_streaming_commands(s))
1992 av_log(s, AV_LOG_WARNING,
1993 "Unable to answer to TEARDOWN\n");
1997 RTSPMessageHeader reply;
1998 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
2001 /* XXX: parse message */
2002 if (rt->state != RTSP_STATE_STREAMING)
2007 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2008 return AVERROR(ETIMEDOUT);
2009 } else if (n < 0 && errno != EINTR)
2010 return AVERROR(errno);
2014 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2015 const uint8_t *buf, int len)
2017 RTSPState *rt = s->priv_data;
2021 if (rt->nb_rtsp_streams == 1) {
2022 *rtsp_st = rt->rtsp_streams[0];
2025 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2026 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2028 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2029 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2032 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2033 *rtsp_st = rt->rtsp_streams[i];
2040 av_log(s, AV_LOG_WARNING,
2041 "Unable to pick stream for packet - SSRC not known for "
2043 return AVERROR(EAGAIN);
2046 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2047 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2048 *rtsp_st = rt->rtsp_streams[i];
2054 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2055 return AVERROR(EAGAIN);
2058 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2060 RTSPState *rt = s->priv_data;
2062 RTSPStream *rtsp_st, *first_queue_st = NULL;
2063 int64_t wait_end = 0;
2065 if (rt->nb_byes == rt->nb_rtsp_streams)
2068 /* get next frames from the same RTP packet */
2069 if (rt->cur_transport_priv) {
2070 if (rt->transport == RTSP_TRANSPORT_RDT) {
2071 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2072 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2073 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2074 } else if (CONFIG_RTPDEC && rt->ts) {
2075 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2077 rt->recvbuf_pos += ret;
2078 ret = rt->recvbuf_pos < rt->recvbuf_len;
2083 rt->cur_transport_priv = NULL;
2085 } else if (ret == 1) {
2088 rt->cur_transport_priv = NULL;
2092 if (rt->transport == RTSP_TRANSPORT_RTP) {
2094 int64_t first_queue_time = 0;
2095 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2096 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2100 queue_time = ff_rtp_queued_packet_time(rtpctx);
2101 if (queue_time && (queue_time - first_queue_time < 0 ||
2102 !first_queue_time)) {
2103 first_queue_time = queue_time;
2104 first_queue_st = rt->rtsp_streams[i];
2107 if (first_queue_time) {
2108 wait_end = first_queue_time + s->max_delay;
2111 first_queue_st = NULL;
2115 /* read next RTP packet */
2117 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2119 return AVERROR(ENOMEM);
2122 switch(rt->lower_transport) {
2124 #if CONFIG_RTSP_DEMUXER
2125 case RTSP_LOWER_TRANSPORT_TCP:
2126 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2129 case RTSP_LOWER_TRANSPORT_UDP:
2130 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2131 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2132 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2133 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2135 case RTSP_LOWER_TRANSPORT_CUSTOM:
2136 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2137 wait_end && wait_end < av_gettime_relative())
2138 len = AVERROR(EAGAIN);
2140 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2141 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2142 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2143 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2146 if (len == AVERROR(EAGAIN) && first_queue_st &&
2147 rt->transport == RTSP_TRANSPORT_RTP) {
2148 av_log(s, AV_LOG_WARNING,
2149 "max delay reached. need to consume packet\n");
2150 rtsp_st = first_queue_st;
2151 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2158 if (rt->transport == RTSP_TRANSPORT_RDT) {
2159 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2160 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2161 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2162 if (rtsp_st->feedback) {
2163 AVIOContext *pb = NULL;
2164 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2166 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2169 /* Either bad packet, or a RTCP packet. Check if the
2170 * first_rtcp_ntp_time field was initialized. */
2171 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2172 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2173 /* first_rtcp_ntp_time has been initialized for this stream,
2174 * copy the same value to all other uninitialized streams,
2175 * in order to map their timestamp origin to the same ntp time
2178 AVStream *st = NULL;
2179 if (rtsp_st->stream_index >= 0)
2180 st = s->streams[rtsp_st->stream_index];
2181 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2182 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2183 AVStream *st2 = NULL;
2184 if (rt->rtsp_streams[i]->stream_index >= 0)
2185 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2186 if (rtpctx2 && st && st2 &&
2187 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2188 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2189 rtpctx2->rtcp_ts_offset = av_rescale_q(
2190 rtpctx->rtcp_ts_offset, st->time_base,
2194 // Make real NTP start time available in AVFormatContext
2195 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2196 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2198 s->start_time_realtime -=
2199 av_rescale (rtpctx->rtcp_ts_offset,
2200 (uint64_t) rtpctx->st->time_base.num * 1000000,
2201 rtpctx->st->time_base.den);
2205 if (ret == -RTCP_BYE) {
2208 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2209 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2211 if (rt->nb_byes == rt->nb_rtsp_streams)
2215 } else if (CONFIG_RTPDEC && rt->ts) {
2216 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2219 rt->recvbuf_len = len;
2220 rt->recvbuf_pos = ret;
2221 rt->cur_transport_priv = rt->ts;
2228 return AVERROR_INVALIDDATA;
2234 /* more packets may follow, so we save the RTP context */
2235 rt->cur_transport_priv = rtsp_st->transport_priv;
2239 #endif /* CONFIG_RTPDEC */
2241 #if CONFIG_SDP_DEMUXER
2242 static int sdp_probe(AVProbeData *p1)
2244 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2246 /* we look for a line beginning "c=IN IP" */
2247 while (p < p_end && *p != '\0') {
2248 if (sizeof("c=IN IP") - 1 < p_end - p &&
2249 av_strstart(p, "c=IN IP", NULL))
2250 return AVPROBE_SCORE_EXTENSION;
2252 while (p < p_end - 1 && *p != '\n') p++;
2261 static void append_source_addrs(char *buf, int size, const char *name,
2262 int count, struct RTSPSource **addrs)
2267 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2268 for (i = 1; i < count; i++)
2269 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2272 static int sdp_read_header(AVFormatContext *s)
2274 RTSPState *rt = s->priv_data;
2275 RTSPStream *rtsp_st;
2280 if (!ff_network_init())
2281 return AVERROR(EIO);
2283 if (s->max_delay < 0) /* Not set by the caller */
2284 s->max_delay = DEFAULT_REORDERING_DELAY;
2285 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2286 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2288 /* read the whole sdp file */
2289 /* XXX: better loading */
2290 content = av_malloc(SDP_MAX_SIZE);
2292 return AVERROR(ENOMEM);
2293 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2296 return AVERROR_INVALIDDATA;
2298 content[size] ='\0';
2300 err = ff_sdp_parse(s, content);
2304 /* open each RTP stream */
2305 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2307 rtsp_st = rt->rtsp_streams[i];
2309 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2310 AVDictionary *opts = map_to_opts(rt);
2312 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2313 sizeof(rtsp_st->sdp_ip),
2314 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2316 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2318 av_dict_free(&opts);
2321 ff_url_join(url, sizeof(url), "rtp", NULL,
2322 namebuf, rtsp_st->sdp_port,
2323 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2324 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2325 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2326 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2328 append_source_addrs(url, sizeof(url), "sources",
2329 rtsp_st->nb_include_source_addrs,
2330 rtsp_st->include_source_addrs);
2331 append_source_addrs(url, sizeof(url), "block",
2332 rtsp_st->nb_exclude_source_addrs,
2333 rtsp_st->exclude_source_addrs);
2334 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2335 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2337 av_dict_free(&opts);
2340 err = AVERROR_INVALIDDATA;
2344 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2349 ff_rtsp_close_streams(s);
2354 static int sdp_read_close(AVFormatContext *s)
2356 ff_rtsp_close_streams(s);
2361 static const AVClass sdp_demuxer_class = {
2362 .class_name = "SDP demuxer",
2363 .item_name = av_default_item_name,
2364 .option = sdp_options,
2365 .version = LIBAVUTIL_VERSION_INT,
2368 AVInputFormat ff_sdp_demuxer = {
2370 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2371 .priv_data_size = sizeof(RTSPState),
2372 .read_probe = sdp_probe,
2373 .read_header = sdp_read_header,
2374 .read_packet = ff_rtsp_fetch_packet,
2375 .read_close = sdp_read_close,
2376 .priv_class = &sdp_demuxer_class,
2378 #endif /* CONFIG_SDP_DEMUXER */
2380 #if CONFIG_RTP_DEMUXER
2381 static int rtp_probe(AVProbeData *p)
2383 if (av_strstart(p->filename, "rtp:", NULL))
2384 return AVPROBE_SCORE_MAX;
2388 static int rtp_read_header(AVFormatContext *s)
2390 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2391 char host[500], sdp[500];
2393 URLContext* in = NULL;
2395 AVCodecParameters *par = NULL;
2396 struct sockaddr_storage addr;
2398 socklen_t addrlen = sizeof(addr);
2399 RTSPState *rt = s->priv_data;
2401 if (!ff_network_init())
2402 return AVERROR(EIO);
2404 ret = ffurl_open_whitelist(&in, s->filename, AVIO_FLAG_READ,
2405 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2410 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2411 if (ret == AVERROR(EAGAIN))
2416 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2420 if ((recvbuf[0] & 0xc0) != 0x80) {
2421 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2426 if (RTP_PT_IS_RTCP(recvbuf[1]))
2429 payload_type = recvbuf[1] & 0x7f;
2432 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2436 par = avcodec_parameters_alloc();
2438 ret = AVERROR(ENOMEM);
2442 if (ff_rtp_get_codec_info(par, payload_type)) {
2443 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2444 "without an SDP file describing it\n",
2448 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2449 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2450 "properly you need an SDP file "
2454 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2455 NULL, 0, s->filename);
2457 snprintf(sdp, sizeof(sdp),
2458 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2459 addr.ss_family == AF_INET ? 4 : 6, host,
2460 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2461 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2462 port, payload_type);
2463 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2464 avcodec_parameters_free(&par);
2466 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2469 /* sdp_read_header initializes this again */
2472 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2474 ret = sdp_read_header(s);
2479 avcodec_parameters_free(&par);
2486 static const AVClass rtp_demuxer_class = {
2487 .class_name = "RTP demuxer",
2488 .item_name = av_default_item_name,
2489 .option = rtp_options,
2490 .version = LIBAVUTIL_VERSION_INT,
2493 AVInputFormat ff_rtp_demuxer = {
2495 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2496 .priv_data_size = sizeof(RTSPState),
2497 .read_probe = rtp_probe,
2498 .read_header = rtp_read_header,
2499 .read_packet = ff_rtsp_fetch_packet,
2500 .read_close = sdp_read_close,
2501 .flags = AVFMT_NOFILE,
2502 .priv_class = &rtp_demuxer_class,
2504 #endif /* CONFIG_RTP_DEMUXER */