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 av_parse_time(start, buf, 1);
180 get_word_sep(buf, sizeof(buf), "-", &p);
181 av_parse_time(end, buf, 1);
185 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
187 struct addrinfo hints = { 0 }, *ai = NULL;
188 hints.ai_flags = AI_NUMERICHOST;
189 if (getaddrinfo(buf, NULL, &hints, &ai))
191 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
197 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
198 RTSPStream *rtsp_st, AVStream *st)
200 AVCodecContext *codec = st ? st->codec : NULL;
204 codec->codec_id = handler->codec_id;
205 rtsp_st->dynamic_handler = handler;
207 st->need_parsing = handler->need_parsing;
208 if (handler->priv_data_size) {
209 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
210 if (!rtsp_st->dynamic_protocol_context)
211 rtsp_st->dynamic_handler = NULL;
215 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
218 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
219 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
220 rtsp_st->dynamic_protocol_context);
222 if (rtsp_st->dynamic_protocol_context) {
223 if (rtsp_st->dynamic_handler->close)
224 rtsp_st->dynamic_handler->close(
225 rtsp_st->dynamic_protocol_context);
226 av_free(rtsp_st->dynamic_protocol_context);
228 rtsp_st->dynamic_protocol_context = NULL;
229 rtsp_st->dynamic_handler = NULL;
234 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
235 static int sdp_parse_rtpmap(AVFormatContext *s,
236 AVStream *st, RTSPStream *rtsp_st,
237 int payload_type, const char *p)
239 AVCodecContext *codec = st->codec;
245 /* See if we can handle this kind of payload.
246 * The space should normally not be there but some Real streams or
247 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
248 * have a trailing space. */
249 get_word_sep(buf, sizeof(buf), "/ ", &p);
250 if (payload_type < RTP_PT_PRIVATE) {
251 /* We are in a standard case
252 * (from http://www.iana.org/assignments/rtp-parameters). */
253 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
256 if (codec->codec_id == AV_CODEC_ID_NONE) {
257 RTPDynamicProtocolHandler *handler =
258 ff_rtp_handler_find_by_name(buf, codec->codec_type);
259 init_rtp_handler(handler, rtsp_st, st);
260 /* If no dynamic handler was found, check with the list of standard
261 * allocated types, if such a stream for some reason happens to
262 * use a private payload type. This isn't handled in rtpdec.c, since
263 * the format name from the rtpmap line never is passed into rtpdec. */
264 if (!rtsp_st->dynamic_handler)
265 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
268 c = avcodec_find_decoder(codec->codec_id);
274 get_word_sep(buf, sizeof(buf), "/", &p);
276 switch (codec->codec_type) {
277 case AVMEDIA_TYPE_AUDIO:
278 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
279 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
280 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
282 codec->sample_rate = i;
283 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
284 get_word_sep(buf, sizeof(buf), "/", &p);
289 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
291 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
294 case AVMEDIA_TYPE_VIDEO:
295 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
297 avpriv_set_pts_info(st, 32, 1, i);
302 finalize_rtp_handler_init(s, rtsp_st, st);
306 /* parse the attribute line from the fmtp a line of an sdp response. This
307 * is broken out as a function because it is used in rtp_h264.c, which is
309 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
310 char *value, int value_size)
312 *p += strspn(*p, SPACE_CHARS);
314 get_word_sep(attr, attr_size, "=", p);
317 get_word_sep(value, value_size, ";", p);
325 typedef struct SDPParseState {
327 struct sockaddr_storage default_ip;
329 int skip_media; ///< set if an unknown m= line occurs
330 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
331 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
332 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
333 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
336 char delayed_fmtp[2048];
339 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
340 struct RTSPSource ***dest, int *dest_count)
342 RTSPSource *rtsp_src, *rtsp_src2;
344 for (i = 0; i < count; i++) {
346 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
349 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
350 dynarray_add(dest, dest_count, rtsp_src2);
354 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
355 int payload_type, const char *line)
359 for (i = 0; i < rt->nb_rtsp_streams; i++) {
360 RTSPStream *rtsp_st = rt->rtsp_streams[i];
361 if (rtsp_st->sdp_payload_type == payload_type &&
362 rtsp_st->dynamic_handler &&
363 rtsp_st->dynamic_handler->parse_sdp_a_line) {
364 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
365 rtsp_st->dynamic_protocol_context, line);
370 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
371 int letter, const char *buf)
373 RTSPState *rt = s->priv_data;
374 char buf1[64], st_type[64];
376 enum AVMediaType codec_type;
380 RTSPSource *rtsp_src;
381 struct sockaddr_storage sdp_ip;
384 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
387 if (s1->skip_media && letter != 'm')
391 get_word(buf1, sizeof(buf1), &p);
392 if (strcmp(buf1, "IN") != 0)
394 get_word(buf1, sizeof(buf1), &p);
395 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
397 get_word_sep(buf1, sizeof(buf1), "/", &p);
398 if (get_sockaddr(buf1, &sdp_ip))
403 get_word_sep(buf1, sizeof(buf1), "/", &p);
406 if (s->nb_streams == 0) {
407 s1->default_ip = sdp_ip;
408 s1->default_ttl = ttl;
410 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
411 rtsp_st->sdp_ip = sdp_ip;
412 rtsp_st->sdp_ttl = ttl;
416 av_dict_set(&s->metadata, "title", p, 0);
419 if (s->nb_streams == 0) {
420 av_dict_set(&s->metadata, "comment", p, 0);
429 codec_type = AVMEDIA_TYPE_UNKNOWN;
430 get_word(st_type, sizeof(st_type), &p);
431 if (!strcmp(st_type, "audio")) {
432 codec_type = AVMEDIA_TYPE_AUDIO;
433 } else if (!strcmp(st_type, "video")) {
434 codec_type = AVMEDIA_TYPE_VIDEO;
435 } else if (!strcmp(st_type, "application")) {
436 codec_type = AVMEDIA_TYPE_DATA;
437 } else if (!strcmp(st_type, "text")) {
438 codec_type = AVMEDIA_TYPE_SUBTITLE;
440 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
444 rtsp_st = av_mallocz(sizeof(RTSPStream));
447 rtsp_st->stream_index = -1;
448 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
450 rtsp_st->sdp_ip = s1->default_ip;
451 rtsp_st->sdp_ttl = s1->default_ttl;
453 copy_default_source_addrs(s1->default_include_source_addrs,
454 s1->nb_default_include_source_addrs,
455 &rtsp_st->include_source_addrs,
456 &rtsp_st->nb_include_source_addrs);
457 copy_default_source_addrs(s1->default_exclude_source_addrs,
458 s1->nb_default_exclude_source_addrs,
459 &rtsp_st->exclude_source_addrs,
460 &rtsp_st->nb_exclude_source_addrs);
462 get_word(buf1, sizeof(buf1), &p); /* port */
463 rtsp_st->sdp_port = atoi(buf1);
465 get_word(buf1, sizeof(buf1), &p); /* protocol */
466 if (!strcmp(buf1, "udp"))
467 rt->transport = RTSP_TRANSPORT_RAW;
468 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
469 rtsp_st->feedback = 1;
471 /* XXX: handle list of formats */
472 get_word(buf1, sizeof(buf1), &p); /* format list */
473 rtsp_st->sdp_payload_type = atoi(buf1);
475 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
476 /* no corresponding stream */
477 if (rt->transport == RTSP_TRANSPORT_RAW) {
478 if (CONFIG_RTPDEC && !rt->ts)
479 rt->ts = avpriv_mpegts_parse_open(s);
481 RTPDynamicProtocolHandler *handler;
482 handler = ff_rtp_handler_find_by_id(
483 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
484 init_rtp_handler(handler, rtsp_st, NULL);
485 finalize_rtp_handler_init(s, rtsp_st, NULL);
487 } else if (rt->server_type == RTSP_SERVER_WMS &&
488 codec_type == AVMEDIA_TYPE_DATA) {
489 /* RTX stream, a stream that carries all the other actual
490 * audio/video streams. Don't expose this to the callers. */
492 st = avformat_new_stream(s, NULL);
495 st->id = rt->nb_rtsp_streams - 1;
496 rtsp_st->stream_index = st->index;
497 st->codec->codec_type = codec_type;
498 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
499 RTPDynamicProtocolHandler *handler;
500 /* if standard payload type, we can find the codec right now */
501 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
502 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
503 st->codec->sample_rate > 0)
504 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
505 /* Even static payload types may need a custom depacketizer */
506 handler = ff_rtp_handler_find_by_id(
507 rtsp_st->sdp_payload_type, st->codec->codec_type);
508 init_rtp_handler(handler, rtsp_st, st);
509 finalize_rtp_handler_init(s, rtsp_st, st);
511 if (rt->default_lang[0])
512 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
514 /* put a default control url */
515 av_strlcpy(rtsp_st->control_url, rt->control_uri,
516 sizeof(rtsp_st->control_url));
519 if (av_strstart(p, "control:", &p)) {
520 if (s->nb_streams == 0) {
521 if (!strncmp(p, "rtsp://", 7))
522 av_strlcpy(rt->control_uri, p,
523 sizeof(rt->control_uri));
526 /* get the control url */
527 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
529 /* XXX: may need to add full url resolution */
530 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
532 if (proto[0] == '\0') {
533 /* relative control URL */
534 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
535 av_strlcat(rtsp_st->control_url, "/",
536 sizeof(rtsp_st->control_url));
537 av_strlcat(rtsp_st->control_url, p,
538 sizeof(rtsp_st->control_url));
540 av_strlcpy(rtsp_st->control_url, p,
541 sizeof(rtsp_st->control_url));
543 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
544 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
545 get_word(buf1, sizeof(buf1), &p);
546 payload_type = atoi(buf1);
547 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
548 if (rtsp_st->stream_index >= 0) {
549 st = s->streams[rtsp_st->stream_index];
550 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
554 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
556 } else if (av_strstart(p, "fmtp:", &p) ||
557 av_strstart(p, "framesize:", &p)) {
558 // let dynamic protocol handlers have a stab at the line.
559 get_word(buf1, sizeof(buf1), &p);
560 payload_type = atoi(buf1);
561 if (s1->seen_rtpmap) {
562 parse_fmtp(s, rt, payload_type, buf);
565 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
567 } else if (av_strstart(p, "range:", &p)) {
570 // this is so that seeking on a streamed file can work.
571 rtsp_parse_range_npt(p, &start, &end);
572 s->start_time = start;
573 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
574 s->duration = (end == AV_NOPTS_VALUE) ?
575 AV_NOPTS_VALUE : end - start;
576 } else if (av_strstart(p, "lang:", &p)) {
577 if (s->nb_streams > 0) {
578 get_word(buf1, sizeof(buf1), &p);
579 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
580 if (rtsp_st->stream_index >= 0) {
581 st = s->streams[rtsp_st->stream_index];
582 av_dict_set(&st->metadata, "language", buf1, 0);
585 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
586 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
588 rt->transport = RTSP_TRANSPORT_RDT;
589 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
591 st = s->streams[s->nb_streams - 1];
592 st->codec->sample_rate = atoi(p);
593 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
595 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
596 get_word(buf1, sizeof(buf1), &p); // ignore tag
597 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
598 p += strspn(p, SPACE_CHARS);
599 if (av_strstart(p, "inline:", &p))
600 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
601 } else if (av_strstart(p, "source-filter:", &p)) {
603 get_word(buf1, sizeof(buf1), &p);
604 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
606 exclude = !strcmp(buf1, "excl");
608 get_word(buf1, sizeof(buf1), &p);
609 if (strcmp(buf1, "IN") != 0)
611 get_word(buf1, sizeof(buf1), &p);
612 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
614 // not checking that the destination address actually matches or is wildcard
615 get_word(buf1, sizeof(buf1), &p);
618 rtsp_src = av_mallocz(sizeof(*rtsp_src));
621 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
623 if (s->nb_streams == 0) {
624 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
626 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
627 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
630 if (s->nb_streams == 0) {
631 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
633 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
634 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
639 if (rt->server_type == RTSP_SERVER_WMS)
640 ff_wms_parse_sdp_a_line(s, p);
641 if (s->nb_streams > 0) {
642 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
644 if (rt->server_type == RTSP_SERVER_REAL)
645 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
647 if (rtsp_st->dynamic_handler &&
648 rtsp_st->dynamic_handler->parse_sdp_a_line)
649 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
650 rtsp_st->stream_index,
651 rtsp_st->dynamic_protocol_context, buf);
658 int ff_sdp_parse(AVFormatContext *s, const char *content)
660 RTSPState *rt = s->priv_data;
663 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
664 * contain long SDP lines containing complete ASF Headers (several
665 * kB) or arrays of MDPR (RM stream descriptor) headers plus
666 * "rulebooks" describing their properties. Therefore, the SDP line
669 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
670 * in rtpdec_xiph.c. */
672 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
676 p += strspn(p, SPACE_CHARS);
684 /* get the content */
686 while (*p != '\n' && *p != '\r' && *p != '\0') {
687 if ((q - buf) < sizeof(buf) - 1)
692 sdp_parse_line(s, s1, letter, buf);
694 while (*p != '\n' && *p != '\0')
700 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
701 av_freep(&s1->default_include_source_addrs[i]);
702 av_freep(&s1->default_include_source_addrs);
703 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
704 av_freep(&s1->default_exclude_source_addrs[i]);
705 av_freep(&s1->default_exclude_source_addrs);
707 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
708 if (!rt->p) return AVERROR(ENOMEM);
711 #endif /* CONFIG_RTPDEC */
713 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
715 RTSPState *rt = s->priv_data;
718 for (i = 0; i < rt->nb_rtsp_streams; i++) {
719 RTSPStream *rtsp_st = rt->rtsp_streams[i];
722 if (rtsp_st->transport_priv) {
724 AVFormatContext *rtpctx = rtsp_st->transport_priv;
725 av_write_trailer(rtpctx);
726 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
727 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
728 ff_rtsp_tcp_write_packet(s, rtsp_st);
729 ffio_free_dyn_buf(&rtpctx->pb);
731 avio_closep(&rtpctx->pb);
733 avformat_free_context(rtpctx);
734 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
735 ff_rdt_parse_close(rtsp_st->transport_priv);
736 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
737 ff_rtp_parse_close(rtsp_st->transport_priv);
739 rtsp_st->transport_priv = NULL;
740 if (rtsp_st->rtp_handle)
741 ffurl_close(rtsp_st->rtp_handle);
742 rtsp_st->rtp_handle = NULL;
746 /* close and free RTSP streams */
747 void ff_rtsp_close_streams(AVFormatContext *s)
749 RTSPState *rt = s->priv_data;
753 ff_rtsp_undo_setup(s, 0);
754 for (i = 0; i < rt->nb_rtsp_streams; i++) {
755 rtsp_st = rt->rtsp_streams[i];
757 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
758 if (rtsp_st->dynamic_handler->close)
759 rtsp_st->dynamic_handler->close(
760 rtsp_st->dynamic_protocol_context);
761 av_free(rtsp_st->dynamic_protocol_context);
763 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
764 av_freep(&rtsp_st->include_source_addrs[j]);
765 av_freep(&rtsp_st->include_source_addrs);
766 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
767 av_freep(&rtsp_st->exclude_source_addrs[j]);
768 av_freep(&rtsp_st->exclude_source_addrs);
773 av_freep(&rt->rtsp_streams);
775 avformat_close_input(&rt->asf_ctx);
777 if (CONFIG_RTPDEC && rt->ts)
778 avpriv_mpegts_parse_close(rt->ts);
780 av_freep(&rt->recvbuf);
783 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
785 RTSPState *rt = s->priv_data;
787 int reordering_queue_size = rt->reordering_queue_size;
788 if (reordering_queue_size < 0) {
789 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
790 reordering_queue_size = 0;
792 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
795 /* open the RTP context */
796 if (rtsp_st->stream_index >= 0)
797 st = s->streams[rtsp_st->stream_index];
799 s->ctx_flags |= AVFMTCTX_NOHEADER;
801 if (CONFIG_RTSP_MUXER && s->oformat) {
802 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
803 s, st, rtsp_st->rtp_handle,
804 RTSP_TCP_MAX_PACKET_SIZE,
805 rtsp_st->stream_index);
806 /* Ownership of rtp_handle is passed to the rtp mux context */
807 rtsp_st->rtp_handle = NULL;
810 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
811 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
812 return 0; // Don't need to open any parser here
813 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
814 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
815 rtsp_st->dynamic_protocol_context,
816 rtsp_st->dynamic_handler);
817 else if (CONFIG_RTPDEC)
818 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
819 rtsp_st->sdp_payload_type,
820 reordering_queue_size);
822 if (!rtsp_st->transport_priv) {
823 return AVERROR(ENOMEM);
824 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
825 if (rtsp_st->dynamic_handler) {
826 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
827 rtsp_st->dynamic_protocol_context,
828 rtsp_st->dynamic_handler);
830 if (rtsp_st->crypto_suite[0])
831 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
832 rtsp_st->crypto_suite,
833 rtsp_st->crypto_params);
839 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
840 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
847 q += strspn(q, SPACE_CHARS);
848 v = strtol(q, &p, 10);
852 v = strtol(p, &p, 10);
861 /* XXX: only one transport specification is parsed */
862 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
864 char transport_protocol[16];
866 char lower_transport[16];
868 RTSPTransportField *th;
871 reply->nb_transports = 0;
874 p += strspn(p, SPACE_CHARS);
878 th = &reply->transports[reply->nb_transports];
880 get_word_sep(transport_protocol, sizeof(transport_protocol),
882 if (!av_strcasecmp (transport_protocol, "rtp")) {
883 get_word_sep(profile, sizeof(profile), "/;,", &p);
884 lower_transport[0] = '\0';
885 /* rtp/avp/<protocol> */
887 get_word_sep(lower_transport, sizeof(lower_transport),
890 th->transport = RTSP_TRANSPORT_RTP;
891 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
892 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
893 /* x-pn-tng/<protocol> */
894 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
896 th->transport = RTSP_TRANSPORT_RDT;
897 } else if (!av_strcasecmp(transport_protocol, "raw")) {
898 get_word_sep(profile, sizeof(profile), "/;,", &p);
899 lower_transport[0] = '\0';
900 /* raw/raw/<protocol> */
902 get_word_sep(lower_transport, sizeof(lower_transport),
905 th->transport = RTSP_TRANSPORT_RAW;
907 if (!av_strcasecmp(lower_transport, "TCP"))
908 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
910 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
914 /* get each parameter */
915 while (*p != '\0' && *p != ',') {
916 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
917 if (!strcmp(parameter, "port")) {
920 rtsp_parse_range(&th->port_min, &th->port_max, &p);
922 } else if (!strcmp(parameter, "client_port")) {
925 rtsp_parse_range(&th->client_port_min,
926 &th->client_port_max, &p);
928 } else if (!strcmp(parameter, "server_port")) {
931 rtsp_parse_range(&th->server_port_min,
932 &th->server_port_max, &p);
934 } else if (!strcmp(parameter, "interleaved")) {
937 rtsp_parse_range(&th->interleaved_min,
938 &th->interleaved_max, &p);
940 } else if (!strcmp(parameter, "multicast")) {
941 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
942 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
943 } else if (!strcmp(parameter, "ttl")) {
947 th->ttl = strtol(p, &end, 10);
950 } else if (!strcmp(parameter, "destination")) {
953 get_word_sep(buf, sizeof(buf), ";,", &p);
954 get_sockaddr(buf, &th->destination);
956 } else if (!strcmp(parameter, "source")) {
959 get_word_sep(buf, sizeof(buf), ";,", &p);
960 av_strlcpy(th->source, buf, sizeof(th->source));
962 } else if (!strcmp(parameter, "mode")) {
965 get_word_sep(buf, sizeof(buf), ";, ", &p);
966 if (!strcmp(buf, "record") ||
967 !strcmp(buf, "receive"))
972 while (*p != ';' && *p != '\0' && *p != ',')
980 reply->nb_transports++;
984 static void handle_rtp_info(RTSPState *rt, const char *url,
985 uint32_t seq, uint32_t rtptime)
988 if (!rtptime || !url[0])
990 if (rt->transport != RTSP_TRANSPORT_RTP)
992 for (i = 0; i < rt->nb_rtsp_streams; i++) {
993 RTSPStream *rtsp_st = rt->rtsp_streams[i];
994 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
997 if (!strcmp(rtsp_st->control_url, url)) {
998 rtpctx->base_timestamp = rtptime;
1004 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1007 char key[20], value[1024], url[1024] = "";
1008 uint32_t seq = 0, rtptime = 0;
1011 p += strspn(p, SPACE_CHARS);
1014 get_word_sep(key, sizeof(key), "=", &p);
1018 get_word_sep(value, sizeof(value), ";, ", &p);
1020 if (!strcmp(key, "url"))
1021 av_strlcpy(url, value, sizeof(url));
1022 else if (!strcmp(key, "seq"))
1023 seq = strtoul(value, NULL, 10);
1024 else if (!strcmp(key, "rtptime"))
1025 rtptime = strtoul(value, NULL, 10);
1027 handle_rtp_info(rt, url, seq, rtptime);
1036 handle_rtp_info(rt, url, seq, rtptime);
1039 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1040 RTSPState *rt, const char *method)
1044 /* NOTE: we do case independent match for broken servers */
1046 if (av_stristart(p, "Session:", &p)) {
1048 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1049 if (av_stristart(p, ";timeout=", &p) &&
1050 (t = strtol(p, NULL, 10)) > 0) {
1053 } else if (av_stristart(p, "Content-Length:", &p)) {
1054 reply->content_length = strtol(p, NULL, 10);
1055 } else if (av_stristart(p, "Transport:", &p)) {
1056 rtsp_parse_transport(reply, p);
1057 } else if (av_stristart(p, "CSeq:", &p)) {
1058 reply->seq = strtol(p, NULL, 10);
1059 } else if (av_stristart(p, "Range:", &p)) {
1060 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1061 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1062 p += strspn(p, SPACE_CHARS);
1063 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1064 } else if (av_stristart(p, "Server:", &p)) {
1065 p += strspn(p, SPACE_CHARS);
1066 av_strlcpy(reply->server, p, sizeof(reply->server));
1067 } else if (av_stristart(p, "Notice:", &p) ||
1068 av_stristart(p, "X-Notice:", &p)) {
1069 reply->notice = strtol(p, NULL, 10);
1070 } else if (av_stristart(p, "Location:", &p)) {
1071 p += strspn(p, SPACE_CHARS);
1072 av_strlcpy(reply->location, p , sizeof(reply->location));
1073 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1074 p += strspn(p, SPACE_CHARS);
1075 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1076 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1077 p += strspn(p, SPACE_CHARS);
1078 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1079 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1080 p += strspn(p, SPACE_CHARS);
1081 if (method && !strcmp(method, "DESCRIBE"))
1082 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1083 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1084 p += strspn(p, SPACE_CHARS);
1085 if (method && !strcmp(method, "PLAY"))
1086 rtsp_parse_rtp_info(rt, p);
1087 } else if (av_stristart(p, "Public:", &p) && rt) {
1088 if (strstr(p, "GET_PARAMETER") &&
1089 method && !strcmp(method, "OPTIONS"))
1090 rt->get_parameter_supported = 1;
1091 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1092 p += strspn(p, SPACE_CHARS);
1093 rt->accept_dynamic_rate = atoi(p);
1094 } else if (av_stristart(p, "Content-Type:", &p)) {
1095 p += strspn(p, SPACE_CHARS);
1096 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1100 /* skip a RTP/TCP interleaved packet */
1101 void ff_rtsp_skip_packet(AVFormatContext *s)
1103 RTSPState *rt = s->priv_data;
1107 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1110 len = AV_RB16(buf + 1);
1112 av_dlog(s, "skipping RTP packet len=%d\n", len);
1117 if (len1 > sizeof(buf))
1119 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1126 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1127 unsigned char **content_ptr,
1128 int return_on_interleaved_data, const char *method)
1130 RTSPState *rt = s->priv_data;
1131 char buf[4096], buf1[1024], *q;
1134 int ret, content_length, line_count = 0, request = 0;
1135 unsigned char *content = NULL;
1141 memset(reply, 0, sizeof(*reply));
1143 /* parse reply (XXX: use buffers) */
1144 rt->last_reply[0] = '\0';
1148 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1149 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1155 /* XXX: only parse it if first char on line ? */
1156 if (return_on_interleaved_data) {
1159 ff_rtsp_skip_packet(s);
1160 } else if (ch != '\r') {
1161 if ((q - buf) < sizeof(buf) - 1)
1167 av_dlog(s, "line='%s'\n", buf);
1169 /* test if last line */
1173 if (line_count == 0) {
1174 /* get reply code */
1175 get_word(buf1, sizeof(buf1), &p);
1176 if (!strncmp(buf1, "RTSP/", 5)) {
1177 get_word(buf1, sizeof(buf1), &p);
1178 reply->status_code = atoi(buf1);
1179 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1181 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1182 get_word(buf1, sizeof(buf1), &p); // object
1186 ff_rtsp_parse_line(reply, p, rt, method);
1187 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1188 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1193 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1194 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1196 content_length = reply->content_length;
1197 if (content_length > 0) {
1198 /* leave some room for a trailing '\0' (useful for simple parsing) */
1199 content = av_malloc(content_length + 1);
1201 return AVERROR(ENOMEM);
1202 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1203 content[content_length] = '\0';
1206 *content_ptr = content;
1212 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1213 const char* ptr = buf;
1215 if (!strcmp(reply->reason, "OPTIONS")) {
1216 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1218 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1219 if (reply->session_id[0])
1220 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1223 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1225 av_strlcat(buf, "\r\n", sizeof(buf));
1227 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1228 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1231 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1233 rt->last_cmd_time = av_gettime_relative();
1234 /* Even if the request from the server had data, it is not the data
1235 * that the caller wants or expects. The memory could also be leaked
1236 * if the actual following reply has content data. */
1238 av_freep(content_ptr);
1239 /* If method is set, this is called from ff_rtsp_send_cmd,
1240 * where a reply to exactly this request is awaited. For
1241 * callers from within packet receiving, we just want to
1242 * return to the caller and go back to receiving packets. */
1248 if (rt->seq != reply->seq) {
1249 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1250 rt->seq, reply->seq);
1254 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1255 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1256 reply->notice == 2306 /* Continuous Feed Terminated */) {
1257 rt->state = RTSP_STATE_IDLE;
1258 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1259 return AVERROR(EIO); /* data or server error */
1260 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1261 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1262 return AVERROR(EPERM);
1268 * Send a command to the RTSP server without waiting for the reply.
1270 * @param s RTSP (de)muxer context
1271 * @param method the method for the request
1272 * @param url the target url for the request
1273 * @param headers extra header lines to include in the request
1274 * @param send_content if non-null, the data to send as request body content
1275 * @param send_content_length the length of the send_content data, or 0 if
1276 * send_content is null
1278 * @return zero if success, nonzero otherwise
1280 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1281 const char *method, const char *url,
1282 const char *headers,
1283 const unsigned char *send_content,
1284 int send_content_length)
1286 RTSPState *rt = s->priv_data;
1287 char buf[4096], *out_buf;
1288 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1290 /* Add in RTSP headers */
1293 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1295 av_strlcat(buf, headers, sizeof(buf));
1296 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1297 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1298 if (rt->session_id[0] != '\0' && (!headers ||
1299 !strstr(headers, "\nIf-Match:"))) {
1300 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1303 char *str = ff_http_auth_create_response(&rt->auth_state,
1304 rt->auth, url, method);
1306 av_strlcat(buf, str, sizeof(buf));
1309 if (send_content_length > 0 && send_content)
1310 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1311 av_strlcat(buf, "\r\n", sizeof(buf));
1313 /* base64 encode rtsp if tunneling */
1314 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1315 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1316 out_buf = base64buf;
1319 av_dlog(s, "Sending:\n%s--\n", buf);
1321 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1322 if (send_content_length > 0 && send_content) {
1323 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1324 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1325 "with content data not supported\n");
1326 return AVERROR_PATCHWELCOME;
1328 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1330 rt->last_cmd_time = av_gettime_relative();
1335 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1336 const char *url, const char *headers)
1338 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1341 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1342 const char *headers, RTSPMessageHeader *reply,
1343 unsigned char **content_ptr)
1345 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1346 content_ptr, NULL, 0);
1349 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1350 const char *method, const char *url,
1352 RTSPMessageHeader *reply,
1353 unsigned char **content_ptr,
1354 const unsigned char *send_content,
1355 int send_content_length)
1357 RTSPState *rt = s->priv_data;
1358 HTTPAuthType cur_auth_type;
1359 int ret, attempts = 0;
1362 cur_auth_type = rt->auth_state.auth_type;
1363 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1365 send_content_length)))
1368 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1372 if (reply->status_code == 401 &&
1373 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1374 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1377 if (reply->status_code > 400){
1378 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1382 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1388 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1389 int lower_transport, const char *real_challenge)
1391 RTSPState *rt = s->priv_data;
1392 int rtx = 0, j, i, err, interleave = 0, port_off;
1393 RTSPStream *rtsp_st;
1394 RTSPMessageHeader reply1, *reply = &reply1;
1396 const char *trans_pref;
1398 if (rt->transport == RTSP_TRANSPORT_RDT)
1399 trans_pref = "x-pn-tng";
1400 else if (rt->transport == RTSP_TRANSPORT_RAW)
1401 trans_pref = "RAW/RAW";
1403 trans_pref = "RTP/AVP";
1405 /* default timeout: 1 minute */
1408 /* Choose a random starting offset within the first half of the
1409 * port range, to allow for a number of ports to try even if the offset
1410 * happens to be at the end of the random range. */
1411 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1412 /* even random offset */
1413 port_off -= port_off & 0x01;
1415 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1416 char transport[2048];
1419 * WMS serves all UDP data over a single connection, the RTX, which
1420 * isn't necessarily the first in the SDP but has to be the first
1421 * to be set up, else the second/third SETUP will fail with a 461.
1423 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1424 rt->server_type == RTSP_SERVER_WMS) {
1427 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1428 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1430 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1434 if (rtx == rt->nb_rtsp_streams)
1435 return -1; /* no RTX found */
1436 rtsp_st = rt->rtsp_streams[rtx];
1438 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1440 rtsp_st = rt->rtsp_streams[i];
1443 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1446 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1447 port = reply->transports[0].client_port_min;
1451 /* first try in specified port range */
1452 while (j <= rt->rtp_port_max) {
1453 AVDictionary *opts = map_to_opts(rt);
1455 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1456 "?localport=%d", j);
1457 /* we will use two ports per rtp stream (rtp and rtcp) */
1459 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1460 &s->interrupt_callback, &opts);
1462 av_dict_free(&opts);
1467 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1472 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1474 snprintf(transport, sizeof(transport) - 1,
1475 "%s/UDP;", trans_pref);
1476 if (rt->server_type != RTSP_SERVER_REAL)
1477 av_strlcat(transport, "unicast;", sizeof(transport));
1478 av_strlcatf(transport, sizeof(transport),
1479 "client_port=%d", port);
1480 if (rt->transport == RTSP_TRANSPORT_RTP &&
1481 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1482 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1486 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1487 /* For WMS streams, the application streams are only used for
1488 * UDP. When trying to set it up for TCP streams, the server
1489 * will return an error. Therefore, we skip those streams. */
1490 if (rt->server_type == RTSP_SERVER_WMS &&
1491 (rtsp_st->stream_index < 0 ||
1492 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1495 snprintf(transport, sizeof(transport) - 1,
1496 "%s/TCP;", trans_pref);
1497 if (rt->transport != RTSP_TRANSPORT_RDT)
1498 av_strlcat(transport, "unicast;", sizeof(transport));
1499 av_strlcatf(transport, sizeof(transport),
1500 "interleaved=%d-%d",
1501 interleave, interleave + 1);
1505 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1506 snprintf(transport, sizeof(transport) - 1,
1507 "%s/UDP;multicast", trans_pref);
1510 av_strlcat(transport, ";mode=record", sizeof(transport));
1511 } else if (rt->server_type == RTSP_SERVER_REAL ||
1512 rt->server_type == RTSP_SERVER_WMS)
1513 av_strlcat(transport, ";mode=play", sizeof(transport));
1514 snprintf(cmd, sizeof(cmd),
1515 "Transport: %s\r\n",
1517 if (rt->accept_dynamic_rate)
1518 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1519 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1520 char real_res[41], real_csum[9];
1521 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1523 av_strlcatf(cmd, sizeof(cmd),
1525 "RealChallenge2: %s, sd=%s\r\n",
1526 rt->session_id, real_res, real_csum);
1528 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1529 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1532 } else if (reply->status_code != RTSP_STATUS_OK ||
1533 reply->nb_transports != 1) {
1534 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1538 /* XXX: same protocol for all streams is required */
1540 if (reply->transports[0].lower_transport != rt->lower_transport ||
1541 reply->transports[0].transport != rt->transport) {
1542 err = AVERROR_INVALIDDATA;
1546 rt->lower_transport = reply->transports[0].lower_transport;
1547 rt->transport = reply->transports[0].transport;
1550 /* Fail if the server responded with another lower transport mode
1551 * than what we requested. */
1552 if (reply->transports[0].lower_transport != lower_transport) {
1553 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1554 err = AVERROR_INVALIDDATA;
1558 switch(reply->transports[0].lower_transport) {
1559 case RTSP_LOWER_TRANSPORT_TCP:
1560 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1561 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1564 case RTSP_LOWER_TRANSPORT_UDP: {
1565 char url[1024], options[30] = "";
1566 const char *peer = host;
1568 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1569 av_strlcpy(options, "?connect=1", sizeof(options));
1570 /* Use source address if specified */
1571 if (reply->transports[0].source[0])
1572 peer = reply->transports[0].source;
1573 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1574 reply->transports[0].server_port_min, "%s", options);
1575 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1576 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1577 err = AVERROR_INVALIDDATA;
1582 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1583 char url[1024], namebuf[50], optbuf[20] = "";
1584 struct sockaddr_storage addr;
1587 if (reply->transports[0].destination.ss_family) {
1588 addr = reply->transports[0].destination;
1589 port = reply->transports[0].port_min;
1590 ttl = reply->transports[0].ttl;
1592 addr = rtsp_st->sdp_ip;
1593 port = rtsp_st->sdp_port;
1594 ttl = rtsp_st->sdp_ttl;
1597 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1598 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1599 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1600 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1601 port, "%s", optbuf);
1602 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1603 &s->interrupt_callback, NULL) < 0) {
1604 err = AVERROR_INVALIDDATA;
1611 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1615 if (rt->nb_rtsp_streams && reply->timeout > 0)
1616 rt->timeout = reply->timeout;
1618 if (rt->server_type == RTSP_SERVER_REAL)
1619 rt->need_subscription = 1;
1624 ff_rtsp_undo_setup(s, 0);
1628 void ff_rtsp_close_connections(AVFormatContext *s)
1630 RTSPState *rt = s->priv_data;
1631 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1632 ffurl_close(rt->rtsp_hd);
1633 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1636 int ff_rtsp_connect(AVFormatContext *s)
1638 RTSPState *rt = s->priv_data;
1639 char proto[128], host[1024], path[1024];
1640 char tcpname[1024], cmd[2048], auth[128];
1641 const char *lower_rtsp_proto = "tcp";
1642 int port, err, tcp_fd;
1643 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1644 int lower_transport_mask = 0;
1645 int default_port = RTSP_DEFAULT_PORT;
1646 char real_challenge[64] = "";
1647 struct sockaddr_storage peer;
1648 socklen_t peer_len = sizeof(peer);
1650 if (rt->rtp_port_max < rt->rtp_port_min) {
1651 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1652 "than min port %d\n", rt->rtp_port_max,
1654 return AVERROR(EINVAL);
1657 if (!ff_network_init())
1658 return AVERROR(EIO);
1660 if (s->max_delay < 0) /* Not set by the caller */
1661 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1663 rt->control_transport = RTSP_MODE_PLAIN;
1664 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1665 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1666 rt->control_transport = RTSP_MODE_TUNNEL;
1668 /* Only pass through valid flags from here */
1669 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1672 /* extract hostname and port */
1673 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1674 host, sizeof(host), &port, path, sizeof(path), s->filename);
1676 if (!strcmp(proto, "rtsps")) {
1677 lower_rtsp_proto = "tls";
1678 default_port = RTSPS_DEFAULT_PORT;
1679 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1683 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1686 port = default_port;
1688 lower_transport_mask = rt->lower_transport_mask;
1690 if (!lower_transport_mask)
1691 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1694 /* Only UDP or TCP - UDP multicast isn't supported. */
1695 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1696 (1 << RTSP_LOWER_TRANSPORT_TCP);
1697 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1698 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1699 "only UDP and TCP are supported for output.\n");
1700 err = AVERROR(EINVAL);
1705 /* Construct the URI used in request; this is similar to s->filename,
1706 * but with authentication credentials removed and RTSP specific options
1708 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1709 host, port, "%s", path);
1711 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1712 /* set up initial handshake for tunneling */
1713 char httpname[1024];
1714 char sessioncookie[17];
1717 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1718 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1719 av_get_random_seed(), av_get_random_seed());
1722 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1723 &s->interrupt_callback) < 0) {
1728 /* generate GET headers */
1729 snprintf(headers, sizeof(headers),
1730 "x-sessioncookie: %s\r\n"
1731 "Accept: application/x-rtsp-tunnelled\r\n"
1732 "Pragma: no-cache\r\n"
1733 "Cache-Control: no-cache\r\n",
1735 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1737 /* complete the connection */
1738 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1744 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1745 &s->interrupt_callback) < 0 ) {
1750 /* generate POST headers */
1751 snprintf(headers, sizeof(headers),
1752 "x-sessioncookie: %s\r\n"
1753 "Content-Type: application/x-rtsp-tunnelled\r\n"
1754 "Pragma: no-cache\r\n"
1755 "Cache-Control: no-cache\r\n"
1756 "Content-Length: 32767\r\n"
1757 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1759 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1760 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1762 /* Initialize the authentication state for the POST session. The HTTP
1763 * protocol implementation doesn't properly handle multi-pass
1764 * authentication for POST requests, since it would require one of
1766 * - implementing Expect: 100-continue, which many HTTP servers
1767 * don't support anyway, even less the RTSP servers that do HTTP
1769 * - sending the whole POST data until getting a 401 reply specifying
1770 * what authentication method to use, then resending all that data
1771 * - waiting for potential 401 replies directly after sending the
1772 * POST header (waiting for some unspecified time)
1773 * Therefore, we copy the full auth state, which works for both basic
1774 * and digest. (For digest, we would have to synchronize the nonce
1775 * count variable between the two sessions, if we'd do more requests
1776 * with the original session, though.)
1778 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1780 /* complete the connection */
1781 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1787 /* open the tcp connection */
1788 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1790 "?timeout=%d", rt->stimeout);
1791 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1792 &s->interrupt_callback, NULL)) < 0) {
1796 rt->rtsp_hd_out = rt->rtsp_hd;
1800 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1805 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1806 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1807 NULL, 0, NI_NUMERICHOST);
1810 /* request options supported by the server; this also detects server
1812 for (rt->server_type = RTSP_SERVER_RTP;;) {
1814 if (rt->server_type == RTSP_SERVER_REAL)
1817 * The following entries are required for proper
1818 * streaming from a Realmedia server. They are
1819 * interdependent in some way although we currently
1820 * don't quite understand how. Values were copied
1821 * from mplayer SVN r23589.
1822 * ClientChallenge is a 16-byte ID in hex
1823 * CompanyID is a 16-byte ID in base64
1825 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1826 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1827 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1828 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1830 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1831 if (reply->status_code != RTSP_STATUS_OK) {
1832 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1836 /* detect server type if not standard-compliant RTP */
1837 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1838 rt->server_type = RTSP_SERVER_REAL;
1840 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1841 rt->server_type = RTSP_SERVER_WMS;
1842 } else if (rt->server_type == RTSP_SERVER_REAL)
1843 strcpy(real_challenge, reply->real_challenge);
1847 if (CONFIG_RTSP_DEMUXER && s->iformat)
1848 err = ff_rtsp_setup_input_streams(s, reply);
1849 else if (CONFIG_RTSP_MUXER)
1850 err = ff_rtsp_setup_output_streams(s, host);
1857 int lower_transport = ff_log2_tab[lower_transport_mask &
1858 ~(lower_transport_mask - 1)];
1860 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1861 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1862 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1864 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1865 rt->server_type == RTSP_SERVER_REAL ?
1866 real_challenge : NULL);
1869 lower_transport_mask &= ~(1 << lower_transport);
1870 if (lower_transport_mask == 0 && err == 1) {
1871 err = AVERROR(EPROTONOSUPPORT);
1876 rt->lower_transport_mask = lower_transport_mask;
1877 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1878 rt->state = RTSP_STATE_IDLE;
1879 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1882 ff_rtsp_close_streams(s);
1883 ff_rtsp_close_connections(s);
1884 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1885 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1886 rt->session_id[0] = '\0';
1887 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1895 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1898 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1899 uint8_t *buf, int buf_size, int64_t wait_end)
1901 RTSPState *rt = s->priv_data;
1902 RTSPStream *rtsp_st;
1903 int n, i, ret, tcp_fd, timeout_cnt = 0;
1905 struct pollfd *p = rt->p;
1906 int *fds = NULL, fdsnum, fdsidx;
1909 if (ff_check_interrupt(&s->interrupt_callback))
1910 return AVERROR_EXIT;
1911 if (wait_end && wait_end - av_gettime_relative() < 0)
1912 return AVERROR(EAGAIN);
1915 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1916 p[max_p].fd = tcp_fd;
1917 p[max_p++].events = POLLIN;
1921 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1922 rtsp_st = rt->rtsp_streams[i];
1923 if (rtsp_st->rtp_handle) {
1924 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1926 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1930 av_log(s, AV_LOG_ERROR,
1931 "Number of fds %d not supported\n", fdsnum);
1932 return AVERROR_INVALIDDATA;
1934 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1935 p[max_p].fd = fds[fdsidx];
1936 p[max_p++].events = POLLIN;
1941 n = poll(p, max_p, POLL_TIMEOUT_MS);
1943 int j = 1 - (tcp_fd == -1);
1945 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1946 rtsp_st = rt->rtsp_streams[i];
1947 if (rtsp_st->rtp_handle) {
1948 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1949 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1951 *prtsp_st = rtsp_st;
1958 #if CONFIG_RTSP_DEMUXER
1959 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1960 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1961 if (rt->state == RTSP_STATE_STREAMING) {
1962 if (!ff_rtsp_parse_streaming_commands(s))
1965 av_log(s, AV_LOG_WARNING,
1966 "Unable to answer to TEARDOWN\n");
1970 RTSPMessageHeader reply;
1971 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1974 /* XXX: parse message */
1975 if (rt->state != RTSP_STATE_STREAMING)
1980 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1981 return AVERROR(ETIMEDOUT);
1982 } else if (n < 0 && errno != EINTR)
1983 return AVERROR(errno);
1987 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1988 const uint8_t *buf, int len)
1990 RTSPState *rt = s->priv_data;
1994 if (rt->nb_rtsp_streams == 1) {
1995 *rtsp_st = rt->rtsp_streams[0];
1998 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1999 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2001 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2002 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2005 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2006 *rtsp_st = rt->rtsp_streams[i];
2013 av_log(s, AV_LOG_WARNING,
2014 "Unable to pick stream for packet - SSRC not known for "
2016 return AVERROR(EAGAIN);
2019 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2020 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2021 *rtsp_st = rt->rtsp_streams[i];
2027 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2028 return AVERROR(EAGAIN);
2031 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2033 RTSPState *rt = s->priv_data;
2035 RTSPStream *rtsp_st, *first_queue_st = NULL;
2036 int64_t wait_end = 0;
2038 if (rt->nb_byes == rt->nb_rtsp_streams)
2041 /* get next frames from the same RTP packet */
2042 if (rt->cur_transport_priv) {
2043 if (rt->transport == RTSP_TRANSPORT_RDT) {
2044 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2045 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2046 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2047 } else if (CONFIG_RTPDEC && rt->ts) {
2048 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2050 rt->recvbuf_pos += ret;
2051 ret = rt->recvbuf_pos < rt->recvbuf_len;
2056 rt->cur_transport_priv = NULL;
2058 } else if (ret == 1) {
2061 rt->cur_transport_priv = NULL;
2065 if (rt->transport == RTSP_TRANSPORT_RTP) {
2067 int64_t first_queue_time = 0;
2068 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2069 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2073 queue_time = ff_rtp_queued_packet_time(rtpctx);
2074 if (queue_time && (queue_time - first_queue_time < 0 ||
2075 !first_queue_time)) {
2076 first_queue_time = queue_time;
2077 first_queue_st = rt->rtsp_streams[i];
2080 if (first_queue_time) {
2081 wait_end = first_queue_time + s->max_delay;
2084 first_queue_st = NULL;
2088 /* read next RTP packet */
2090 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2092 return AVERROR(ENOMEM);
2095 switch(rt->lower_transport) {
2097 #if CONFIG_RTSP_DEMUXER
2098 case RTSP_LOWER_TRANSPORT_TCP:
2099 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2102 case RTSP_LOWER_TRANSPORT_UDP:
2103 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2104 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2105 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2106 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2108 case RTSP_LOWER_TRANSPORT_CUSTOM:
2109 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2110 wait_end && wait_end < av_gettime_relative())
2111 len = AVERROR(EAGAIN);
2113 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2114 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2115 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2116 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2119 if (len == AVERROR(EAGAIN) && first_queue_st &&
2120 rt->transport == RTSP_TRANSPORT_RTP) {
2121 rtsp_st = first_queue_st;
2122 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2129 if (rt->transport == RTSP_TRANSPORT_RDT) {
2130 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2131 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2132 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2133 if (rtsp_st->feedback) {
2134 AVIOContext *pb = NULL;
2135 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2137 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2140 /* Either bad packet, or a RTCP packet. Check if the
2141 * first_rtcp_ntp_time field was initialized. */
2142 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2143 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2144 /* first_rtcp_ntp_time has been initialized for this stream,
2145 * copy the same value to all other uninitialized streams,
2146 * in order to map their timestamp origin to the same ntp time
2149 AVStream *st = NULL;
2150 if (rtsp_st->stream_index >= 0)
2151 st = s->streams[rtsp_st->stream_index];
2152 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2153 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2154 AVStream *st2 = NULL;
2155 if (rt->rtsp_streams[i]->stream_index >= 0)
2156 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2157 if (rtpctx2 && st && st2 &&
2158 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2159 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2160 rtpctx2->rtcp_ts_offset = av_rescale_q(
2161 rtpctx->rtcp_ts_offset, st->time_base,
2165 // Make real NTP start time available in AVFormatContext
2166 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2167 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2169 s->start_time_realtime -=
2170 av_rescale (rtpctx->rtcp_ts_offset,
2171 (uint64_t) rtpctx->st->time_base.num * 1000000,
2172 rtpctx->st->time_base.den);
2176 if (ret == -RTCP_BYE) {
2179 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2180 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2182 if (rt->nb_byes == rt->nb_rtsp_streams)
2186 } else if (CONFIG_RTPDEC && rt->ts) {
2187 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2190 rt->recvbuf_len = len;
2191 rt->recvbuf_pos = ret;
2192 rt->cur_transport_priv = rt->ts;
2199 return AVERROR_INVALIDDATA;
2205 /* more packets may follow, so we save the RTP context */
2206 rt->cur_transport_priv = rtsp_st->transport_priv;
2210 #endif /* CONFIG_RTPDEC */
2212 #if CONFIG_SDP_DEMUXER
2213 static int sdp_probe(AVProbeData *p1)
2215 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2217 /* we look for a line beginning "c=IN IP" */
2218 while (p < p_end && *p != '\0') {
2219 if (sizeof("c=IN IP") - 1 < p_end - p &&
2220 av_strstart(p, "c=IN IP", NULL))
2221 return AVPROBE_SCORE_EXTENSION;
2223 while (p < p_end - 1 && *p != '\n') p++;
2232 static void append_source_addrs(char *buf, int size, const char *name,
2233 int count, struct RTSPSource **addrs)
2238 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2239 for (i = 1; i < count; i++)
2240 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2243 static int sdp_read_header(AVFormatContext *s)
2245 RTSPState *rt = s->priv_data;
2246 RTSPStream *rtsp_st;
2251 if (!ff_network_init())
2252 return AVERROR(EIO);
2254 if (s->max_delay < 0) /* Not set by the caller */
2255 s->max_delay = DEFAULT_REORDERING_DELAY;
2256 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2257 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2259 /* read the whole sdp file */
2260 /* XXX: better loading */
2261 content = av_malloc(SDP_MAX_SIZE);
2262 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2265 return AVERROR_INVALIDDATA;
2267 content[size] ='\0';
2269 err = ff_sdp_parse(s, content);
2273 /* open each RTP stream */
2274 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2276 rtsp_st = rt->rtsp_streams[i];
2278 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2279 AVDictionary *opts = map_to_opts(rt);
2281 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2282 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2283 ff_url_join(url, sizeof(url), "rtp", NULL,
2284 namebuf, rtsp_st->sdp_port,
2285 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2286 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2287 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2288 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2290 append_source_addrs(url, sizeof(url), "sources",
2291 rtsp_st->nb_include_source_addrs,
2292 rtsp_st->include_source_addrs);
2293 append_source_addrs(url, sizeof(url), "block",
2294 rtsp_st->nb_exclude_source_addrs,
2295 rtsp_st->exclude_source_addrs);
2296 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2297 &s->interrupt_callback, &opts);
2299 av_dict_free(&opts);
2302 err = AVERROR_INVALIDDATA;
2306 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2311 ff_rtsp_close_streams(s);
2316 static int sdp_read_close(AVFormatContext *s)
2318 ff_rtsp_close_streams(s);
2323 static const AVClass sdp_demuxer_class = {
2324 .class_name = "SDP demuxer",
2325 .item_name = av_default_item_name,
2326 .option = sdp_options,
2327 .version = LIBAVUTIL_VERSION_INT,
2330 AVInputFormat ff_sdp_demuxer = {
2332 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2333 .priv_data_size = sizeof(RTSPState),
2334 .read_probe = sdp_probe,
2335 .read_header = sdp_read_header,
2336 .read_packet = ff_rtsp_fetch_packet,
2337 .read_close = sdp_read_close,
2338 .priv_class = &sdp_demuxer_class,
2340 #endif /* CONFIG_SDP_DEMUXER */
2342 #if CONFIG_RTP_DEMUXER
2343 static int rtp_probe(AVProbeData *p)
2345 if (av_strstart(p->filename, "rtp:", NULL))
2346 return AVPROBE_SCORE_MAX;
2350 static int rtp_read_header(AVFormatContext *s)
2352 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2353 char host[500], sdp[500];
2355 URLContext* in = NULL;
2357 AVCodecContext codec = { 0 };
2358 struct sockaddr_storage addr;
2360 socklen_t addrlen = sizeof(addr);
2361 RTSPState *rt = s->priv_data;
2363 if (!ff_network_init())
2364 return AVERROR(EIO);
2366 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2367 &s->interrupt_callback, NULL);
2372 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2373 if (ret == AVERROR(EAGAIN))
2378 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2382 if ((recvbuf[0] & 0xc0) != 0x80) {
2383 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2388 if (RTP_PT_IS_RTCP(recvbuf[1]))
2391 payload_type = recvbuf[1] & 0x7f;
2394 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2398 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2399 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2400 "without an SDP file describing it\n",
2404 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2405 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2406 "properly you need an SDP file "
2410 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2411 NULL, 0, s->filename);
2413 snprintf(sdp, sizeof(sdp),
2414 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2415 addr.ss_family == AF_INET ? 4 : 6, host,
2416 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2417 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2418 port, payload_type);
2419 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2421 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2424 /* sdp_read_header initializes this again */
2427 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2429 ret = sdp_read_header(s);
2440 static const AVClass rtp_demuxer_class = {
2441 .class_name = "RTP demuxer",
2442 .item_name = av_default_item_name,
2443 .option = rtp_options,
2444 .version = LIBAVUTIL_VERSION_INT,
2447 AVInputFormat ff_rtp_demuxer = {
2449 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2450 .priv_data_size = sizeof(RTSPState),
2451 .read_probe = rtp_probe,
2452 .read_header = rtp_read_header,
2453 .read_packet = ff_rtsp_fetch_packet,
2454 .read_close = sdp_read_close,
2455 .flags = AVFMT_NOFILE,
2456 .priv_class = &rtp_demuxer_class,
2458 #endif /* CONFIG_RTP_DEMUXER */