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 AVCodecContext *codec = st ? st->codec : NULL;
213 codec->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 AVCodecContext *codec = st->codec;
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 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
265 if (codec->codec_id == AV_CODEC_ID_NONE) {
266 RTPDynamicProtocolHandler *handler =
267 ff_rtp_handler_find_by_name(buf, codec->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 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
277 c = avcodec_find_decoder(codec->codec_id);
283 get_word_sep(buf, sizeof(buf), "/", &p);
285 switch (codec->codec_type) {
286 case AVMEDIA_TYPE_AUDIO:
287 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
288 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
289 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
291 codec->sample_rate = i;
292 avpriv_set_pts_info(st, 32, 1, codec->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->codec->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->codec, rtsp_st->sdp_payload_type);
511 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
512 st->codec->sample_rate > 0)
513 avpriv_set_pts_info(st, 32, 1, st->codec->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->codec->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, "range:", &p)) {
579 // this is so that seeking on a streamed file can work.
580 rtsp_parse_range_npt(p, &start, &end);
581 s->start_time = start;
582 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
583 s->duration = (end == AV_NOPTS_VALUE) ?
584 AV_NOPTS_VALUE : end - start;
585 } else if (av_strstart(p, "lang:", &p)) {
586 if (s->nb_streams > 0) {
587 get_word(buf1, sizeof(buf1), &p);
588 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
589 if (rtsp_st->stream_index >= 0) {
590 st = s->streams[rtsp_st->stream_index];
591 av_dict_set(&st->metadata, "language", buf1, 0);
594 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
595 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
597 rt->transport = RTSP_TRANSPORT_RDT;
598 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
600 st = s->streams[s->nb_streams - 1];
601 st->codec->sample_rate = atoi(p);
602 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
604 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
605 get_word(buf1, sizeof(buf1), &p); // ignore tag
606 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
607 p += strspn(p, SPACE_CHARS);
608 if (av_strstart(p, "inline:", &p))
609 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
610 } else if (av_strstart(p, "source-filter:", &p)) {
612 get_word(buf1, sizeof(buf1), &p);
613 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
615 exclude = !strcmp(buf1, "excl");
617 get_word(buf1, sizeof(buf1), &p);
618 if (strcmp(buf1, "IN") != 0)
620 get_word(buf1, sizeof(buf1), &p);
621 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
623 // not checking that the destination address actually matches or is wildcard
624 get_word(buf1, sizeof(buf1), &p);
627 rtsp_src = av_mallocz(sizeof(*rtsp_src));
630 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
632 if (s->nb_streams == 0) {
633 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
635 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
636 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
639 if (s->nb_streams == 0) {
640 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
642 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
643 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
648 if (rt->server_type == RTSP_SERVER_WMS)
649 ff_wms_parse_sdp_a_line(s, p);
650 if (s->nb_streams > 0) {
651 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
653 if (rt->server_type == RTSP_SERVER_REAL)
654 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
656 if (rtsp_st->dynamic_handler &&
657 rtsp_st->dynamic_handler->parse_sdp_a_line)
658 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
659 rtsp_st->stream_index,
660 rtsp_st->dynamic_protocol_context, buf);
667 int ff_sdp_parse(AVFormatContext *s, const char *content)
669 RTSPState *rt = s->priv_data;
672 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
673 * contain long SDP lines containing complete ASF Headers (several
674 * kB) or arrays of MDPR (RM stream descriptor) headers plus
675 * "rulebooks" describing their properties. Therefore, the SDP line
678 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
679 * in rtpdec_xiph.c. */
681 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
685 p += strspn(p, SPACE_CHARS);
693 /* get the content */
695 while (*p != '\n' && *p != '\r' && *p != '\0') {
696 if ((q - buf) < sizeof(buf) - 1)
701 sdp_parse_line(s, s1, letter, buf);
703 while (*p != '\n' && *p != '\0')
709 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
710 av_freep(&s1->default_include_source_addrs[i]);
711 av_freep(&s1->default_include_source_addrs);
712 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
713 av_freep(&s1->default_exclude_source_addrs[i]);
714 av_freep(&s1->default_exclude_source_addrs);
716 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
717 if (!rt->p) return AVERROR(ENOMEM);
720 #endif /* CONFIG_RTPDEC */
722 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
724 RTSPState *rt = s->priv_data;
727 for (i = 0; i < rt->nb_rtsp_streams; i++) {
728 RTSPStream *rtsp_st = rt->rtsp_streams[i];
731 if (rtsp_st->transport_priv) {
733 AVFormatContext *rtpctx = rtsp_st->transport_priv;
734 av_write_trailer(rtpctx);
735 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
736 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
737 ff_rtsp_tcp_write_packet(s, rtsp_st);
738 ffio_free_dyn_buf(&rtpctx->pb);
740 avio_closep(&rtpctx->pb);
742 avformat_free_context(rtpctx);
743 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
744 ff_rdt_parse_close(rtsp_st->transport_priv);
745 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
746 ff_rtp_parse_close(rtsp_st->transport_priv);
748 rtsp_st->transport_priv = NULL;
749 if (rtsp_st->rtp_handle)
750 ffurl_close(rtsp_st->rtp_handle);
751 rtsp_st->rtp_handle = NULL;
755 /* close and free RTSP streams */
756 void ff_rtsp_close_streams(AVFormatContext *s)
758 RTSPState *rt = s->priv_data;
762 ff_rtsp_undo_setup(s, 0);
763 for (i = 0; i < rt->nb_rtsp_streams; i++) {
764 rtsp_st = rt->rtsp_streams[i];
766 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
767 if (rtsp_st->dynamic_handler->close)
768 rtsp_st->dynamic_handler->close(
769 rtsp_st->dynamic_protocol_context);
770 av_free(rtsp_st->dynamic_protocol_context);
772 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
773 av_freep(&rtsp_st->include_source_addrs[j]);
774 av_freep(&rtsp_st->include_source_addrs);
775 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
776 av_freep(&rtsp_st->exclude_source_addrs[j]);
777 av_freep(&rtsp_st->exclude_source_addrs);
782 av_freep(&rt->rtsp_streams);
784 avformat_close_input(&rt->asf_ctx);
786 if (CONFIG_RTPDEC && rt->ts)
787 avpriv_mpegts_parse_close(rt->ts);
789 av_freep(&rt->recvbuf);
792 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
794 RTSPState *rt = s->priv_data;
796 int reordering_queue_size = rt->reordering_queue_size;
797 if (reordering_queue_size < 0) {
798 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
799 reordering_queue_size = 0;
801 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
804 /* open the RTP context */
805 if (rtsp_st->stream_index >= 0)
806 st = s->streams[rtsp_st->stream_index];
808 s->ctx_flags |= AVFMTCTX_NOHEADER;
810 if (CONFIG_RTSP_MUXER && s->oformat && st) {
811 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
812 s, st, rtsp_st->rtp_handle,
813 RTSP_TCP_MAX_PACKET_SIZE,
814 rtsp_st->stream_index);
815 /* Ownership of rtp_handle is passed to the rtp mux context */
816 rtsp_st->rtp_handle = NULL;
819 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
820 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
821 return 0; // Don't need to open any parser here
822 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
823 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
824 rtsp_st->dynamic_protocol_context,
825 rtsp_st->dynamic_handler);
826 else if (CONFIG_RTPDEC)
827 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
828 rtsp_st->sdp_payload_type,
829 reordering_queue_size);
831 if (!rtsp_st->transport_priv) {
832 return AVERROR(ENOMEM);
833 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
834 if (rtsp_st->dynamic_handler) {
835 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
836 rtsp_st->dynamic_protocol_context,
837 rtsp_st->dynamic_handler);
839 if (rtsp_st->crypto_suite[0])
840 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
841 rtsp_st->crypto_suite,
842 rtsp_st->crypto_params);
848 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
849 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
856 q += strspn(q, SPACE_CHARS);
857 v = strtol(q, &p, 10);
861 v = strtol(p, &p, 10);
870 /* XXX: only one transport specification is parsed */
871 static void rtsp_parse_transport(AVFormatContext *s,
872 RTSPMessageHeader *reply, const char *p)
874 char transport_protocol[16];
876 char lower_transport[16];
878 RTSPTransportField *th;
881 reply->nb_transports = 0;
884 p += strspn(p, SPACE_CHARS);
888 th = &reply->transports[reply->nb_transports];
890 get_word_sep(transport_protocol, sizeof(transport_protocol),
892 if (!av_strcasecmp (transport_protocol, "rtp")) {
893 get_word_sep(profile, sizeof(profile), "/;,", &p);
894 lower_transport[0] = '\0';
895 /* rtp/avp/<protocol> */
897 get_word_sep(lower_transport, sizeof(lower_transport),
900 th->transport = RTSP_TRANSPORT_RTP;
901 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
902 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
903 /* x-pn-tng/<protocol> */
904 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
906 th->transport = RTSP_TRANSPORT_RDT;
907 } else if (!av_strcasecmp(transport_protocol, "raw")) {
908 get_word_sep(profile, sizeof(profile), "/;,", &p);
909 lower_transport[0] = '\0';
910 /* raw/raw/<protocol> */
912 get_word_sep(lower_transport, sizeof(lower_transport),
915 th->transport = RTSP_TRANSPORT_RAW;
917 if (!av_strcasecmp(lower_transport, "TCP"))
918 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
920 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
924 /* get each parameter */
925 while (*p != '\0' && *p != ',') {
926 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
927 if (!strcmp(parameter, "port")) {
930 rtsp_parse_range(&th->port_min, &th->port_max, &p);
932 } else if (!strcmp(parameter, "client_port")) {
935 rtsp_parse_range(&th->client_port_min,
936 &th->client_port_max, &p);
938 } else if (!strcmp(parameter, "server_port")) {
941 rtsp_parse_range(&th->server_port_min,
942 &th->server_port_max, &p);
944 } else if (!strcmp(parameter, "interleaved")) {
947 rtsp_parse_range(&th->interleaved_min,
948 &th->interleaved_max, &p);
950 } else if (!strcmp(parameter, "multicast")) {
951 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
952 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
953 } else if (!strcmp(parameter, "ttl")) {
957 th->ttl = strtol(p, &end, 10);
960 } else if (!strcmp(parameter, "destination")) {
963 get_word_sep(buf, sizeof(buf), ";,", &p);
964 get_sockaddr(s, buf, &th->destination);
966 } else if (!strcmp(parameter, "source")) {
969 get_word_sep(buf, sizeof(buf), ";,", &p);
970 av_strlcpy(th->source, buf, sizeof(th->source));
972 } else if (!strcmp(parameter, "mode")) {
975 get_word_sep(buf, sizeof(buf), ";, ", &p);
976 if (!strcmp(buf, "record") ||
977 !strcmp(buf, "receive"))
982 while (*p != ';' && *p != '\0' && *p != ',')
990 reply->nb_transports++;
991 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
996 static void handle_rtp_info(RTSPState *rt, const char *url,
997 uint32_t seq, uint32_t rtptime)
1000 if (!rtptime || !url[0])
1002 if (rt->transport != RTSP_TRANSPORT_RTP)
1004 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1005 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1006 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1009 if (!strcmp(rtsp_st->control_url, url)) {
1010 rtpctx->base_timestamp = rtptime;
1016 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1019 char key[20], value[1024], url[1024] = "";
1020 uint32_t seq = 0, rtptime = 0;
1023 p += strspn(p, SPACE_CHARS);
1026 get_word_sep(key, sizeof(key), "=", &p);
1030 get_word_sep(value, sizeof(value), ";, ", &p);
1032 if (!strcmp(key, "url"))
1033 av_strlcpy(url, value, sizeof(url));
1034 else if (!strcmp(key, "seq"))
1035 seq = strtoul(value, NULL, 10);
1036 else if (!strcmp(key, "rtptime"))
1037 rtptime = strtoul(value, NULL, 10);
1039 handle_rtp_info(rt, url, seq, rtptime);
1048 handle_rtp_info(rt, url, seq, rtptime);
1051 void ff_rtsp_parse_line(AVFormatContext *s,
1052 RTSPMessageHeader *reply, const char *buf,
1053 RTSPState *rt, const char *method)
1057 /* NOTE: we do case independent match for broken servers */
1059 if (av_stristart(p, "Session:", &p)) {
1061 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1062 if (av_stristart(p, ";timeout=", &p) &&
1063 (t = strtol(p, NULL, 10)) > 0) {
1066 } else if (av_stristart(p, "Content-Length:", &p)) {
1067 reply->content_length = strtol(p, NULL, 10);
1068 } else if (av_stristart(p, "Transport:", &p)) {
1069 rtsp_parse_transport(s, reply, p);
1070 } else if (av_stristart(p, "CSeq:", &p)) {
1071 reply->seq = strtol(p, NULL, 10);
1072 } else if (av_stristart(p, "Range:", &p)) {
1073 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1074 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1075 p += strspn(p, SPACE_CHARS);
1076 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1077 } else if (av_stristart(p, "Server:", &p)) {
1078 p += strspn(p, SPACE_CHARS);
1079 av_strlcpy(reply->server, p, sizeof(reply->server));
1080 } else if (av_stristart(p, "Notice:", &p) ||
1081 av_stristart(p, "X-Notice:", &p)) {
1082 reply->notice = strtol(p, NULL, 10);
1083 } else if (av_stristart(p, "Location:", &p)) {
1084 p += strspn(p, SPACE_CHARS);
1085 av_strlcpy(reply->location, p , sizeof(reply->location));
1086 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1087 p += strspn(p, SPACE_CHARS);
1088 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1089 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1090 p += strspn(p, SPACE_CHARS);
1091 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1092 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1093 p += strspn(p, SPACE_CHARS);
1094 if (method && !strcmp(method, "DESCRIBE"))
1095 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1096 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1097 p += strspn(p, SPACE_CHARS);
1098 if (method && !strcmp(method, "PLAY"))
1099 rtsp_parse_rtp_info(rt, p);
1100 } else if (av_stristart(p, "Public:", &p) && rt) {
1101 if (strstr(p, "GET_PARAMETER") &&
1102 method && !strcmp(method, "OPTIONS"))
1103 rt->get_parameter_supported = 1;
1104 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1105 p += strspn(p, SPACE_CHARS);
1106 rt->accept_dynamic_rate = atoi(p);
1107 } else if (av_stristart(p, "Content-Type:", &p)) {
1108 p += strspn(p, SPACE_CHARS);
1109 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1113 /* skip a RTP/TCP interleaved packet */
1114 void ff_rtsp_skip_packet(AVFormatContext *s)
1116 RTSPState *rt = s->priv_data;
1120 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1123 len = AV_RB16(buf + 1);
1125 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1130 if (len1 > sizeof(buf))
1132 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1139 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1140 unsigned char **content_ptr,
1141 int return_on_interleaved_data, const char *method)
1143 RTSPState *rt = s->priv_data;
1144 char buf[4096], buf1[1024], *q;
1147 int ret, content_length, line_count = 0, request = 0;
1148 unsigned char *content = NULL;
1154 memset(reply, 0, sizeof(*reply));
1156 /* parse reply (XXX: use buffers) */
1157 rt->last_reply[0] = '\0';
1161 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1162 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1167 if (ch == '$' && q == buf) {
1168 if (return_on_interleaved_data) {
1171 ff_rtsp_skip_packet(s);
1172 } else if (ch != '\r') {
1173 if ((q - buf) < sizeof(buf) - 1)
1179 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1181 /* test if last line */
1185 if (line_count == 0) {
1186 /* get reply code */
1187 get_word(buf1, sizeof(buf1), &p);
1188 if (!strncmp(buf1, "RTSP/", 5)) {
1189 get_word(buf1, sizeof(buf1), &p);
1190 reply->status_code = atoi(buf1);
1191 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1193 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1194 get_word(buf1, sizeof(buf1), &p); // object
1198 ff_rtsp_parse_line(s, reply, p, rt, method);
1199 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1200 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1205 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1206 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1208 content_length = reply->content_length;
1209 if (content_length > 0) {
1210 /* leave some room for a trailing '\0' (useful for simple parsing) */
1211 content = av_malloc(content_length + 1);
1213 return AVERROR(ENOMEM);
1214 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1215 content[content_length] = '\0';
1218 *content_ptr = content;
1224 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1225 const char* ptr = buf;
1227 if (!strcmp(reply->reason, "OPTIONS")) {
1228 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1230 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1231 if (reply->session_id[0])
1232 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1235 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1237 av_strlcat(buf, "\r\n", sizeof(buf));
1239 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1240 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1243 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1245 rt->last_cmd_time = av_gettime_relative();
1246 /* Even if the request from the server had data, it is not the data
1247 * that the caller wants or expects. The memory could also be leaked
1248 * if the actual following reply has content data. */
1250 av_freep(content_ptr);
1251 /* If method is set, this is called from ff_rtsp_send_cmd,
1252 * where a reply to exactly this request is awaited. For
1253 * callers from within packet receiving, we just want to
1254 * return to the caller and go back to receiving packets. */
1260 if (rt->seq != reply->seq) {
1261 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1262 rt->seq, reply->seq);
1266 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1267 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1268 reply->notice == 2306 /* Continuous Feed Terminated */) {
1269 rt->state = RTSP_STATE_IDLE;
1270 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1271 return AVERROR(EIO); /* data or server error */
1272 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1273 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1274 return AVERROR(EPERM);
1280 * Send a command to the RTSP server without waiting for the reply.
1282 * @param s RTSP (de)muxer context
1283 * @param method the method for the request
1284 * @param url the target url for the request
1285 * @param headers extra header lines to include in the request
1286 * @param send_content if non-null, the data to send as request body content
1287 * @param send_content_length the length of the send_content data, or 0 if
1288 * send_content is null
1290 * @return zero if success, nonzero otherwise
1292 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1293 const char *method, const char *url,
1294 const char *headers,
1295 const unsigned char *send_content,
1296 int send_content_length)
1298 RTSPState *rt = s->priv_data;
1299 char buf[4096], *out_buf;
1300 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1302 /* Add in RTSP headers */
1305 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1307 av_strlcat(buf, headers, sizeof(buf));
1308 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1309 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1310 if (rt->session_id[0] != '\0' && (!headers ||
1311 !strstr(headers, "\nIf-Match:"))) {
1312 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1315 char *str = ff_http_auth_create_response(&rt->auth_state,
1316 rt->auth, url, method);
1318 av_strlcat(buf, str, sizeof(buf));
1321 if (send_content_length > 0 && send_content)
1322 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1323 av_strlcat(buf, "\r\n", sizeof(buf));
1325 /* base64 encode rtsp if tunneling */
1326 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1327 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1328 out_buf = base64buf;
1331 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1333 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1334 if (send_content_length > 0 && send_content) {
1335 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1336 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1337 "with content data not supported\n");
1338 return AVERROR_PATCHWELCOME;
1340 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1342 rt->last_cmd_time = av_gettime_relative();
1347 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1348 const char *url, const char *headers)
1350 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1353 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1354 const char *headers, RTSPMessageHeader *reply,
1355 unsigned char **content_ptr)
1357 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1358 content_ptr, NULL, 0);
1361 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1362 const char *method, const char *url,
1364 RTSPMessageHeader *reply,
1365 unsigned char **content_ptr,
1366 const unsigned char *send_content,
1367 int send_content_length)
1369 RTSPState *rt = s->priv_data;
1370 HTTPAuthType cur_auth_type;
1371 int ret, attempts = 0;
1374 cur_auth_type = rt->auth_state.auth_type;
1375 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1377 send_content_length)))
1380 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1384 if (reply->status_code == 401 &&
1385 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1386 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1389 if (reply->status_code > 400){
1390 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1394 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1400 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1401 int lower_transport, const char *real_challenge)
1403 RTSPState *rt = s->priv_data;
1404 int rtx = 0, j, i, err, interleave = 0, port_off;
1405 RTSPStream *rtsp_st;
1406 RTSPMessageHeader reply1, *reply = &reply1;
1408 const char *trans_pref;
1410 if (rt->transport == RTSP_TRANSPORT_RDT)
1411 trans_pref = "x-pn-tng";
1412 else if (rt->transport == RTSP_TRANSPORT_RAW)
1413 trans_pref = "RAW/RAW";
1415 trans_pref = "RTP/AVP";
1417 /* default timeout: 1 minute */
1420 /* Choose a random starting offset within the first half of the
1421 * port range, to allow for a number of ports to try even if the offset
1422 * happens to be at the end of the random range. */
1423 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1424 /* even random offset */
1425 port_off -= port_off & 0x01;
1427 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1428 char transport[2048];
1431 * WMS serves all UDP data over a single connection, the RTX, which
1432 * isn't necessarily the first in the SDP but has to be the first
1433 * to be set up, else the second/third SETUP will fail with a 461.
1435 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1436 rt->server_type == RTSP_SERVER_WMS) {
1439 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1440 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1442 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1446 if (rtx == rt->nb_rtsp_streams)
1447 return -1; /* no RTX found */
1448 rtsp_st = rt->rtsp_streams[rtx];
1450 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1452 rtsp_st = rt->rtsp_streams[i];
1455 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1458 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1459 port = reply->transports[0].client_port_min;
1463 /* first try in specified port range */
1464 while (j <= rt->rtp_port_max) {
1465 AVDictionary *opts = map_to_opts(rt);
1467 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1468 "?localport=%d", j);
1469 /* we will use two ports per rtp stream (rtp and rtcp) */
1471 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1472 &s->interrupt_callback, &opts);
1474 av_dict_free(&opts);
1479 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1484 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1486 snprintf(transport, sizeof(transport) - 1,
1487 "%s/UDP;", trans_pref);
1488 if (rt->server_type != RTSP_SERVER_REAL)
1489 av_strlcat(transport, "unicast;", sizeof(transport));
1490 av_strlcatf(transport, sizeof(transport),
1491 "client_port=%d", port);
1492 if (rt->transport == RTSP_TRANSPORT_RTP &&
1493 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1494 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1498 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1499 /* For WMS streams, the application streams are only used for
1500 * UDP. When trying to set it up for TCP streams, the server
1501 * will return an error. Therefore, we skip those streams. */
1502 if (rt->server_type == RTSP_SERVER_WMS &&
1503 (rtsp_st->stream_index < 0 ||
1504 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1507 snprintf(transport, sizeof(transport) - 1,
1508 "%s/TCP;", trans_pref);
1509 if (rt->transport != RTSP_TRANSPORT_RDT)
1510 av_strlcat(transport, "unicast;", sizeof(transport));
1511 av_strlcatf(transport, sizeof(transport),
1512 "interleaved=%d-%d",
1513 interleave, interleave + 1);
1517 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1518 snprintf(transport, sizeof(transport) - 1,
1519 "%s/UDP;multicast", trans_pref);
1522 av_strlcat(transport, ";mode=record", sizeof(transport));
1523 } else if (rt->server_type == RTSP_SERVER_REAL ||
1524 rt->server_type == RTSP_SERVER_WMS)
1525 av_strlcat(transport, ";mode=play", sizeof(transport));
1526 snprintf(cmd, sizeof(cmd),
1527 "Transport: %s\r\n",
1529 if (rt->accept_dynamic_rate)
1530 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1531 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1532 char real_res[41], real_csum[9];
1533 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1535 av_strlcatf(cmd, sizeof(cmd),
1537 "RealChallenge2: %s, sd=%s\r\n",
1538 rt->session_id, real_res, real_csum);
1540 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1541 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1544 } else if (reply->status_code != RTSP_STATUS_OK ||
1545 reply->nb_transports != 1) {
1546 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1550 /* XXX: same protocol for all streams is required */
1552 if (reply->transports[0].lower_transport != rt->lower_transport ||
1553 reply->transports[0].transport != rt->transport) {
1554 err = AVERROR_INVALIDDATA;
1558 rt->lower_transport = reply->transports[0].lower_transport;
1559 rt->transport = reply->transports[0].transport;
1562 /* Fail if the server responded with another lower transport mode
1563 * than what we requested. */
1564 if (reply->transports[0].lower_transport != lower_transport) {
1565 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1566 err = AVERROR_INVALIDDATA;
1570 switch(reply->transports[0].lower_transport) {
1571 case RTSP_LOWER_TRANSPORT_TCP:
1572 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1573 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1576 case RTSP_LOWER_TRANSPORT_UDP: {
1577 char url[1024], options[30] = "";
1578 const char *peer = host;
1580 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1581 av_strlcpy(options, "?connect=1", sizeof(options));
1582 /* Use source address if specified */
1583 if (reply->transports[0].source[0])
1584 peer = reply->transports[0].source;
1585 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1586 reply->transports[0].server_port_min, "%s", options);
1587 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1588 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1589 err = AVERROR_INVALIDDATA;
1594 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1595 char url[1024], namebuf[50], optbuf[20] = "";
1596 struct sockaddr_storage addr;
1599 if (reply->transports[0].destination.ss_family) {
1600 addr = reply->transports[0].destination;
1601 port = reply->transports[0].port_min;
1602 ttl = reply->transports[0].ttl;
1604 addr = rtsp_st->sdp_ip;
1605 port = rtsp_st->sdp_port;
1606 ttl = rtsp_st->sdp_ttl;
1609 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1610 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1611 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1612 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1613 port, "%s", optbuf);
1614 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1615 &s->interrupt_callback, NULL) < 0) {
1616 err = AVERROR_INVALIDDATA;
1623 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1627 if (rt->nb_rtsp_streams && reply->timeout > 0)
1628 rt->timeout = reply->timeout;
1630 if (rt->server_type == RTSP_SERVER_REAL)
1631 rt->need_subscription = 1;
1636 ff_rtsp_undo_setup(s, 0);
1640 void ff_rtsp_close_connections(AVFormatContext *s)
1642 RTSPState *rt = s->priv_data;
1643 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1644 ffurl_close(rt->rtsp_hd);
1645 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1648 int ff_rtsp_connect(AVFormatContext *s)
1650 RTSPState *rt = s->priv_data;
1651 char proto[128], host[1024], path[1024];
1652 char tcpname[1024], cmd[2048], auth[128];
1653 const char *lower_rtsp_proto = "tcp";
1654 int port, err, tcp_fd;
1655 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1656 int lower_transport_mask = 0;
1657 int default_port = RTSP_DEFAULT_PORT;
1658 char real_challenge[64] = "";
1659 struct sockaddr_storage peer;
1660 socklen_t peer_len = sizeof(peer);
1662 if (rt->rtp_port_max < rt->rtp_port_min) {
1663 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1664 "than min port %d\n", rt->rtp_port_max,
1666 return AVERROR(EINVAL);
1669 if (!ff_network_init())
1670 return AVERROR(EIO);
1672 if (s->max_delay < 0) /* Not set by the caller */
1673 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1675 rt->control_transport = RTSP_MODE_PLAIN;
1676 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1677 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1678 rt->control_transport = RTSP_MODE_TUNNEL;
1680 /* Only pass through valid flags from here */
1681 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1684 /* extract hostname and port */
1685 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1686 host, sizeof(host), &port, path, sizeof(path), s->filename);
1688 if (!strcmp(proto, "rtsps")) {
1689 lower_rtsp_proto = "tls";
1690 default_port = RTSPS_DEFAULT_PORT;
1691 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1695 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1698 port = default_port;
1700 lower_transport_mask = rt->lower_transport_mask;
1702 if (!lower_transport_mask)
1703 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1706 /* Only UDP or TCP - UDP multicast isn't supported. */
1707 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1708 (1 << RTSP_LOWER_TRANSPORT_TCP);
1709 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1710 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1711 "only UDP and TCP are supported for output.\n");
1712 err = AVERROR(EINVAL);
1717 /* Construct the URI used in request; this is similar to s->filename,
1718 * but with authentication credentials removed and RTSP specific options
1720 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1721 host, port, "%s", path);
1723 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1724 /* set up initial handshake for tunneling */
1725 char httpname[1024];
1726 char sessioncookie[17];
1729 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1730 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1731 av_get_random_seed(), av_get_random_seed());
1734 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1735 &s->interrupt_callback) < 0) {
1740 /* generate GET headers */
1741 snprintf(headers, sizeof(headers),
1742 "x-sessioncookie: %s\r\n"
1743 "Accept: application/x-rtsp-tunnelled\r\n"
1744 "Pragma: no-cache\r\n"
1745 "Cache-Control: no-cache\r\n",
1747 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1749 /* complete the connection */
1750 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1756 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1757 &s->interrupt_callback) < 0 ) {
1762 /* generate POST headers */
1763 snprintf(headers, sizeof(headers),
1764 "x-sessioncookie: %s\r\n"
1765 "Content-Type: application/x-rtsp-tunnelled\r\n"
1766 "Pragma: no-cache\r\n"
1767 "Cache-Control: no-cache\r\n"
1768 "Content-Length: 32767\r\n"
1769 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1771 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1772 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1774 /* Initialize the authentication state for the POST session. The HTTP
1775 * protocol implementation doesn't properly handle multi-pass
1776 * authentication for POST requests, since it would require one of
1778 * - implementing Expect: 100-continue, which many HTTP servers
1779 * don't support anyway, even less the RTSP servers that do HTTP
1781 * - sending the whole POST data until getting a 401 reply specifying
1782 * what authentication method to use, then resending all that data
1783 * - waiting for potential 401 replies directly after sending the
1784 * POST header (waiting for some unspecified time)
1785 * Therefore, we copy the full auth state, which works for both basic
1786 * and digest. (For digest, we would have to synchronize the nonce
1787 * count variable between the two sessions, if we'd do more requests
1788 * with the original session, though.)
1790 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1792 /* complete the connection */
1793 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1799 /* open the tcp connection */
1800 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1802 "?timeout=%d", rt->stimeout);
1803 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1804 &s->interrupt_callback, NULL)) < 0) {
1808 rt->rtsp_hd_out = rt->rtsp_hd;
1812 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1817 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1818 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1819 NULL, 0, NI_NUMERICHOST);
1822 /* request options supported by the server; this also detects server
1824 for (rt->server_type = RTSP_SERVER_RTP;;) {
1826 if (rt->server_type == RTSP_SERVER_REAL)
1829 * The following entries are required for proper
1830 * streaming from a Realmedia server. They are
1831 * interdependent in some way although we currently
1832 * don't quite understand how. Values were copied
1833 * from mplayer SVN r23589.
1834 * ClientChallenge is a 16-byte ID in hex
1835 * CompanyID is a 16-byte ID in base64
1837 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1838 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1839 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1840 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1842 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1843 if (reply->status_code != RTSP_STATUS_OK) {
1844 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1848 /* detect server type if not standard-compliant RTP */
1849 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1850 rt->server_type = RTSP_SERVER_REAL;
1852 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1853 rt->server_type = RTSP_SERVER_WMS;
1854 } else if (rt->server_type == RTSP_SERVER_REAL)
1855 strcpy(real_challenge, reply->real_challenge);
1859 if (CONFIG_RTSP_DEMUXER && s->iformat)
1860 err = ff_rtsp_setup_input_streams(s, reply);
1861 else if (CONFIG_RTSP_MUXER)
1862 err = ff_rtsp_setup_output_streams(s, host);
1869 int lower_transport = ff_log2_tab[lower_transport_mask &
1870 ~(lower_transport_mask - 1)];
1872 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1873 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1874 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1876 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1877 rt->server_type == RTSP_SERVER_REAL ?
1878 real_challenge : NULL);
1881 lower_transport_mask &= ~(1 << lower_transport);
1882 if (lower_transport_mask == 0 && err == 1) {
1883 err = AVERROR(EPROTONOSUPPORT);
1888 rt->lower_transport_mask = lower_transport_mask;
1889 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1890 rt->state = RTSP_STATE_IDLE;
1891 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1894 ff_rtsp_close_streams(s);
1895 ff_rtsp_close_connections(s);
1896 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1897 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1898 rt->session_id[0] = '\0';
1899 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1907 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1910 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1911 uint8_t *buf, int buf_size, int64_t wait_end)
1913 RTSPState *rt = s->priv_data;
1914 RTSPStream *rtsp_st;
1915 int n, i, ret, tcp_fd, timeout_cnt = 0;
1917 struct pollfd *p = rt->p;
1918 int *fds = NULL, fdsnum, fdsidx;
1921 if (ff_check_interrupt(&s->interrupt_callback))
1922 return AVERROR_EXIT;
1923 if (wait_end && wait_end - av_gettime_relative() < 0)
1924 return AVERROR(EAGAIN);
1927 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1928 p[max_p].fd = tcp_fd;
1929 p[max_p++].events = POLLIN;
1933 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1934 rtsp_st = rt->rtsp_streams[i];
1935 if (rtsp_st->rtp_handle) {
1936 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1938 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1942 av_log(s, AV_LOG_ERROR,
1943 "Number of fds %d not supported\n", fdsnum);
1944 return AVERROR_INVALIDDATA;
1946 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1947 p[max_p].fd = fds[fdsidx];
1948 p[max_p++].events = POLLIN;
1953 n = poll(p, max_p, POLL_TIMEOUT_MS);
1955 int j = 1 - (tcp_fd == -1);
1957 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1958 rtsp_st = rt->rtsp_streams[i];
1959 if (rtsp_st->rtp_handle) {
1960 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1961 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1963 *prtsp_st = rtsp_st;
1970 #if CONFIG_RTSP_DEMUXER
1971 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1972 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1973 if (rt->state == RTSP_STATE_STREAMING) {
1974 if (!ff_rtsp_parse_streaming_commands(s))
1977 av_log(s, AV_LOG_WARNING,
1978 "Unable to answer to TEARDOWN\n");
1982 RTSPMessageHeader reply;
1983 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1986 /* XXX: parse message */
1987 if (rt->state != RTSP_STATE_STREAMING)
1992 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1993 return AVERROR(ETIMEDOUT);
1994 } else if (n < 0 && errno != EINTR)
1995 return AVERROR(errno);
1999 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2000 const uint8_t *buf, int len)
2002 RTSPState *rt = s->priv_data;
2006 if (rt->nb_rtsp_streams == 1) {
2007 *rtsp_st = rt->rtsp_streams[0];
2010 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2011 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2013 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2014 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2017 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2018 *rtsp_st = rt->rtsp_streams[i];
2025 av_log(s, AV_LOG_WARNING,
2026 "Unable to pick stream for packet - SSRC not known for "
2028 return AVERROR(EAGAIN);
2031 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2032 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2033 *rtsp_st = rt->rtsp_streams[i];
2039 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2040 return AVERROR(EAGAIN);
2043 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2045 RTSPState *rt = s->priv_data;
2047 RTSPStream *rtsp_st, *first_queue_st = NULL;
2048 int64_t wait_end = 0;
2050 if (rt->nb_byes == rt->nb_rtsp_streams)
2053 /* get next frames from the same RTP packet */
2054 if (rt->cur_transport_priv) {
2055 if (rt->transport == RTSP_TRANSPORT_RDT) {
2056 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2057 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2058 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2059 } else if (CONFIG_RTPDEC && rt->ts) {
2060 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2062 rt->recvbuf_pos += ret;
2063 ret = rt->recvbuf_pos < rt->recvbuf_len;
2068 rt->cur_transport_priv = NULL;
2070 } else if (ret == 1) {
2073 rt->cur_transport_priv = NULL;
2077 if (rt->transport == RTSP_TRANSPORT_RTP) {
2079 int64_t first_queue_time = 0;
2080 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2081 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2085 queue_time = ff_rtp_queued_packet_time(rtpctx);
2086 if (queue_time && (queue_time - first_queue_time < 0 ||
2087 !first_queue_time)) {
2088 first_queue_time = queue_time;
2089 first_queue_st = rt->rtsp_streams[i];
2092 if (first_queue_time) {
2093 wait_end = first_queue_time + s->max_delay;
2096 first_queue_st = NULL;
2100 /* read next RTP packet */
2102 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2104 return AVERROR(ENOMEM);
2107 switch(rt->lower_transport) {
2109 #if CONFIG_RTSP_DEMUXER
2110 case RTSP_LOWER_TRANSPORT_TCP:
2111 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2114 case RTSP_LOWER_TRANSPORT_UDP:
2115 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2116 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2117 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2118 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2120 case RTSP_LOWER_TRANSPORT_CUSTOM:
2121 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2122 wait_end && wait_end < av_gettime_relative())
2123 len = AVERROR(EAGAIN);
2125 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2126 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2127 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2128 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2131 if (len == AVERROR(EAGAIN) && first_queue_st &&
2132 rt->transport == RTSP_TRANSPORT_RTP) {
2133 av_log(s, AV_LOG_WARNING,
2134 "max delay reached. need to consume packet\n");
2135 rtsp_st = first_queue_st;
2136 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2143 if (rt->transport == RTSP_TRANSPORT_RDT) {
2144 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2145 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2146 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2147 if (rtsp_st->feedback) {
2148 AVIOContext *pb = NULL;
2149 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2151 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2154 /* Either bad packet, or a RTCP packet. Check if the
2155 * first_rtcp_ntp_time field was initialized. */
2156 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2157 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2158 /* first_rtcp_ntp_time has been initialized for this stream,
2159 * copy the same value to all other uninitialized streams,
2160 * in order to map their timestamp origin to the same ntp time
2163 AVStream *st = NULL;
2164 if (rtsp_st->stream_index >= 0)
2165 st = s->streams[rtsp_st->stream_index];
2166 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2167 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2168 AVStream *st2 = NULL;
2169 if (rt->rtsp_streams[i]->stream_index >= 0)
2170 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2171 if (rtpctx2 && st && st2 &&
2172 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2173 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2174 rtpctx2->rtcp_ts_offset = av_rescale_q(
2175 rtpctx->rtcp_ts_offset, st->time_base,
2179 // Make real NTP start time available in AVFormatContext
2180 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2181 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2183 s->start_time_realtime -=
2184 av_rescale (rtpctx->rtcp_ts_offset,
2185 (uint64_t) rtpctx->st->time_base.num * 1000000,
2186 rtpctx->st->time_base.den);
2190 if (ret == -RTCP_BYE) {
2193 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2194 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2196 if (rt->nb_byes == rt->nb_rtsp_streams)
2200 } else if (CONFIG_RTPDEC && rt->ts) {
2201 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2204 rt->recvbuf_len = len;
2205 rt->recvbuf_pos = ret;
2206 rt->cur_transport_priv = rt->ts;
2213 return AVERROR_INVALIDDATA;
2219 /* more packets may follow, so we save the RTP context */
2220 rt->cur_transport_priv = rtsp_st->transport_priv;
2224 #endif /* CONFIG_RTPDEC */
2226 #if CONFIG_SDP_DEMUXER
2227 static int sdp_probe(AVProbeData *p1)
2229 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2231 /* we look for a line beginning "c=IN IP" */
2232 while (p < p_end && *p != '\0') {
2233 if (sizeof("c=IN IP") - 1 < p_end - p &&
2234 av_strstart(p, "c=IN IP", NULL))
2235 return AVPROBE_SCORE_EXTENSION;
2237 while (p < p_end - 1 && *p != '\n') p++;
2246 static void append_source_addrs(char *buf, int size, const char *name,
2247 int count, struct RTSPSource **addrs)
2252 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2253 for (i = 1; i < count; i++)
2254 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2257 static int sdp_read_header(AVFormatContext *s)
2259 RTSPState *rt = s->priv_data;
2260 RTSPStream *rtsp_st;
2265 if (!ff_network_init())
2266 return AVERROR(EIO);
2268 if (s->max_delay < 0) /* Not set by the caller */
2269 s->max_delay = DEFAULT_REORDERING_DELAY;
2270 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2271 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2273 /* read the whole sdp file */
2274 /* XXX: better loading */
2275 content = av_malloc(SDP_MAX_SIZE);
2277 return AVERROR(ENOMEM);
2278 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2281 return AVERROR_INVALIDDATA;
2283 content[size] ='\0';
2285 err = ff_sdp_parse(s, content);
2289 /* open each RTP stream */
2290 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2292 rtsp_st = rt->rtsp_streams[i];
2294 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2295 AVDictionary *opts = map_to_opts(rt);
2297 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2298 sizeof(rtsp_st->sdp_ip),
2299 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2301 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2303 av_dict_free(&opts);
2306 ff_url_join(url, sizeof(url), "rtp", NULL,
2307 namebuf, rtsp_st->sdp_port,
2308 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2309 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2310 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2311 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2313 append_source_addrs(url, sizeof(url), "sources",
2314 rtsp_st->nb_include_source_addrs,
2315 rtsp_st->include_source_addrs);
2316 append_source_addrs(url, sizeof(url), "block",
2317 rtsp_st->nb_exclude_source_addrs,
2318 rtsp_st->exclude_source_addrs);
2319 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2320 &s->interrupt_callback, &opts);
2322 av_dict_free(&opts);
2325 err = AVERROR_INVALIDDATA;
2329 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2334 ff_rtsp_close_streams(s);
2339 static int sdp_read_close(AVFormatContext *s)
2341 ff_rtsp_close_streams(s);
2346 static const AVClass sdp_demuxer_class = {
2347 .class_name = "SDP demuxer",
2348 .item_name = av_default_item_name,
2349 .option = sdp_options,
2350 .version = LIBAVUTIL_VERSION_INT,
2353 AVInputFormat ff_sdp_demuxer = {
2355 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2356 .priv_data_size = sizeof(RTSPState),
2357 .read_probe = sdp_probe,
2358 .read_header = sdp_read_header,
2359 .read_packet = ff_rtsp_fetch_packet,
2360 .read_close = sdp_read_close,
2361 .priv_class = &sdp_demuxer_class,
2363 #endif /* CONFIG_SDP_DEMUXER */
2365 #if CONFIG_RTP_DEMUXER
2366 static int rtp_probe(AVProbeData *p)
2368 if (av_strstart(p->filename, "rtp:", NULL))
2369 return AVPROBE_SCORE_MAX;
2373 static int rtp_read_header(AVFormatContext *s)
2375 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2376 char host[500], sdp[500];
2378 URLContext* in = NULL;
2380 AVCodecContext codec = { 0 };
2381 struct sockaddr_storage addr;
2383 socklen_t addrlen = sizeof(addr);
2384 RTSPState *rt = s->priv_data;
2386 if (!ff_network_init())
2387 return AVERROR(EIO);
2389 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2390 &s->interrupt_callback, NULL);
2395 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2396 if (ret == AVERROR(EAGAIN))
2401 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2405 if ((recvbuf[0] & 0xc0) != 0x80) {
2406 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2411 if (RTP_PT_IS_RTCP(recvbuf[1]))
2414 payload_type = recvbuf[1] & 0x7f;
2417 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2421 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2422 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2423 "without an SDP file describing it\n",
2427 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2428 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2429 "properly you need an SDP file "
2433 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2434 NULL, 0, s->filename);
2436 snprintf(sdp, sizeof(sdp),
2437 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2438 addr.ss_family == AF_INET ? 4 : 6, host,
2439 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2440 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2441 port, payload_type);
2442 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2444 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2447 /* sdp_read_header initializes this again */
2450 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2452 ret = sdp_read_header(s);
2463 static const AVClass rtp_demuxer_class = {
2464 .class_name = "RTP demuxer",
2465 .item_name = av_default_item_name,
2466 .option = rtp_options,
2467 .version = LIBAVUTIL_VERSION_INT,
2470 AVInputFormat ff_rtp_demuxer = {
2472 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2473 .priv_data_size = sizeof(RTSPState),
2474 .read_probe = rtp_probe,
2475 .read_header = rtp_read_header,
2476 .read_packet = ff_rtsp_fetch_packet,
2477 .read_close = sdp_read_close,
2478 .flags = AVFMT_NOFILE,
2479 .priv_class = &rtp_demuxer_class,
2481 #endif /* CONFIG_RTP_DEMUXER */