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 { "listen_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 #if FF_API_OLD_RTSP_OPTIONS
98 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
99 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
101 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
104 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
105 #if FF_API_OLD_RTSP_OPTIONS
106 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
111 static const AVOption sdp_options[] = {
112 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
113 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
114 { "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" },
115 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
120 static const AVOption rtp_options[] = {
121 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
127 static AVDictionary *map_to_opts(RTSPState *rt)
129 AVDictionary *opts = NULL;
132 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
133 av_dict_set(&opts, "buffer_size", buf, 0);
138 static void get_word_until_chars(char *buf, int buf_size,
139 const char *sep, const char **pp)
145 p += strspn(p, SPACE_CHARS);
147 while (!strchr(sep, *p) && *p != '\0') {
148 if ((q - buf) < buf_size - 1)
157 static void get_word_sep(char *buf, int buf_size, const char *sep,
160 if (**pp == '/') (*pp)++;
161 get_word_until_chars(buf, buf_size, sep, pp);
164 static void get_word(char *buf, int buf_size, const char **pp)
166 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
169 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
171 * Used for seeking in the rtp stream.
173 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
177 p += strspn(p, SPACE_CHARS);
178 if (!av_stristart(p, "npt=", &p))
181 *start = AV_NOPTS_VALUE;
182 *end = AV_NOPTS_VALUE;
184 get_word_sep(buf, sizeof(buf), "-", &p);
185 if (av_parse_time(start, buf, 1) < 0)
189 get_word_sep(buf, sizeof(buf), "-", &p);
190 if (av_parse_time(end, buf, 1) < 0)
191 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
195 static int get_sockaddr(AVFormatContext *s,
196 const char *buf, struct sockaddr_storage *sock)
198 struct addrinfo hints = { 0 }, *ai = NULL;
201 hints.ai_flags = AI_NUMERICHOST;
202 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
203 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
208 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
214 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
215 RTSPStream *rtsp_st, AVStream *st)
217 AVCodecParameters *par = st ? st->codecpar : NULL;
221 par->codec_id = handler->codec_id;
222 rtsp_st->dynamic_handler = handler;
224 st->need_parsing = handler->need_parsing;
225 if (handler->priv_data_size) {
226 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
227 if (!rtsp_st->dynamic_protocol_context)
228 rtsp_st->dynamic_handler = NULL;
232 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
235 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
236 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
237 rtsp_st->dynamic_protocol_context);
239 if (rtsp_st->dynamic_protocol_context) {
240 if (rtsp_st->dynamic_handler->close)
241 rtsp_st->dynamic_handler->close(
242 rtsp_st->dynamic_protocol_context);
243 av_free(rtsp_st->dynamic_protocol_context);
245 rtsp_st->dynamic_protocol_context = NULL;
246 rtsp_st->dynamic_handler = NULL;
251 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
252 static int sdp_parse_rtpmap(AVFormatContext *s,
253 AVStream *st, RTSPStream *rtsp_st,
254 int payload_type, const char *p)
256 AVCodecParameters *par = st->codecpar;
259 const AVCodecDescriptor *desc;
262 /* See if we can handle this kind of payload.
263 * The space should normally not be there but some Real streams or
264 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
265 * have a trailing space. */
266 get_word_sep(buf, sizeof(buf), "/ ", &p);
267 if (payload_type < RTP_PT_PRIVATE) {
268 /* We are in a standard case
269 * (from http://www.iana.org/assignments/rtp-parameters). */
270 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
273 if (par->codec_id == AV_CODEC_ID_NONE) {
274 const RTPDynamicProtocolHandler *handler =
275 ff_rtp_handler_find_by_name(buf, par->codec_type);
276 init_rtp_handler(handler, rtsp_st, st);
277 /* If no dynamic handler was found, check with the list of standard
278 * allocated types, if such a stream for some reason happens to
279 * use a private payload type. This isn't handled in rtpdec.c, since
280 * the format name from the rtpmap line never is passed into rtpdec. */
281 if (!rtsp_st->dynamic_handler)
282 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
285 desc = avcodec_descriptor_get(par->codec_id);
286 if (desc && desc->name)
291 get_word_sep(buf, sizeof(buf), "/", &p);
293 switch (par->codec_type) {
294 case AVMEDIA_TYPE_AUDIO:
295 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
296 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
297 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
299 par->sample_rate = i;
300 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
301 get_word_sep(buf, sizeof(buf), "/", &p);
306 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
308 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
311 case AVMEDIA_TYPE_VIDEO:
312 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
314 avpriv_set_pts_info(st, 32, 1, i);
319 finalize_rtp_handler_init(s, rtsp_st, st);
323 /* parse the attribute line from the fmtp a line of an sdp response. This
324 * is broken out as a function because it is used in rtp_h264.c, which is
326 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
327 char *value, int value_size)
329 *p += strspn(*p, SPACE_CHARS);
331 get_word_sep(attr, attr_size, "=", p);
334 get_word_sep(value, value_size, ";", p);
342 typedef struct SDPParseState {
344 struct sockaddr_storage default_ip;
346 int skip_media; ///< set if an unknown m= line occurs
347 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
348 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
349 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
350 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
353 char delayed_fmtp[2048];
356 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
357 struct RTSPSource ***dest, int *dest_count)
359 RTSPSource *rtsp_src, *rtsp_src2;
361 for (i = 0; i < count; i++) {
363 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
366 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
367 dynarray_add(dest, dest_count, rtsp_src2);
371 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
372 int payload_type, const char *line)
376 for (i = 0; i < rt->nb_rtsp_streams; i++) {
377 RTSPStream *rtsp_st = rt->rtsp_streams[i];
378 if (rtsp_st->sdp_payload_type == payload_type &&
379 rtsp_st->dynamic_handler &&
380 rtsp_st->dynamic_handler->parse_sdp_a_line) {
381 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
382 rtsp_st->dynamic_protocol_context, line);
387 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
388 int letter, const char *buf)
390 RTSPState *rt = s->priv_data;
391 char buf1[64], st_type[64];
393 enum AVMediaType codec_type;
397 RTSPSource *rtsp_src;
398 struct sockaddr_storage sdp_ip;
401 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
404 if (s1->skip_media && letter != 'm')
408 get_word(buf1, sizeof(buf1), &p);
409 if (strcmp(buf1, "IN") != 0)
411 get_word(buf1, sizeof(buf1), &p);
412 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
414 get_word_sep(buf1, sizeof(buf1), "/", &p);
415 if (get_sockaddr(s, buf1, &sdp_ip))
420 get_word_sep(buf1, sizeof(buf1), "/", &p);
423 if (s->nb_streams == 0) {
424 s1->default_ip = sdp_ip;
425 s1->default_ttl = ttl;
427 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
428 rtsp_st->sdp_ip = sdp_ip;
429 rtsp_st->sdp_ttl = ttl;
433 av_dict_set(&s->metadata, "title", p, 0);
436 if (s->nb_streams == 0) {
437 av_dict_set(&s->metadata, "comment", p, 0);
446 codec_type = AVMEDIA_TYPE_UNKNOWN;
447 get_word(st_type, sizeof(st_type), &p);
448 if (!strcmp(st_type, "audio")) {
449 codec_type = AVMEDIA_TYPE_AUDIO;
450 } else if (!strcmp(st_type, "video")) {
451 codec_type = AVMEDIA_TYPE_VIDEO;
452 } else if (!strcmp(st_type, "application")) {
453 codec_type = AVMEDIA_TYPE_DATA;
454 } else if (!strcmp(st_type, "text")) {
455 codec_type = AVMEDIA_TYPE_SUBTITLE;
457 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
461 rtsp_st = av_mallocz(sizeof(RTSPStream));
464 rtsp_st->stream_index = -1;
465 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
467 rtsp_st->sdp_ip = s1->default_ip;
468 rtsp_st->sdp_ttl = s1->default_ttl;
470 copy_default_source_addrs(s1->default_include_source_addrs,
471 s1->nb_default_include_source_addrs,
472 &rtsp_st->include_source_addrs,
473 &rtsp_st->nb_include_source_addrs);
474 copy_default_source_addrs(s1->default_exclude_source_addrs,
475 s1->nb_default_exclude_source_addrs,
476 &rtsp_st->exclude_source_addrs,
477 &rtsp_st->nb_exclude_source_addrs);
479 get_word(buf1, sizeof(buf1), &p); /* port */
480 rtsp_st->sdp_port = atoi(buf1);
482 get_word(buf1, sizeof(buf1), &p); /* protocol */
483 if (!strcmp(buf1, "udp"))
484 rt->transport = RTSP_TRANSPORT_RAW;
485 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
486 rtsp_st->feedback = 1;
488 /* XXX: handle list of formats */
489 get_word(buf1, sizeof(buf1), &p); /* format list */
490 rtsp_st->sdp_payload_type = atoi(buf1);
492 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
493 /* no corresponding stream */
494 if (rt->transport == RTSP_TRANSPORT_RAW) {
495 if (CONFIG_RTPDEC && !rt->ts)
496 rt->ts = avpriv_mpegts_parse_open(s);
498 const RTPDynamicProtocolHandler *handler;
499 handler = ff_rtp_handler_find_by_id(
500 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
501 init_rtp_handler(handler, rtsp_st, NULL);
502 finalize_rtp_handler_init(s, rtsp_st, NULL);
504 } else if (rt->server_type == RTSP_SERVER_WMS &&
505 codec_type == AVMEDIA_TYPE_DATA) {
506 /* RTX stream, a stream that carries all the other actual
507 * audio/video streams. Don't expose this to the callers. */
509 st = avformat_new_stream(s, NULL);
512 st->id = rt->nb_rtsp_streams - 1;
513 rtsp_st->stream_index = st->index;
514 st->codecpar->codec_type = codec_type;
515 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
516 const RTPDynamicProtocolHandler *handler;
517 /* if standard payload type, we can find the codec right now */
518 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
519 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
520 st->codecpar->sample_rate > 0)
521 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
522 /* Even static payload types may need a custom depacketizer */
523 handler = ff_rtp_handler_find_by_id(
524 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
525 init_rtp_handler(handler, rtsp_st, st);
526 finalize_rtp_handler_init(s, rtsp_st, st);
528 if (rt->default_lang[0])
529 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
531 /* put a default control url */
532 av_strlcpy(rtsp_st->control_url, rt->control_uri,
533 sizeof(rtsp_st->control_url));
536 if (av_strstart(p, "control:", &p)) {
537 if (s->nb_streams == 0) {
538 if (!strncmp(p, "rtsp://", 7))
539 av_strlcpy(rt->control_uri, p,
540 sizeof(rt->control_uri));
543 /* get the control url */
544 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
546 /* XXX: may need to add full url resolution */
547 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
549 if (proto[0] == '\0') {
550 /* relative control URL */
551 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
552 av_strlcat(rtsp_st->control_url, "/",
553 sizeof(rtsp_st->control_url));
554 av_strlcat(rtsp_st->control_url, p,
555 sizeof(rtsp_st->control_url));
557 av_strlcpy(rtsp_st->control_url, p,
558 sizeof(rtsp_st->control_url));
560 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
561 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
562 get_word(buf1, sizeof(buf1), &p);
563 payload_type = atoi(buf1);
564 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
565 if (rtsp_st->stream_index >= 0) {
566 st = s->streams[rtsp_st->stream_index];
567 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
571 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
573 } else if (av_strstart(p, "fmtp:", &p) ||
574 av_strstart(p, "framesize:", &p)) {
575 // let dynamic protocol handlers have a stab at the line.
576 get_word(buf1, sizeof(buf1), &p);
577 payload_type = atoi(buf1);
578 if (s1->seen_rtpmap) {
579 parse_fmtp(s, rt, payload_type, buf);
582 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
584 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
585 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
586 get_word(buf1, sizeof(buf1), &p);
587 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
588 } else if (av_strstart(p, "range:", &p)) {
591 // this is so that seeking on a streamed file can work.
592 rtsp_parse_range_npt(p, &start, &end);
593 s->start_time = start;
594 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
595 s->duration = (end == AV_NOPTS_VALUE) ?
596 AV_NOPTS_VALUE : end - start;
597 } else if (av_strstart(p, "lang:", &p)) {
598 if (s->nb_streams > 0) {
599 get_word(buf1, sizeof(buf1), &p);
600 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
601 if (rtsp_st->stream_index >= 0) {
602 st = s->streams[rtsp_st->stream_index];
603 av_dict_set(&st->metadata, "language", buf1, 0);
606 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
607 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
609 rt->transport = RTSP_TRANSPORT_RDT;
610 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
612 st = s->streams[s->nb_streams - 1];
613 st->codecpar->sample_rate = atoi(p);
614 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
616 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
617 get_word(buf1, sizeof(buf1), &p); // ignore tag
618 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
619 p += strspn(p, SPACE_CHARS);
620 if (av_strstart(p, "inline:", &p))
621 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
622 } else if (av_strstart(p, "source-filter:", &p)) {
624 get_word(buf1, sizeof(buf1), &p);
625 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
627 exclude = !strcmp(buf1, "excl");
629 get_word(buf1, sizeof(buf1), &p);
630 if (strcmp(buf1, "IN") != 0)
632 get_word(buf1, sizeof(buf1), &p);
633 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
635 // not checking that the destination address actually matches or is wildcard
636 get_word(buf1, sizeof(buf1), &p);
639 rtsp_src = av_mallocz(sizeof(*rtsp_src));
642 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
644 if (s->nb_streams == 0) {
645 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
647 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
648 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
651 if (s->nb_streams == 0) {
652 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
654 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
655 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
660 if (rt->server_type == RTSP_SERVER_WMS)
661 ff_wms_parse_sdp_a_line(s, p);
662 if (s->nb_streams > 0) {
663 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
665 if (rt->server_type == RTSP_SERVER_REAL)
666 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
668 if (rtsp_st->dynamic_handler &&
669 rtsp_st->dynamic_handler->parse_sdp_a_line)
670 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
671 rtsp_st->stream_index,
672 rtsp_st->dynamic_protocol_context, buf);
679 int ff_sdp_parse(AVFormatContext *s, const char *content)
683 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
684 * contain long SDP lines containing complete ASF Headers (several
685 * kB) or arrays of MDPR (RM stream descriptor) headers plus
686 * "rulebooks" describing their properties. Therefore, the SDP line
689 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
690 * in rtpdec_xiph.c. */
692 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
696 p += strspn(p, SPACE_CHARS);
704 /* get the content */
706 while (*p != '\n' && *p != '\r' && *p != '\0') {
707 if ((q - buf) < sizeof(buf) - 1)
712 sdp_parse_line(s, s1, letter, buf);
714 while (*p != '\n' && *p != '\0')
720 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
721 av_freep(&s1->default_include_source_addrs[i]);
722 av_freep(&s1->default_include_source_addrs);
723 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
724 av_freep(&s1->default_exclude_source_addrs[i]);
725 av_freep(&s1->default_exclude_source_addrs);
729 #endif /* CONFIG_RTPDEC */
731 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
733 RTSPState *rt = s->priv_data;
736 for (i = 0; i < rt->nb_rtsp_streams; i++) {
737 RTSPStream *rtsp_st = rt->rtsp_streams[i];
740 if (rtsp_st->transport_priv) {
742 AVFormatContext *rtpctx = rtsp_st->transport_priv;
743 av_write_trailer(rtpctx);
744 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
745 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
746 ff_rtsp_tcp_write_packet(s, rtsp_st);
747 ffio_free_dyn_buf(&rtpctx->pb);
749 avio_closep(&rtpctx->pb);
751 avformat_free_context(rtpctx);
752 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
753 ff_rdt_parse_close(rtsp_st->transport_priv);
754 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
755 ff_rtp_parse_close(rtsp_st->transport_priv);
757 rtsp_st->transport_priv = NULL;
758 if (rtsp_st->rtp_handle)
759 ffurl_close(rtsp_st->rtp_handle);
760 rtsp_st->rtp_handle = NULL;
764 /* close and free RTSP streams */
765 void ff_rtsp_close_streams(AVFormatContext *s)
767 RTSPState *rt = s->priv_data;
771 ff_rtsp_undo_setup(s, 0);
772 for (i = 0; i < rt->nb_rtsp_streams; i++) {
773 rtsp_st = rt->rtsp_streams[i];
775 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
776 if (rtsp_st->dynamic_handler->close)
777 rtsp_st->dynamic_handler->close(
778 rtsp_st->dynamic_protocol_context);
779 av_free(rtsp_st->dynamic_protocol_context);
781 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
782 av_freep(&rtsp_st->include_source_addrs[j]);
783 av_freep(&rtsp_st->include_source_addrs);
784 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
785 av_freep(&rtsp_st->exclude_source_addrs[j]);
786 av_freep(&rtsp_st->exclude_source_addrs);
791 av_freep(&rt->rtsp_streams);
793 avformat_close_input(&rt->asf_ctx);
795 if (CONFIG_RTPDEC && rt->ts)
796 avpriv_mpegts_parse_close(rt->ts);
798 av_freep(&rt->recvbuf);
801 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
803 RTSPState *rt = s->priv_data;
805 int reordering_queue_size = rt->reordering_queue_size;
806 if (reordering_queue_size < 0) {
807 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
808 reordering_queue_size = 0;
810 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
813 /* open the RTP context */
814 if (rtsp_st->stream_index >= 0)
815 st = s->streams[rtsp_st->stream_index];
817 s->ctx_flags |= AVFMTCTX_NOHEADER;
819 if (CONFIG_RTSP_MUXER && s->oformat && st) {
820 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
821 s, st, rtsp_st->rtp_handle,
822 RTSP_TCP_MAX_PACKET_SIZE,
823 rtsp_st->stream_index);
824 /* Ownership of rtp_handle is passed to the rtp mux context */
825 rtsp_st->rtp_handle = NULL;
828 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
829 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
830 return 0; // Don't need to open any parser here
831 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
832 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
833 rtsp_st->dynamic_protocol_context,
834 rtsp_st->dynamic_handler);
835 else if (CONFIG_RTPDEC)
836 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
837 rtsp_st->sdp_payload_type,
838 reordering_queue_size);
840 if (!rtsp_st->transport_priv) {
841 return AVERROR(ENOMEM);
842 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
844 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
845 rtpctx->ssrc = rtsp_st->ssrc;
846 if (rtsp_st->dynamic_handler) {
847 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
848 rtsp_st->dynamic_protocol_context,
849 rtsp_st->dynamic_handler);
851 if (rtsp_st->crypto_suite[0])
852 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
853 rtsp_st->crypto_suite,
854 rtsp_st->crypto_params);
860 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
861 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
868 q += strspn(q, SPACE_CHARS);
869 v = strtol(q, &p, 10);
873 v = strtol(p, &p, 10);
882 /* XXX: only one transport specification is parsed */
883 static void rtsp_parse_transport(AVFormatContext *s,
884 RTSPMessageHeader *reply, const char *p)
886 char transport_protocol[16];
888 char lower_transport[16];
890 RTSPTransportField *th;
893 reply->nb_transports = 0;
896 p += strspn(p, SPACE_CHARS);
900 th = &reply->transports[reply->nb_transports];
902 get_word_sep(transport_protocol, sizeof(transport_protocol),
904 if (!av_strcasecmp (transport_protocol, "rtp")) {
905 get_word_sep(profile, sizeof(profile), "/;,", &p);
906 lower_transport[0] = '\0';
907 /* rtp/avp/<protocol> */
909 get_word_sep(lower_transport, sizeof(lower_transport),
912 th->transport = RTSP_TRANSPORT_RTP;
913 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
914 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
915 /* x-pn-tng/<protocol> */
916 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
918 th->transport = RTSP_TRANSPORT_RDT;
919 } else if (!av_strcasecmp(transport_protocol, "raw")) {
920 get_word_sep(profile, sizeof(profile), "/;,", &p);
921 lower_transport[0] = '\0';
922 /* raw/raw/<protocol> */
924 get_word_sep(lower_transport, sizeof(lower_transport),
927 th->transport = RTSP_TRANSPORT_RAW;
929 if (!av_strcasecmp(lower_transport, "TCP"))
930 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
932 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
936 /* get each parameter */
937 while (*p != '\0' && *p != ',') {
938 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
939 if (!strcmp(parameter, "port")) {
942 rtsp_parse_range(&th->port_min, &th->port_max, &p);
944 } else if (!strcmp(parameter, "client_port")) {
947 rtsp_parse_range(&th->client_port_min,
948 &th->client_port_max, &p);
950 } else if (!strcmp(parameter, "server_port")) {
953 rtsp_parse_range(&th->server_port_min,
954 &th->server_port_max, &p);
956 } else if (!strcmp(parameter, "interleaved")) {
959 rtsp_parse_range(&th->interleaved_min,
960 &th->interleaved_max, &p);
962 } else if (!strcmp(parameter, "multicast")) {
963 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
964 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
965 } else if (!strcmp(parameter, "ttl")) {
969 th->ttl = strtol(p, &end, 10);
972 } else if (!strcmp(parameter, "destination")) {
975 get_word_sep(buf, sizeof(buf), ";,", &p);
976 get_sockaddr(s, buf, &th->destination);
978 } else if (!strcmp(parameter, "source")) {
981 get_word_sep(buf, sizeof(buf), ";,", &p);
982 av_strlcpy(th->source, buf, sizeof(th->source));
984 } else if (!strcmp(parameter, "mode")) {
987 get_word_sep(buf, sizeof(buf), ";, ", &p);
988 if (!strcmp(buf, "record") ||
989 !strcmp(buf, "receive"))
994 while (*p != ';' && *p != '\0' && *p != ',')
1002 reply->nb_transports++;
1003 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1008 static void handle_rtp_info(RTSPState *rt, const char *url,
1009 uint32_t seq, uint32_t rtptime)
1012 if (!rtptime || !url[0])
1014 if (rt->transport != RTSP_TRANSPORT_RTP)
1016 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1017 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1018 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1021 if (!strcmp(rtsp_st->control_url, url)) {
1022 rtpctx->base_timestamp = rtptime;
1028 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1031 char key[20], value[1024], url[1024] = "";
1032 uint32_t seq = 0, rtptime = 0;
1035 p += strspn(p, SPACE_CHARS);
1038 get_word_sep(key, sizeof(key), "=", &p);
1042 get_word_sep(value, sizeof(value), ";, ", &p);
1044 if (!strcmp(key, "url"))
1045 av_strlcpy(url, value, sizeof(url));
1046 else if (!strcmp(key, "seq"))
1047 seq = strtoul(value, NULL, 10);
1048 else if (!strcmp(key, "rtptime"))
1049 rtptime = strtoul(value, NULL, 10);
1051 handle_rtp_info(rt, url, seq, rtptime);
1060 handle_rtp_info(rt, url, seq, rtptime);
1063 void ff_rtsp_parse_line(AVFormatContext *s,
1064 RTSPMessageHeader *reply, const char *buf,
1065 RTSPState *rt, const char *method)
1069 /* NOTE: we do case independent match for broken servers */
1071 if (av_stristart(p, "Session:", &p)) {
1073 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1074 if (av_stristart(p, ";timeout=", &p) &&
1075 (t = strtol(p, NULL, 10)) > 0) {
1078 } else if (av_stristart(p, "Content-Length:", &p)) {
1079 reply->content_length = strtol(p, NULL, 10);
1080 } else if (av_stristart(p, "Transport:", &p)) {
1081 rtsp_parse_transport(s, reply, p);
1082 } else if (av_stristart(p, "CSeq:", &p)) {
1083 reply->seq = strtol(p, NULL, 10);
1084 } else if (av_stristart(p, "Range:", &p)) {
1085 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1086 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1087 p += strspn(p, SPACE_CHARS);
1088 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1089 } else if (av_stristart(p, "Server:", &p)) {
1090 p += strspn(p, SPACE_CHARS);
1091 av_strlcpy(reply->server, p, sizeof(reply->server));
1092 } else if (av_stristart(p, "Notice:", &p) ||
1093 av_stristart(p, "X-Notice:", &p)) {
1094 reply->notice = strtol(p, NULL, 10);
1095 } else if (av_stristart(p, "Location:", &p)) {
1096 p += strspn(p, SPACE_CHARS);
1097 av_strlcpy(reply->location, p , sizeof(reply->location));
1098 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1099 p += strspn(p, SPACE_CHARS);
1100 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1101 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1102 p += strspn(p, SPACE_CHARS);
1103 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1104 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1105 p += strspn(p, SPACE_CHARS);
1106 if (method && !strcmp(method, "DESCRIBE"))
1107 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1108 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1109 p += strspn(p, SPACE_CHARS);
1110 if (method && !strcmp(method, "PLAY"))
1111 rtsp_parse_rtp_info(rt, p);
1112 } else if (av_stristart(p, "Public:", &p) && rt) {
1113 if (strstr(p, "GET_PARAMETER") &&
1114 method && !strcmp(method, "OPTIONS"))
1115 rt->get_parameter_supported = 1;
1116 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1117 p += strspn(p, SPACE_CHARS);
1118 rt->accept_dynamic_rate = atoi(p);
1119 } else if (av_stristart(p, "Content-Type:", &p)) {
1120 p += strspn(p, SPACE_CHARS);
1121 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1125 /* skip a RTP/TCP interleaved packet */
1126 void ff_rtsp_skip_packet(AVFormatContext *s)
1128 RTSPState *rt = s->priv_data;
1132 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1135 len = AV_RB16(buf + 1);
1137 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1142 if (len1 > sizeof(buf))
1144 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1151 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1152 unsigned char **content_ptr,
1153 int return_on_interleaved_data, const char *method)
1155 RTSPState *rt = s->priv_data;
1156 char buf[4096], buf1[1024], *q;
1159 int ret, content_length, line_count = 0, request = 0;
1160 unsigned char *content = NULL;
1166 memset(reply, 0, sizeof(*reply));
1168 /* parse reply (XXX: use buffers) */
1169 rt->last_reply[0] = '\0';
1173 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1174 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1179 if (ch == '$' && q == buf) {
1180 if (return_on_interleaved_data) {
1183 ff_rtsp_skip_packet(s);
1184 } else if (ch != '\r') {
1185 if ((q - buf) < sizeof(buf) - 1)
1191 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1193 /* test if last line */
1197 if (line_count == 0) {
1198 /* get reply code */
1199 get_word(buf1, sizeof(buf1), &p);
1200 if (!strncmp(buf1, "RTSP/", 5)) {
1201 get_word(buf1, sizeof(buf1), &p);
1202 reply->status_code = atoi(buf1);
1203 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1205 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1206 get_word(buf1, sizeof(buf1), &p); // object
1210 ff_rtsp_parse_line(s, reply, p, rt, method);
1211 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1212 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1217 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1218 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1220 content_length = reply->content_length;
1221 if (content_length > 0) {
1222 /* leave some room for a trailing '\0' (useful for simple parsing) */
1223 content = av_malloc(content_length + 1);
1225 return AVERROR(ENOMEM);
1226 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1227 content[content_length] = '\0';
1230 *content_ptr = content;
1236 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1237 const char* ptr = buf;
1239 if (!strcmp(reply->reason, "OPTIONS")) {
1240 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1242 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1243 if (reply->session_id[0])
1244 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1247 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1249 av_strlcat(buf, "\r\n", sizeof(buf));
1251 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1252 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1255 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1257 rt->last_cmd_time = av_gettime_relative();
1258 /* Even if the request from the server had data, it is not the data
1259 * that the caller wants or expects. The memory could also be leaked
1260 * if the actual following reply has content data. */
1262 av_freep(content_ptr);
1263 /* If method is set, this is called from ff_rtsp_send_cmd,
1264 * where a reply to exactly this request is awaited. For
1265 * callers from within packet receiving, we just want to
1266 * return to the caller and go back to receiving packets. */
1272 if (rt->seq != reply->seq) {
1273 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1274 rt->seq, reply->seq);
1278 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1279 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1280 reply->notice == 2306 /* Continuous Feed Terminated */) {
1281 rt->state = RTSP_STATE_IDLE;
1282 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1283 return AVERROR(EIO); /* data or server error */
1284 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1285 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1286 return AVERROR(EPERM);
1292 * Send a command to the RTSP server without waiting for the reply.
1294 * @param s RTSP (de)muxer context
1295 * @param method the method for the request
1296 * @param url the target url for the request
1297 * @param headers extra header lines to include in the request
1298 * @param send_content if non-null, the data to send as request body content
1299 * @param send_content_length the length of the send_content data, or 0 if
1300 * send_content is null
1302 * @return zero if success, nonzero otherwise
1304 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1305 const char *method, const char *url,
1306 const char *headers,
1307 const unsigned char *send_content,
1308 int send_content_length)
1310 RTSPState *rt = s->priv_data;
1311 char buf[4096], *out_buf;
1312 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1314 /* Add in RTSP headers */
1317 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1319 av_strlcat(buf, headers, sizeof(buf));
1320 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1321 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1322 if (rt->session_id[0] != '\0' && (!headers ||
1323 !strstr(headers, "\nIf-Match:"))) {
1324 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1327 char *str = ff_http_auth_create_response(&rt->auth_state,
1328 rt->auth, url, method);
1330 av_strlcat(buf, str, sizeof(buf));
1333 if (send_content_length > 0 && send_content)
1334 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1335 av_strlcat(buf, "\r\n", sizeof(buf));
1337 /* base64 encode rtsp if tunneling */
1338 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1339 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1340 out_buf = base64buf;
1343 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1345 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1346 if (send_content_length > 0 && send_content) {
1347 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1348 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1349 return AVERROR_PATCHWELCOME;
1351 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1353 rt->last_cmd_time = av_gettime_relative();
1358 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1359 const char *url, const char *headers)
1361 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1364 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1365 const char *headers, RTSPMessageHeader *reply,
1366 unsigned char **content_ptr)
1368 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1369 content_ptr, NULL, 0);
1372 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1373 const char *method, const char *url,
1375 RTSPMessageHeader *reply,
1376 unsigned char **content_ptr,
1377 const unsigned char *send_content,
1378 int send_content_length)
1380 RTSPState *rt = s->priv_data;
1381 HTTPAuthType cur_auth_type;
1382 int ret, attempts = 0;
1385 cur_auth_type = rt->auth_state.auth_type;
1386 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1388 send_content_length)))
1391 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1395 if (reply->status_code == 401 &&
1396 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1397 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1400 if (reply->status_code > 400){
1401 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1405 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1411 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1412 int lower_transport, const char *real_challenge)
1414 RTSPState *rt = s->priv_data;
1415 int rtx = 0, j, i, err, interleave = 0, port_off;
1416 RTSPStream *rtsp_st;
1417 RTSPMessageHeader reply1, *reply = &reply1;
1419 const char *trans_pref;
1421 if (rt->transport == RTSP_TRANSPORT_RDT)
1422 trans_pref = "x-pn-tng";
1423 else if (rt->transport == RTSP_TRANSPORT_RAW)
1424 trans_pref = "RAW/RAW";
1426 trans_pref = "RTP/AVP";
1428 /* default timeout: 1 minute */
1431 /* Choose a random starting offset within the first half of the
1432 * port range, to allow for a number of ports to try even if the offset
1433 * happens to be at the end of the random range. */
1434 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1435 /* even random offset */
1436 port_off -= port_off & 0x01;
1438 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1439 char transport[2048];
1442 * WMS serves all UDP data over a single connection, the RTX, which
1443 * isn't necessarily the first in the SDP but has to be the first
1444 * to be set up, else the second/third SETUP will fail with a 461.
1446 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1447 rt->server_type == RTSP_SERVER_WMS) {
1450 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1451 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1453 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1457 if (rtx == rt->nb_rtsp_streams)
1458 return -1; /* no RTX found */
1459 rtsp_st = rt->rtsp_streams[rtx];
1461 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1463 rtsp_st = rt->rtsp_streams[i];
1466 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1469 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1470 port = reply->transports[0].client_port_min;
1474 /* first try in specified port range */
1475 while (j <= rt->rtp_port_max) {
1476 AVDictionary *opts = map_to_opts(rt);
1478 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1479 "?localport=%d", j);
1480 /* we will use two ports per rtp stream (rtp and rtcp) */
1482 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1483 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1485 av_dict_free(&opts);
1490 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1495 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1497 snprintf(transport, sizeof(transport) - 1,
1498 "%s/UDP;", trans_pref);
1499 if (rt->server_type != RTSP_SERVER_REAL)
1500 av_strlcat(transport, "unicast;", sizeof(transport));
1501 av_strlcatf(transport, sizeof(transport),
1502 "client_port=%d", port);
1503 if (rt->transport == RTSP_TRANSPORT_RTP &&
1504 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1505 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1509 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1510 /* For WMS streams, the application streams are only used for
1511 * UDP. When trying to set it up for TCP streams, the server
1512 * will return an error. Therefore, we skip those streams. */
1513 if (rt->server_type == RTSP_SERVER_WMS &&
1514 (rtsp_st->stream_index < 0 ||
1515 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1518 snprintf(transport, sizeof(transport) - 1,
1519 "%s/TCP;", trans_pref);
1520 if (rt->transport != RTSP_TRANSPORT_RDT)
1521 av_strlcat(transport, "unicast;", sizeof(transport));
1522 av_strlcatf(transport, sizeof(transport),
1523 "interleaved=%d-%d",
1524 interleave, interleave + 1);
1528 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1529 snprintf(transport, sizeof(transport) - 1,
1530 "%s/UDP;multicast", trans_pref);
1533 av_strlcat(transport, ";mode=record", sizeof(transport));
1534 } else if (rt->server_type == RTSP_SERVER_REAL ||
1535 rt->server_type == RTSP_SERVER_WMS)
1536 av_strlcat(transport, ";mode=play", sizeof(transport));
1537 snprintf(cmd, sizeof(cmd),
1538 "Transport: %s\r\n",
1540 if (rt->accept_dynamic_rate)
1541 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1542 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1543 char real_res[41], real_csum[9];
1544 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1546 av_strlcatf(cmd, sizeof(cmd),
1548 "RealChallenge2: %s, sd=%s\r\n",
1549 rt->session_id, real_res, real_csum);
1551 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1552 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1555 } else if (reply->status_code != RTSP_STATUS_OK ||
1556 reply->nb_transports != 1) {
1557 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1561 /* XXX: same protocol for all streams is required */
1563 if (reply->transports[0].lower_transport != rt->lower_transport ||
1564 reply->transports[0].transport != rt->transport) {
1565 err = AVERROR_INVALIDDATA;
1569 rt->lower_transport = reply->transports[0].lower_transport;
1570 rt->transport = reply->transports[0].transport;
1573 /* Fail if the server responded with another lower transport mode
1574 * than what we requested. */
1575 if (reply->transports[0].lower_transport != lower_transport) {
1576 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1577 err = AVERROR_INVALIDDATA;
1581 switch(reply->transports[0].lower_transport) {
1582 case RTSP_LOWER_TRANSPORT_TCP:
1583 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1584 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1587 case RTSP_LOWER_TRANSPORT_UDP: {
1588 char url[1024], options[30] = "";
1589 const char *peer = host;
1591 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1592 av_strlcpy(options, "?connect=1", sizeof(options));
1593 /* Use source address if specified */
1594 if (reply->transports[0].source[0])
1595 peer = reply->transports[0].source;
1596 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1597 reply->transports[0].server_port_min, "%s", options);
1598 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1599 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1600 err = AVERROR_INVALIDDATA;
1605 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1606 char url[1024], namebuf[50], optbuf[20] = "";
1607 struct sockaddr_storage addr;
1610 if (reply->transports[0].destination.ss_family) {
1611 addr = reply->transports[0].destination;
1612 port = reply->transports[0].port_min;
1613 ttl = reply->transports[0].ttl;
1615 addr = rtsp_st->sdp_ip;
1616 port = rtsp_st->sdp_port;
1617 ttl = rtsp_st->sdp_ttl;
1620 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1621 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1622 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1623 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1624 port, "%s", optbuf);
1625 if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1626 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL) < 0) {
1627 err = AVERROR_INVALIDDATA;
1634 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1638 if (rt->nb_rtsp_streams && reply->timeout > 0)
1639 rt->timeout = reply->timeout;
1641 if (rt->server_type == RTSP_SERVER_REAL)
1642 rt->need_subscription = 1;
1647 ff_rtsp_undo_setup(s, 0);
1651 void ff_rtsp_close_connections(AVFormatContext *s)
1653 RTSPState *rt = s->priv_data;
1654 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1655 ffurl_close(rt->rtsp_hd);
1656 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1659 int ff_rtsp_connect(AVFormatContext *s)
1661 RTSPState *rt = s->priv_data;
1662 char proto[128], host[1024], path[1024];
1663 char tcpname[1024], cmd[2048], auth[128];
1664 const char *lower_rtsp_proto = "tcp";
1665 int port, err, tcp_fd;
1666 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1667 int lower_transport_mask = 0;
1668 int default_port = RTSP_DEFAULT_PORT;
1669 char real_challenge[64] = "";
1670 struct sockaddr_storage peer;
1671 socklen_t peer_len = sizeof(peer);
1673 if (rt->rtp_port_max < rt->rtp_port_min) {
1674 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1675 "than min port %d\n", rt->rtp_port_max,
1677 return AVERROR(EINVAL);
1680 if (!ff_network_init())
1681 return AVERROR(EIO);
1683 if (s->max_delay < 0) /* Not set by the caller */
1684 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1686 rt->control_transport = RTSP_MODE_PLAIN;
1687 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1688 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1689 rt->control_transport = RTSP_MODE_TUNNEL;
1691 /* Only pass through valid flags from here */
1692 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1695 /* extract hostname and port */
1696 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1697 host, sizeof(host), &port, path, sizeof(path), s->url);
1699 if (!strcmp(proto, "rtsps")) {
1700 lower_rtsp_proto = "tls";
1701 default_port = RTSPS_DEFAULT_PORT;
1702 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1706 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1709 port = default_port;
1711 lower_transport_mask = rt->lower_transport_mask;
1713 if (!lower_transport_mask)
1714 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1717 /* Only UDP or TCP - UDP multicast isn't supported. */
1718 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1719 (1 << RTSP_LOWER_TRANSPORT_TCP);
1720 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1721 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1722 "only UDP and TCP are supported for output.\n");
1723 err = AVERROR(EINVAL);
1728 /* Construct the URI used in request; this is similar to s->url,
1729 * but with authentication credentials removed and RTSP specific options
1731 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1732 host, port, "%s", path);
1734 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1735 /* set up initial handshake for tunneling */
1736 char httpname[1024];
1737 char sessioncookie[17];
1740 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1741 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1742 av_get_random_seed(), av_get_random_seed());
1745 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1746 &s->interrupt_callback) < 0) {
1751 /* generate GET headers */
1752 snprintf(headers, sizeof(headers),
1753 "x-sessioncookie: %s\r\n"
1754 "Accept: application/x-rtsp-tunnelled\r\n"
1755 "Pragma: no-cache\r\n"
1756 "Cache-Control: no-cache\r\n",
1758 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1760 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1761 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1762 if (!rt->rtsp_hd->protocol_whitelist) {
1763 err = AVERROR(ENOMEM);
1768 /* complete the connection */
1769 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1775 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1776 &s->interrupt_callback) < 0 ) {
1781 /* generate POST headers */
1782 snprintf(headers, sizeof(headers),
1783 "x-sessioncookie: %s\r\n"
1784 "Content-Type: application/x-rtsp-tunnelled\r\n"
1785 "Pragma: no-cache\r\n"
1786 "Cache-Control: no-cache\r\n"
1787 "Content-Length: 32767\r\n"
1788 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1790 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1791 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1793 /* Initialize the authentication state for the POST session. The HTTP
1794 * protocol implementation doesn't properly handle multi-pass
1795 * authentication for POST requests, since it would require one of
1797 * - implementing Expect: 100-continue, which many HTTP servers
1798 * don't support anyway, even less the RTSP servers that do HTTP
1800 * - sending the whole POST data until getting a 401 reply specifying
1801 * what authentication method to use, then resending all that data
1802 * - waiting for potential 401 replies directly after sending the
1803 * POST header (waiting for some unspecified time)
1804 * Therefore, we copy the full auth state, which works for both basic
1805 * and digest. (For digest, we would have to synchronize the nonce
1806 * count variable between the two sessions, if we'd do more requests
1807 * with the original session, though.)
1809 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1811 /* complete the connection */
1812 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1818 /* open the tcp connection */
1819 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1821 "?timeout=%d", rt->stimeout);
1822 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1823 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1827 rt->rtsp_hd_out = rt->rtsp_hd;
1831 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1836 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1837 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1838 NULL, 0, NI_NUMERICHOST);
1841 /* request options supported by the server; this also detects server
1843 for (rt->server_type = RTSP_SERVER_RTP;;) {
1845 if (rt->server_type == RTSP_SERVER_REAL)
1848 * The following entries are required for proper
1849 * streaming from a Realmedia server. They are
1850 * interdependent in some way although we currently
1851 * don't quite understand how. Values were copied
1852 * from mplayer SVN r23589.
1853 * ClientChallenge is a 16-byte ID in hex
1854 * CompanyID is a 16-byte ID in base64
1856 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1857 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1858 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1859 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1861 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1862 if (reply->status_code != RTSP_STATUS_OK) {
1863 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1867 /* detect server type if not standard-compliant RTP */
1868 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1869 rt->server_type = RTSP_SERVER_REAL;
1871 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1872 rt->server_type = RTSP_SERVER_WMS;
1873 } else if (rt->server_type == RTSP_SERVER_REAL)
1874 strcpy(real_challenge, reply->real_challenge);
1878 if (CONFIG_RTSP_DEMUXER && s->iformat)
1879 err = ff_rtsp_setup_input_streams(s, reply);
1880 else if (CONFIG_RTSP_MUXER)
1881 err = ff_rtsp_setup_output_streams(s, host);
1888 int lower_transport = ff_log2_tab[lower_transport_mask &
1889 ~(lower_transport_mask - 1)];
1891 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1892 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1893 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1895 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1896 rt->server_type == RTSP_SERVER_REAL ?
1897 real_challenge : NULL);
1900 lower_transport_mask &= ~(1 << lower_transport);
1901 if (lower_transport_mask == 0 && err == 1) {
1902 err = AVERROR(EPROTONOSUPPORT);
1907 rt->lower_transport_mask = lower_transport_mask;
1908 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1909 rt->state = RTSP_STATE_IDLE;
1910 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1913 ff_rtsp_close_streams(s);
1914 ff_rtsp_close_connections(s);
1915 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1916 char *new_url = av_strdup(reply->location);
1918 err = AVERROR(ENOMEM);
1921 ff_format_set_url(s, new_url);
1922 rt->session_id[0] = '\0';
1923 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1932 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1935 static int parse_rtsp_message(AVFormatContext *s)
1937 RTSPState *rt = s->priv_data;
1940 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1941 if (rt->state == RTSP_STATE_STREAMING) {
1942 if (!ff_rtsp_parse_streaming_commands(s))
1945 av_log(s, AV_LOG_WARNING,
1946 "Unable to answer to TEARDOWN\n");
1950 RTSPMessageHeader reply;
1951 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1954 /* XXX: parse message */
1955 if (rt->state != RTSP_STATE_STREAMING)
1962 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1963 uint8_t *buf, int buf_size, int64_t wait_end)
1965 RTSPState *rt = s->priv_data;
1966 RTSPStream *rtsp_st;
1967 int n, i, ret, timeout_cnt = 0;
1968 struct pollfd *p = rt->p;
1969 int *fds = NULL, fdsnum, fdsidx;
1972 p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
1974 return AVERROR(ENOMEM);
1977 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1978 p[rt->max_p++].events = POLLIN;
1980 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1981 rtsp_st = rt->rtsp_streams[i];
1982 if (rtsp_st->rtp_handle) {
1983 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1985 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1989 av_log(s, AV_LOG_ERROR,
1990 "Number of fds %d not supported\n", fdsnum);
1991 return AVERROR_INVALIDDATA;
1993 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1994 p[rt->max_p].fd = fds[fdsidx];
1995 p[rt->max_p++].events = POLLIN;
2003 if (ff_check_interrupt(&s->interrupt_callback))
2004 return AVERROR_EXIT;
2005 if (wait_end && wait_end - av_gettime_relative() < 0)
2006 return AVERROR(EAGAIN);
2007 n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2009 int j = rt->rtsp_hd ? 1 : 0;
2011 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2012 rtsp_st = rt->rtsp_streams[i];
2013 if (rtsp_st->rtp_handle) {
2014 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2015 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2017 *prtsp_st = rtsp_st;
2024 #if CONFIG_RTSP_DEMUXER
2025 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2026 if ((ret = parse_rtsp_message(s)) < 0) {
2031 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2032 return AVERROR(ETIMEDOUT);
2033 } else if (n < 0 && errno != EINTR)
2034 return AVERROR(errno);
2038 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2039 const uint8_t *buf, int len)
2041 RTSPState *rt = s->priv_data;
2045 if (rt->nb_rtsp_streams == 1) {
2046 *rtsp_st = rt->rtsp_streams[0];
2049 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2050 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2052 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2053 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2056 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2057 *rtsp_st = rt->rtsp_streams[i];
2064 av_log(s, AV_LOG_WARNING,
2065 "Unable to pick stream for packet - SSRC not known for "
2067 return AVERROR(EAGAIN);
2070 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2071 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2072 *rtsp_st = rt->rtsp_streams[i];
2078 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2079 return AVERROR(EAGAIN);
2082 static int read_packet(AVFormatContext *s,
2083 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2086 RTSPState *rt = s->priv_data;
2089 switch(rt->lower_transport) {
2091 #if CONFIG_RTSP_DEMUXER
2092 case RTSP_LOWER_TRANSPORT_TCP:
2093 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2096 case RTSP_LOWER_TRANSPORT_UDP:
2097 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2098 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2099 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2100 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2102 case RTSP_LOWER_TRANSPORT_CUSTOM:
2103 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2104 wait_end && wait_end < av_gettime_relative())
2105 len = AVERROR(EAGAIN);
2107 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2108 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2109 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2110 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2120 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2122 RTSPState *rt = s->priv_data;
2124 RTSPStream *rtsp_st, *first_queue_st = NULL;
2125 int64_t wait_end = 0;
2127 if (rt->nb_byes == rt->nb_rtsp_streams)
2130 /* get next frames from the same RTP packet */
2131 if (rt->cur_transport_priv) {
2132 if (rt->transport == RTSP_TRANSPORT_RDT) {
2133 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2134 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2135 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2136 } else if (CONFIG_RTPDEC && rt->ts) {
2137 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2139 rt->recvbuf_pos += ret;
2140 ret = rt->recvbuf_pos < rt->recvbuf_len;
2145 rt->cur_transport_priv = NULL;
2147 } else if (ret == 1) {
2150 rt->cur_transport_priv = NULL;
2154 if (rt->transport == RTSP_TRANSPORT_RTP) {
2156 int64_t first_queue_time = 0;
2157 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2158 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2162 queue_time = ff_rtp_queued_packet_time(rtpctx);
2163 if (queue_time && (queue_time - first_queue_time < 0 ||
2164 !first_queue_time)) {
2165 first_queue_time = queue_time;
2166 first_queue_st = rt->rtsp_streams[i];
2169 if (first_queue_time) {
2170 wait_end = first_queue_time + s->max_delay;
2173 first_queue_st = NULL;
2177 /* read next RTP packet */
2179 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2181 return AVERROR(ENOMEM);
2184 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2185 if (len == AVERROR(EAGAIN) && first_queue_st &&
2186 rt->transport == RTSP_TRANSPORT_RTP) {
2187 av_log(s, AV_LOG_WARNING,
2188 "max delay reached. need to consume packet\n");
2189 rtsp_st = first_queue_st;
2190 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2196 if (rt->transport == RTSP_TRANSPORT_RDT) {
2197 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2198 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2199 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2200 if (rtsp_st->feedback) {
2201 AVIOContext *pb = NULL;
2202 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2204 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2207 /* Either bad packet, or a RTCP packet. Check if the
2208 * first_rtcp_ntp_time field was initialized. */
2209 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2210 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2211 /* first_rtcp_ntp_time has been initialized for this stream,
2212 * copy the same value to all other uninitialized streams,
2213 * in order to map their timestamp origin to the same ntp time
2216 AVStream *st = NULL;
2217 if (rtsp_st->stream_index >= 0)
2218 st = s->streams[rtsp_st->stream_index];
2219 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2220 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2221 AVStream *st2 = NULL;
2222 if (rt->rtsp_streams[i]->stream_index >= 0)
2223 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2224 if (rtpctx2 && st && st2 &&
2225 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2226 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2227 rtpctx2->rtcp_ts_offset = av_rescale_q(
2228 rtpctx->rtcp_ts_offset, st->time_base,
2232 // Make real NTP start time available in AVFormatContext
2233 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2234 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2236 s->start_time_realtime -=
2237 av_rescale (rtpctx->rtcp_ts_offset,
2238 (uint64_t) rtpctx->st->time_base.num * 1000000,
2239 rtpctx->st->time_base.den);
2243 if (ret == -RTCP_BYE) {
2246 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2247 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2249 if (rt->nb_byes == rt->nb_rtsp_streams)
2253 } else if (CONFIG_RTPDEC && rt->ts) {
2254 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2257 rt->recvbuf_len = len;
2258 rt->recvbuf_pos = ret;
2259 rt->cur_transport_priv = rt->ts;
2266 return AVERROR_INVALIDDATA;
2272 /* more packets may follow, so we save the RTP context */
2273 rt->cur_transport_priv = rtsp_st->transport_priv;
2277 #endif /* CONFIG_RTPDEC */
2279 #if CONFIG_SDP_DEMUXER
2280 static int sdp_probe(AVProbeData *p1)
2282 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2284 /* we look for a line beginning "c=IN IP" */
2285 while (p < p_end && *p != '\0') {
2286 if (sizeof("c=IN IP") - 1 < p_end - p &&
2287 av_strstart(p, "c=IN IP", NULL))
2288 return AVPROBE_SCORE_EXTENSION;
2290 while (p < p_end - 1 && *p != '\n') p++;
2299 static void append_source_addrs(char *buf, int size, const char *name,
2300 int count, struct RTSPSource **addrs)
2305 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2306 for (i = 1; i < count; i++)
2307 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2310 static int sdp_read_header(AVFormatContext *s)
2312 RTSPState *rt = s->priv_data;
2313 RTSPStream *rtsp_st;
2318 if (!ff_network_init())
2319 return AVERROR(EIO);
2321 if (s->max_delay < 0) /* Not set by the caller */
2322 s->max_delay = DEFAULT_REORDERING_DELAY;
2323 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2324 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2326 /* read the whole sdp file */
2327 /* XXX: better loading */
2328 content = av_malloc(SDP_MAX_SIZE);
2330 return AVERROR(ENOMEM);
2331 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2334 return AVERROR_INVALIDDATA;
2336 content[size] ='\0';
2338 err = ff_sdp_parse(s, content);
2342 /* open each RTP stream */
2343 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2345 rtsp_st = rt->rtsp_streams[i];
2347 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2348 AVDictionary *opts = map_to_opts(rt);
2350 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2351 sizeof(rtsp_st->sdp_ip),
2352 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2354 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2356 av_dict_free(&opts);
2359 ff_url_join(url, sizeof(url), "rtp", NULL,
2360 namebuf, rtsp_st->sdp_port,
2361 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2362 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2363 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2364 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2366 append_source_addrs(url, sizeof(url), "sources",
2367 rtsp_st->nb_include_source_addrs,
2368 rtsp_st->include_source_addrs);
2369 append_source_addrs(url, sizeof(url), "block",
2370 rtsp_st->nb_exclude_source_addrs,
2371 rtsp_st->exclude_source_addrs);
2372 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2373 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2375 av_dict_free(&opts);
2378 err = AVERROR_INVALIDDATA;
2382 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2387 ff_rtsp_close_streams(s);
2392 static int sdp_read_close(AVFormatContext *s)
2394 ff_rtsp_close_streams(s);
2399 static const AVClass sdp_demuxer_class = {
2400 .class_name = "SDP demuxer",
2401 .item_name = av_default_item_name,
2402 .option = sdp_options,
2403 .version = LIBAVUTIL_VERSION_INT,
2406 AVInputFormat ff_sdp_demuxer = {
2408 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2409 .priv_data_size = sizeof(RTSPState),
2410 .read_probe = sdp_probe,
2411 .read_header = sdp_read_header,
2412 .read_packet = ff_rtsp_fetch_packet,
2413 .read_close = sdp_read_close,
2414 .priv_class = &sdp_demuxer_class,
2416 #endif /* CONFIG_SDP_DEMUXER */
2418 #if CONFIG_RTP_DEMUXER
2419 static int rtp_probe(AVProbeData *p)
2421 if (av_strstart(p->filename, "rtp:", NULL))
2422 return AVPROBE_SCORE_MAX;
2426 static int rtp_read_header(AVFormatContext *s)
2428 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2429 char host[500], sdp[500];
2431 URLContext* in = NULL;
2433 AVCodecParameters *par = NULL;
2434 struct sockaddr_storage addr;
2436 socklen_t addrlen = sizeof(addr);
2437 RTSPState *rt = s->priv_data;
2439 if (!ff_network_init())
2440 return AVERROR(EIO);
2442 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2443 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2448 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2449 if (ret == AVERROR(EAGAIN))
2454 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2458 if ((recvbuf[0] & 0xc0) != 0x80) {
2459 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2464 if (RTP_PT_IS_RTCP(recvbuf[1]))
2467 payload_type = recvbuf[1] & 0x7f;
2470 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2474 par = avcodec_parameters_alloc();
2476 ret = AVERROR(ENOMEM);
2480 if (ff_rtp_get_codec_info(par, payload_type)) {
2481 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2482 "without an SDP file describing it\n",
2486 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2487 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2488 "properly you need an SDP file "
2492 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2495 snprintf(sdp, sizeof(sdp),
2496 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2497 addr.ss_family == AF_INET ? 4 : 6, host,
2498 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2499 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2500 port, payload_type);
2501 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2502 avcodec_parameters_free(&par);
2504 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2507 /* sdp_read_header initializes this again */
2510 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2512 ret = sdp_read_header(s);
2517 avcodec_parameters_free(&par);
2524 static const AVClass rtp_demuxer_class = {
2525 .class_name = "RTP demuxer",
2526 .item_name = av_default_item_name,
2527 .option = rtp_options,
2528 .version = LIBAVUTIL_VERSION_INT,
2531 AVInputFormat ff_rtp_demuxer = {
2533 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2534 .priv_data_size = sizeof(RTSPState),
2535 .read_probe = rtp_probe,
2536 .read_header = rtp_read_header,
2537 .read_packet = ff_rtsp_fetch_packet,
2538 .read_close = sdp_read_close,
2539 .flags = AVFMT_NOFILE,
2540 .priv_class = &rtp_demuxer_class,
2542 #endif /* CONFIG_RTP_DEMUXER */