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 RTSP_REORDERING_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 }
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
82 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
89 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
90 { "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" },
91 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
92 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
93 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
94 { "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 },
95 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
96 RTSP_REORDERING_OPTS(),
97 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
101 static const AVOption sdp_options[] = {
102 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
103 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
104 { "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" },
105 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
106 RTSP_REORDERING_OPTS(),
110 static const AVOption rtp_options[] = {
111 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
112 RTSP_REORDERING_OPTS(),
116 static void get_word_until_chars(char *buf, int buf_size,
117 const char *sep, const char **pp)
123 p += strspn(p, SPACE_CHARS);
125 while (!strchr(sep, *p) && *p != '\0') {
126 if ((q - buf) < buf_size - 1)
135 static void get_word_sep(char *buf, int buf_size, const char *sep,
138 if (**pp == '/') (*pp)++;
139 get_word_until_chars(buf, buf_size, sep, pp);
142 static void get_word(char *buf, int buf_size, const char **pp)
144 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
147 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
149 * Used for seeking in the rtp stream.
151 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
155 p += strspn(p, SPACE_CHARS);
156 if (!av_stristart(p, "npt=", &p))
159 *start = AV_NOPTS_VALUE;
160 *end = AV_NOPTS_VALUE;
162 get_word_sep(buf, sizeof(buf), "-", &p);
163 av_parse_time(start, buf, 1);
166 get_word_sep(buf, sizeof(buf), "-", &p);
167 av_parse_time(end, buf, 1);
171 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
173 struct addrinfo hints = { 0 }, *ai = NULL;
174 hints.ai_flags = AI_NUMERICHOST;
175 if (getaddrinfo(buf, NULL, &hints, &ai))
177 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
183 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
184 RTSPStream *rtsp_st, AVStream *st)
186 AVCodecContext *codec = st ? st->codec : NULL;
190 codec->codec_id = handler->codec_id;
191 rtsp_st->dynamic_handler = handler;
193 st->need_parsing = handler->need_parsing;
194 if (handler->priv_data_size) {
195 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
196 if (!rtsp_st->dynamic_protocol_context)
197 rtsp_st->dynamic_handler = NULL;
201 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
204 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
205 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
206 rtsp_st->dynamic_protocol_context);
208 if (rtsp_st->dynamic_protocol_context) {
209 if (rtsp_st->dynamic_handler->close)
210 rtsp_st->dynamic_handler->close(
211 rtsp_st->dynamic_protocol_context);
212 av_free(rtsp_st->dynamic_protocol_context);
214 rtsp_st->dynamic_protocol_context = NULL;
215 rtsp_st->dynamic_handler = NULL;
220 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
221 static int sdp_parse_rtpmap(AVFormatContext *s,
222 AVStream *st, RTSPStream *rtsp_st,
223 int payload_type, const char *p)
225 AVCodecContext *codec = st->codec;
231 /* See if we can handle this kind of payload.
232 * The space should normally not be there but some Real streams or
233 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
234 * have a trailing space. */
235 get_word_sep(buf, sizeof(buf), "/ ", &p);
236 if (payload_type < RTP_PT_PRIVATE) {
237 /* We are in a standard case
238 * (from http://www.iana.org/assignments/rtp-parameters). */
239 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
242 if (codec->codec_id == AV_CODEC_ID_NONE) {
243 RTPDynamicProtocolHandler *handler =
244 ff_rtp_handler_find_by_name(buf, codec->codec_type);
245 init_rtp_handler(handler, rtsp_st, st);
246 /* If no dynamic handler was found, check with the list of standard
247 * allocated types, if such a stream for some reason happens to
248 * use a private payload type. This isn't handled in rtpdec.c, since
249 * the format name from the rtpmap line never is passed into rtpdec. */
250 if (!rtsp_st->dynamic_handler)
251 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
254 c = avcodec_find_decoder(codec->codec_id);
260 get_word_sep(buf, sizeof(buf), "/", &p);
262 switch (codec->codec_type) {
263 case AVMEDIA_TYPE_AUDIO:
264 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
265 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
266 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
268 codec->sample_rate = i;
269 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
270 get_word_sep(buf, sizeof(buf), "/", &p);
275 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
277 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
280 case AVMEDIA_TYPE_VIDEO:
281 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
283 avpriv_set_pts_info(st, 32, 1, i);
288 finalize_rtp_handler_init(s, rtsp_st, st);
292 /* parse the attribute line from the fmtp a line of an sdp response. This
293 * is broken out as a function because it is used in rtp_h264.c, which is
295 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
296 char *value, int value_size)
298 *p += strspn(*p, SPACE_CHARS);
300 get_word_sep(attr, attr_size, "=", p);
303 get_word_sep(value, value_size, ";", p);
311 typedef struct SDPParseState {
313 struct sockaddr_storage default_ip;
315 int skip_media; ///< set if an unknown m= line occurs
316 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
317 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
318 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
319 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
322 char delayed_fmtp[2048];
325 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
326 struct RTSPSource ***dest, int *dest_count)
328 RTSPSource *rtsp_src, *rtsp_src2;
330 for (i = 0; i < count; i++) {
332 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
335 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
336 dynarray_add(dest, dest_count, rtsp_src2);
340 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
341 int payload_type, const char *line)
345 for (i = 0; i < rt->nb_rtsp_streams; i++) {
346 RTSPStream *rtsp_st = rt->rtsp_streams[i];
347 if (rtsp_st->sdp_payload_type == payload_type &&
348 rtsp_st->dynamic_handler &&
349 rtsp_st->dynamic_handler->parse_sdp_a_line) {
350 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
351 rtsp_st->dynamic_protocol_context, line);
356 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
357 int letter, const char *buf)
359 RTSPState *rt = s->priv_data;
360 char buf1[64], st_type[64];
362 enum AVMediaType codec_type;
366 RTSPSource *rtsp_src;
367 struct sockaddr_storage sdp_ip;
370 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
373 if (s1->skip_media && letter != 'm')
377 get_word(buf1, sizeof(buf1), &p);
378 if (strcmp(buf1, "IN") != 0)
380 get_word(buf1, sizeof(buf1), &p);
381 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
383 get_word_sep(buf1, sizeof(buf1), "/", &p);
384 if (get_sockaddr(buf1, &sdp_ip))
389 get_word_sep(buf1, sizeof(buf1), "/", &p);
392 if (s->nb_streams == 0) {
393 s1->default_ip = sdp_ip;
394 s1->default_ttl = ttl;
396 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
397 rtsp_st->sdp_ip = sdp_ip;
398 rtsp_st->sdp_ttl = ttl;
402 av_dict_set(&s->metadata, "title", p, 0);
405 if (s->nb_streams == 0) {
406 av_dict_set(&s->metadata, "comment", p, 0);
415 codec_type = AVMEDIA_TYPE_UNKNOWN;
416 get_word(st_type, sizeof(st_type), &p);
417 if (!strcmp(st_type, "audio")) {
418 codec_type = AVMEDIA_TYPE_AUDIO;
419 } else if (!strcmp(st_type, "video")) {
420 codec_type = AVMEDIA_TYPE_VIDEO;
421 } else if (!strcmp(st_type, "application")) {
422 codec_type = AVMEDIA_TYPE_DATA;
423 } else if (!strcmp(st_type, "text")) {
424 codec_type = AVMEDIA_TYPE_SUBTITLE;
426 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
430 rtsp_st = av_mallocz(sizeof(RTSPStream));
433 rtsp_st->stream_index = -1;
434 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
436 rtsp_st->sdp_ip = s1->default_ip;
437 rtsp_st->sdp_ttl = s1->default_ttl;
439 copy_default_source_addrs(s1->default_include_source_addrs,
440 s1->nb_default_include_source_addrs,
441 &rtsp_st->include_source_addrs,
442 &rtsp_st->nb_include_source_addrs);
443 copy_default_source_addrs(s1->default_exclude_source_addrs,
444 s1->nb_default_exclude_source_addrs,
445 &rtsp_st->exclude_source_addrs,
446 &rtsp_st->nb_exclude_source_addrs);
448 get_word(buf1, sizeof(buf1), &p); /* port */
449 rtsp_st->sdp_port = atoi(buf1);
451 get_word(buf1, sizeof(buf1), &p); /* protocol */
452 if (!strcmp(buf1, "udp"))
453 rt->transport = RTSP_TRANSPORT_RAW;
454 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
455 rtsp_st->feedback = 1;
457 /* XXX: handle list of formats */
458 get_word(buf1, sizeof(buf1), &p); /* format list */
459 rtsp_st->sdp_payload_type = atoi(buf1);
461 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
462 /* no corresponding stream */
463 if (rt->transport == RTSP_TRANSPORT_RAW) {
464 if (CONFIG_RTPDEC && !rt->ts)
465 rt->ts = avpriv_mpegts_parse_open(s);
467 RTPDynamicProtocolHandler *handler;
468 handler = ff_rtp_handler_find_by_id(
469 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
470 init_rtp_handler(handler, rtsp_st, NULL);
471 finalize_rtp_handler_init(s, rtsp_st, NULL);
473 } else if (rt->server_type == RTSP_SERVER_WMS &&
474 codec_type == AVMEDIA_TYPE_DATA) {
475 /* RTX stream, a stream that carries all the other actual
476 * audio/video streams. Don't expose this to the callers. */
478 st = avformat_new_stream(s, NULL);
481 st->id = rt->nb_rtsp_streams - 1;
482 rtsp_st->stream_index = st->index;
483 st->codec->codec_type = codec_type;
484 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
485 RTPDynamicProtocolHandler *handler;
486 /* if standard payload type, we can find the codec right now */
487 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
488 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
489 st->codec->sample_rate > 0)
490 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
491 /* Even static payload types may need a custom depacketizer */
492 handler = ff_rtp_handler_find_by_id(
493 rtsp_st->sdp_payload_type, st->codec->codec_type);
494 init_rtp_handler(handler, rtsp_st, st);
495 finalize_rtp_handler_init(s, rtsp_st, st);
497 if (rt->default_lang[0])
498 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
500 /* put a default control url */
501 av_strlcpy(rtsp_st->control_url, rt->control_uri,
502 sizeof(rtsp_st->control_url));
505 if (av_strstart(p, "control:", &p)) {
506 if (s->nb_streams == 0) {
507 if (!strncmp(p, "rtsp://", 7))
508 av_strlcpy(rt->control_uri, p,
509 sizeof(rt->control_uri));
512 /* get the control url */
513 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
515 /* XXX: may need to add full url resolution */
516 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
518 if (proto[0] == '\0') {
519 /* relative control URL */
520 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
521 av_strlcat(rtsp_st->control_url, "/",
522 sizeof(rtsp_st->control_url));
523 av_strlcat(rtsp_st->control_url, p,
524 sizeof(rtsp_st->control_url));
526 av_strlcpy(rtsp_st->control_url, p,
527 sizeof(rtsp_st->control_url));
529 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
530 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
531 get_word(buf1, sizeof(buf1), &p);
532 payload_type = atoi(buf1);
533 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
534 if (rtsp_st->stream_index >= 0) {
535 st = s->streams[rtsp_st->stream_index];
536 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
540 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
542 } else if (av_strstart(p, "fmtp:", &p) ||
543 av_strstart(p, "framesize:", &p)) {
544 // let dynamic protocol handlers have a stab at the line.
545 get_word(buf1, sizeof(buf1), &p);
546 payload_type = atoi(buf1);
547 if (s1->seen_rtpmap) {
548 parse_fmtp(s, rt, payload_type, buf);
551 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
553 } else if (av_strstart(p, "range:", &p)) {
556 // this is so that seeking on a streamed file can work.
557 rtsp_parse_range_npt(p, &start, &end);
558 s->start_time = start;
559 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
560 s->duration = (end == AV_NOPTS_VALUE) ?
561 AV_NOPTS_VALUE : end - start;
562 } else if (av_strstart(p, "lang:", &p)) {
563 if (s->nb_streams > 0) {
564 get_word(buf1, sizeof(buf1), &p);
565 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
566 if (rtsp_st->stream_index >= 0) {
567 st = s->streams[rtsp_st->stream_index];
568 av_dict_set(&st->metadata, "language", buf1, 0);
571 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
572 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
574 rt->transport = RTSP_TRANSPORT_RDT;
575 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
577 st = s->streams[s->nb_streams - 1];
578 st->codec->sample_rate = atoi(p);
579 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
581 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
582 get_word(buf1, sizeof(buf1), &p); // ignore tag
583 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
584 p += strspn(p, SPACE_CHARS);
585 if (av_strstart(p, "inline:", &p))
586 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
587 } else if (av_strstart(p, "source-filter:", &p)) {
589 get_word(buf1, sizeof(buf1), &p);
590 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
592 exclude = !strcmp(buf1, "excl");
594 get_word(buf1, sizeof(buf1), &p);
595 if (strcmp(buf1, "IN") != 0)
597 get_word(buf1, sizeof(buf1), &p);
598 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
600 // not checking that the destination address actually matches or is wildcard
601 get_word(buf1, sizeof(buf1), &p);
604 rtsp_src = av_mallocz(sizeof(*rtsp_src));
607 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
609 if (s->nb_streams == 0) {
610 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
612 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
613 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
616 if (s->nb_streams == 0) {
617 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
619 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
620 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
625 if (rt->server_type == RTSP_SERVER_WMS)
626 ff_wms_parse_sdp_a_line(s, p);
627 if (s->nb_streams > 0) {
628 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
630 if (rt->server_type == RTSP_SERVER_REAL)
631 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
633 if (rtsp_st->dynamic_handler &&
634 rtsp_st->dynamic_handler->parse_sdp_a_line)
635 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
636 rtsp_st->stream_index,
637 rtsp_st->dynamic_protocol_context, buf);
644 int ff_sdp_parse(AVFormatContext *s, const char *content)
646 RTSPState *rt = s->priv_data;
649 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
650 * contain long SDP lines containing complete ASF Headers (several
651 * kB) or arrays of MDPR (RM stream descriptor) headers plus
652 * "rulebooks" describing their properties. Therefore, the SDP line
655 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
656 * in rtpdec_xiph.c. */
658 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
662 p += strspn(p, SPACE_CHARS);
670 /* get the content */
672 while (*p != '\n' && *p != '\r' && *p != '\0') {
673 if ((q - buf) < sizeof(buf) - 1)
678 sdp_parse_line(s, s1, letter, buf);
680 while (*p != '\n' && *p != '\0')
686 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
687 av_freep(&s1->default_include_source_addrs[i]);
688 av_freep(&s1->default_include_source_addrs);
689 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
690 av_freep(&s1->default_exclude_source_addrs[i]);
691 av_freep(&s1->default_exclude_source_addrs);
693 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
694 if (!rt->p) return AVERROR(ENOMEM);
697 #endif /* CONFIG_RTPDEC */
699 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
701 RTSPState *rt = s->priv_data;
704 for (i = 0; i < rt->nb_rtsp_streams; i++) {
705 RTSPStream *rtsp_st = rt->rtsp_streams[i];
708 if (rtsp_st->transport_priv) {
710 AVFormatContext *rtpctx = rtsp_st->transport_priv;
711 av_write_trailer(rtpctx);
712 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
713 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
714 ff_rtsp_tcp_write_packet(s, rtsp_st);
715 ffio_free_dyn_buf(&rtpctx->pb);
717 avio_closep(&rtpctx->pb);
719 avformat_free_context(rtpctx);
720 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
721 ff_rdt_parse_close(rtsp_st->transport_priv);
722 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
723 ff_rtp_parse_close(rtsp_st->transport_priv);
725 rtsp_st->transport_priv = NULL;
726 if (rtsp_st->rtp_handle)
727 ffurl_close(rtsp_st->rtp_handle);
728 rtsp_st->rtp_handle = NULL;
732 /* close and free RTSP streams */
733 void ff_rtsp_close_streams(AVFormatContext *s)
735 RTSPState *rt = s->priv_data;
739 ff_rtsp_undo_setup(s, 0);
740 for (i = 0; i < rt->nb_rtsp_streams; i++) {
741 rtsp_st = rt->rtsp_streams[i];
743 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
744 if (rtsp_st->dynamic_handler->close)
745 rtsp_st->dynamic_handler->close(
746 rtsp_st->dynamic_protocol_context);
747 av_free(rtsp_st->dynamic_protocol_context);
749 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
750 av_freep(&rtsp_st->include_source_addrs[j]);
751 av_freep(&rtsp_st->include_source_addrs);
752 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
753 av_freep(&rtsp_st->exclude_source_addrs[j]);
754 av_freep(&rtsp_st->exclude_source_addrs);
759 av_freep(&rt->rtsp_streams);
761 avformat_close_input(&rt->asf_ctx);
763 if (CONFIG_RTPDEC && rt->ts)
764 avpriv_mpegts_parse_close(rt->ts);
766 av_freep(&rt->recvbuf);
769 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
771 RTSPState *rt = s->priv_data;
773 int reordering_queue_size = rt->reordering_queue_size;
774 if (reordering_queue_size < 0) {
775 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
776 reordering_queue_size = 0;
778 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
781 /* open the RTP context */
782 if (rtsp_st->stream_index >= 0)
783 st = s->streams[rtsp_st->stream_index];
785 s->ctx_flags |= AVFMTCTX_NOHEADER;
787 if (CONFIG_RTSP_MUXER && s->oformat) {
788 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
789 s, st, rtsp_st->rtp_handle,
790 RTSP_TCP_MAX_PACKET_SIZE,
791 rtsp_st->stream_index);
792 /* Ownership of rtp_handle is passed to the rtp mux context */
793 rtsp_st->rtp_handle = NULL;
796 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
797 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
798 return 0; // Don't need to open any parser here
799 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
800 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
801 rtsp_st->dynamic_protocol_context,
802 rtsp_st->dynamic_handler);
803 else if (CONFIG_RTPDEC)
804 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
805 rtsp_st->sdp_payload_type,
806 reordering_queue_size);
808 if (!rtsp_st->transport_priv) {
809 return AVERROR(ENOMEM);
810 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
811 if (rtsp_st->dynamic_handler) {
812 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
813 rtsp_st->dynamic_protocol_context,
814 rtsp_st->dynamic_handler);
816 if (rtsp_st->crypto_suite[0])
817 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
818 rtsp_st->crypto_suite,
819 rtsp_st->crypto_params);
825 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
826 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
833 q += strspn(q, SPACE_CHARS);
834 v = strtol(q, &p, 10);
838 v = strtol(p, &p, 10);
847 /* XXX: only one transport specification is parsed */
848 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
850 char transport_protocol[16];
852 char lower_transport[16];
854 RTSPTransportField *th;
857 reply->nb_transports = 0;
860 p += strspn(p, SPACE_CHARS);
864 th = &reply->transports[reply->nb_transports];
866 get_word_sep(transport_protocol, sizeof(transport_protocol),
868 if (!av_strcasecmp (transport_protocol, "rtp")) {
869 get_word_sep(profile, sizeof(profile), "/;,", &p);
870 lower_transport[0] = '\0';
871 /* rtp/avp/<protocol> */
873 get_word_sep(lower_transport, sizeof(lower_transport),
876 th->transport = RTSP_TRANSPORT_RTP;
877 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
878 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
879 /* x-pn-tng/<protocol> */
880 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
882 th->transport = RTSP_TRANSPORT_RDT;
883 } else if (!av_strcasecmp(transport_protocol, "raw")) {
884 get_word_sep(profile, sizeof(profile), "/;,", &p);
885 lower_transport[0] = '\0';
886 /* raw/raw/<protocol> */
888 get_word_sep(lower_transport, sizeof(lower_transport),
891 th->transport = RTSP_TRANSPORT_RAW;
893 if (!av_strcasecmp(lower_transport, "TCP"))
894 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
896 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
900 /* get each parameter */
901 while (*p != '\0' && *p != ',') {
902 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
903 if (!strcmp(parameter, "port")) {
906 rtsp_parse_range(&th->port_min, &th->port_max, &p);
908 } else if (!strcmp(parameter, "client_port")) {
911 rtsp_parse_range(&th->client_port_min,
912 &th->client_port_max, &p);
914 } else if (!strcmp(parameter, "server_port")) {
917 rtsp_parse_range(&th->server_port_min,
918 &th->server_port_max, &p);
920 } else if (!strcmp(parameter, "interleaved")) {
923 rtsp_parse_range(&th->interleaved_min,
924 &th->interleaved_max, &p);
926 } else if (!strcmp(parameter, "multicast")) {
927 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
928 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
929 } else if (!strcmp(parameter, "ttl")) {
933 th->ttl = strtol(p, &end, 10);
936 } else if (!strcmp(parameter, "destination")) {
939 get_word_sep(buf, sizeof(buf), ";,", &p);
940 get_sockaddr(buf, &th->destination);
942 } else if (!strcmp(parameter, "source")) {
945 get_word_sep(buf, sizeof(buf), ";,", &p);
946 av_strlcpy(th->source, buf, sizeof(th->source));
948 } else if (!strcmp(parameter, "mode")) {
951 get_word_sep(buf, sizeof(buf), ";, ", &p);
952 if (!strcmp(buf, "record") ||
953 !strcmp(buf, "receive"))
958 while (*p != ';' && *p != '\0' && *p != ',')
966 reply->nb_transports++;
970 static void handle_rtp_info(RTSPState *rt, const char *url,
971 uint32_t seq, uint32_t rtptime)
974 if (!rtptime || !url[0])
976 if (rt->transport != RTSP_TRANSPORT_RTP)
978 for (i = 0; i < rt->nb_rtsp_streams; i++) {
979 RTSPStream *rtsp_st = rt->rtsp_streams[i];
980 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
983 if (!strcmp(rtsp_st->control_url, url)) {
984 rtpctx->base_timestamp = rtptime;
990 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
993 char key[20], value[1024], url[1024] = "";
994 uint32_t seq = 0, rtptime = 0;
997 p += strspn(p, SPACE_CHARS);
1000 get_word_sep(key, sizeof(key), "=", &p);
1004 get_word_sep(value, sizeof(value), ";, ", &p);
1006 if (!strcmp(key, "url"))
1007 av_strlcpy(url, value, sizeof(url));
1008 else if (!strcmp(key, "seq"))
1009 seq = strtoul(value, NULL, 10);
1010 else if (!strcmp(key, "rtptime"))
1011 rtptime = strtoul(value, NULL, 10);
1013 handle_rtp_info(rt, url, seq, rtptime);
1022 handle_rtp_info(rt, url, seq, rtptime);
1025 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1026 RTSPState *rt, const char *method)
1030 /* NOTE: we do case independent match for broken servers */
1032 if (av_stristart(p, "Session:", &p)) {
1034 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1035 if (av_stristart(p, ";timeout=", &p) &&
1036 (t = strtol(p, NULL, 10)) > 0) {
1039 } else if (av_stristart(p, "Content-Length:", &p)) {
1040 reply->content_length = strtol(p, NULL, 10);
1041 } else if (av_stristart(p, "Transport:", &p)) {
1042 rtsp_parse_transport(reply, p);
1043 } else if (av_stristart(p, "CSeq:", &p)) {
1044 reply->seq = strtol(p, NULL, 10);
1045 } else if (av_stristart(p, "Range:", &p)) {
1046 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1047 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1048 p += strspn(p, SPACE_CHARS);
1049 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1050 } else if (av_stristart(p, "Server:", &p)) {
1051 p += strspn(p, SPACE_CHARS);
1052 av_strlcpy(reply->server, p, sizeof(reply->server));
1053 } else if (av_stristart(p, "Notice:", &p) ||
1054 av_stristart(p, "X-Notice:", &p)) {
1055 reply->notice = strtol(p, NULL, 10);
1056 } else if (av_stristart(p, "Location:", &p)) {
1057 p += strspn(p, SPACE_CHARS);
1058 av_strlcpy(reply->location, p , sizeof(reply->location));
1059 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1060 p += strspn(p, SPACE_CHARS);
1061 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1062 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1063 p += strspn(p, SPACE_CHARS);
1064 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1065 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1066 p += strspn(p, SPACE_CHARS);
1067 if (method && !strcmp(method, "DESCRIBE"))
1068 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1069 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1070 p += strspn(p, SPACE_CHARS);
1071 if (method && !strcmp(method, "PLAY"))
1072 rtsp_parse_rtp_info(rt, p);
1073 } else if (av_stristart(p, "Public:", &p) && rt) {
1074 if (strstr(p, "GET_PARAMETER") &&
1075 method && !strcmp(method, "OPTIONS"))
1076 rt->get_parameter_supported = 1;
1077 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1078 p += strspn(p, SPACE_CHARS);
1079 rt->accept_dynamic_rate = atoi(p);
1080 } else if (av_stristart(p, "Content-Type:", &p)) {
1081 p += strspn(p, SPACE_CHARS);
1082 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1086 /* skip a RTP/TCP interleaved packet */
1087 void ff_rtsp_skip_packet(AVFormatContext *s)
1089 RTSPState *rt = s->priv_data;
1093 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1096 len = AV_RB16(buf + 1);
1098 av_dlog(s, "skipping RTP packet len=%d\n", len);
1103 if (len1 > sizeof(buf))
1105 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1112 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1113 unsigned char **content_ptr,
1114 int return_on_interleaved_data, const char *method)
1116 RTSPState *rt = s->priv_data;
1117 char buf[4096], buf1[1024], *q;
1120 int ret, content_length, line_count = 0, request = 0;
1121 unsigned char *content = NULL;
1127 memset(reply, 0, sizeof(*reply));
1129 /* parse reply (XXX: use buffers) */
1130 rt->last_reply[0] = '\0';
1134 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1135 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1141 /* XXX: only parse it if first char on line ? */
1142 if (return_on_interleaved_data) {
1145 ff_rtsp_skip_packet(s);
1146 } else if (ch != '\r') {
1147 if ((q - buf) < sizeof(buf) - 1)
1153 av_dlog(s, "line='%s'\n", buf);
1155 /* test if last line */
1159 if (line_count == 0) {
1160 /* get reply code */
1161 get_word(buf1, sizeof(buf1), &p);
1162 if (!strncmp(buf1, "RTSP/", 5)) {
1163 get_word(buf1, sizeof(buf1), &p);
1164 reply->status_code = atoi(buf1);
1165 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1167 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1168 get_word(buf1, sizeof(buf1), &p); // object
1172 ff_rtsp_parse_line(reply, p, rt, method);
1173 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1174 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1179 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1180 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1182 content_length = reply->content_length;
1183 if (content_length > 0) {
1184 /* leave some room for a trailing '\0' (useful for simple parsing) */
1185 content = av_malloc(content_length + 1);
1187 return AVERROR(ENOMEM);
1188 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1189 content[content_length] = '\0';
1192 *content_ptr = content;
1198 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1199 const char* ptr = buf;
1201 if (!strcmp(reply->reason, "OPTIONS")) {
1202 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1204 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1205 if (reply->session_id[0])
1206 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1209 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1211 av_strlcat(buf, "\r\n", sizeof(buf));
1213 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1214 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1217 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1219 rt->last_cmd_time = av_gettime_relative();
1220 /* Even if the request from the server had data, it is not the data
1221 * that the caller wants or expects. The memory could also be leaked
1222 * if the actual following reply has content data. */
1224 av_freep(content_ptr);
1225 /* If method is set, this is called from ff_rtsp_send_cmd,
1226 * where a reply to exactly this request is awaited. For
1227 * callers from within packet receiving, we just want to
1228 * return to the caller and go back to receiving packets. */
1234 if (rt->seq != reply->seq) {
1235 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1236 rt->seq, reply->seq);
1240 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1241 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1242 reply->notice == 2306 /* Continuous Feed Terminated */) {
1243 rt->state = RTSP_STATE_IDLE;
1244 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1245 return AVERROR(EIO); /* data or server error */
1246 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1247 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1248 return AVERROR(EPERM);
1254 * Send a command to the RTSP server without waiting for the reply.
1256 * @param s RTSP (de)muxer context
1257 * @param method the method for the request
1258 * @param url the target url for the request
1259 * @param headers extra header lines to include in the request
1260 * @param send_content if non-null, the data to send as request body content
1261 * @param send_content_length the length of the send_content data, or 0 if
1262 * send_content is null
1264 * @return zero if success, nonzero otherwise
1266 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1267 const char *method, const char *url,
1268 const char *headers,
1269 const unsigned char *send_content,
1270 int send_content_length)
1272 RTSPState *rt = s->priv_data;
1273 char buf[4096], *out_buf;
1274 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1276 /* Add in RTSP headers */
1279 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1281 av_strlcat(buf, headers, sizeof(buf));
1282 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1283 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1284 if (rt->session_id[0] != '\0' && (!headers ||
1285 !strstr(headers, "\nIf-Match:"))) {
1286 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1289 char *str = ff_http_auth_create_response(&rt->auth_state,
1290 rt->auth, url, method);
1292 av_strlcat(buf, str, sizeof(buf));
1295 if (send_content_length > 0 && send_content)
1296 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1297 av_strlcat(buf, "\r\n", sizeof(buf));
1299 /* base64 encode rtsp if tunneling */
1300 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1301 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1302 out_buf = base64buf;
1305 av_dlog(s, "Sending:\n%s--\n", buf);
1307 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1308 if (send_content_length > 0 && send_content) {
1309 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1310 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1311 "with content data not supported\n");
1312 return AVERROR_PATCHWELCOME;
1314 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1316 rt->last_cmd_time = av_gettime_relative();
1321 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1322 const char *url, const char *headers)
1324 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1327 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1328 const char *headers, RTSPMessageHeader *reply,
1329 unsigned char **content_ptr)
1331 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1332 content_ptr, NULL, 0);
1335 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1336 const char *method, const char *url,
1338 RTSPMessageHeader *reply,
1339 unsigned char **content_ptr,
1340 const unsigned char *send_content,
1341 int send_content_length)
1343 RTSPState *rt = s->priv_data;
1344 HTTPAuthType cur_auth_type;
1345 int ret, attempts = 0;
1348 cur_auth_type = rt->auth_state.auth_type;
1349 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1351 send_content_length)))
1354 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1358 if (reply->status_code == 401 &&
1359 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1360 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1363 if (reply->status_code > 400){
1364 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1368 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1374 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1375 int lower_transport, const char *real_challenge)
1377 RTSPState *rt = s->priv_data;
1378 int rtx = 0, j, i, err, interleave = 0, port_off;
1379 RTSPStream *rtsp_st;
1380 RTSPMessageHeader reply1, *reply = &reply1;
1382 const char *trans_pref;
1384 if (rt->transport == RTSP_TRANSPORT_RDT)
1385 trans_pref = "x-pn-tng";
1386 else if (rt->transport == RTSP_TRANSPORT_RAW)
1387 trans_pref = "RAW/RAW";
1389 trans_pref = "RTP/AVP";
1391 /* default timeout: 1 minute */
1394 /* Choose a random starting offset within the first half of the
1395 * port range, to allow for a number of ports to try even if the offset
1396 * happens to be at the end of the random range. */
1397 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1398 /* even random offset */
1399 port_off -= port_off & 0x01;
1401 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1402 char transport[2048];
1405 * WMS serves all UDP data over a single connection, the RTX, which
1406 * isn't necessarily the first in the SDP but has to be the first
1407 * to be set up, else the second/third SETUP will fail with a 461.
1409 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1410 rt->server_type == RTSP_SERVER_WMS) {
1413 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1414 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1416 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1420 if (rtx == rt->nb_rtsp_streams)
1421 return -1; /* no RTX found */
1422 rtsp_st = rt->rtsp_streams[rtx];
1424 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1426 rtsp_st = rt->rtsp_streams[i];
1429 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1432 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1433 port = reply->transports[0].client_port_min;
1437 /* first try in specified port range */
1438 while (j <= rt->rtp_port_max) {
1439 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1440 "?localport=%d", j);
1441 /* we will use two ports per rtp stream (rtp and rtcp) */
1443 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1444 &s->interrupt_callback, NULL))
1447 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1452 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1454 snprintf(transport, sizeof(transport) - 1,
1455 "%s/UDP;", trans_pref);
1456 if (rt->server_type != RTSP_SERVER_REAL)
1457 av_strlcat(transport, "unicast;", sizeof(transport));
1458 av_strlcatf(transport, sizeof(transport),
1459 "client_port=%d", port);
1460 if (rt->transport == RTSP_TRANSPORT_RTP &&
1461 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1462 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1466 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1467 /* For WMS streams, the application streams are only used for
1468 * UDP. When trying to set it up for TCP streams, the server
1469 * will return an error. Therefore, we skip those streams. */
1470 if (rt->server_type == RTSP_SERVER_WMS &&
1471 (rtsp_st->stream_index < 0 ||
1472 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1475 snprintf(transport, sizeof(transport) - 1,
1476 "%s/TCP;", trans_pref);
1477 if (rt->transport != RTSP_TRANSPORT_RDT)
1478 av_strlcat(transport, "unicast;", sizeof(transport));
1479 av_strlcatf(transport, sizeof(transport),
1480 "interleaved=%d-%d",
1481 interleave, interleave + 1);
1485 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1486 snprintf(transport, sizeof(transport) - 1,
1487 "%s/UDP;multicast", trans_pref);
1490 av_strlcat(transport, ";mode=record", sizeof(transport));
1491 } else if (rt->server_type == RTSP_SERVER_REAL ||
1492 rt->server_type == RTSP_SERVER_WMS)
1493 av_strlcat(transport, ";mode=play", sizeof(transport));
1494 snprintf(cmd, sizeof(cmd),
1495 "Transport: %s\r\n",
1497 if (rt->accept_dynamic_rate)
1498 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1499 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1500 char real_res[41], real_csum[9];
1501 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1503 av_strlcatf(cmd, sizeof(cmd),
1505 "RealChallenge2: %s, sd=%s\r\n",
1506 rt->session_id, real_res, real_csum);
1508 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1509 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1512 } else if (reply->status_code != RTSP_STATUS_OK ||
1513 reply->nb_transports != 1) {
1514 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1518 /* XXX: same protocol for all streams is required */
1520 if (reply->transports[0].lower_transport != rt->lower_transport ||
1521 reply->transports[0].transport != rt->transport) {
1522 err = AVERROR_INVALIDDATA;
1526 rt->lower_transport = reply->transports[0].lower_transport;
1527 rt->transport = reply->transports[0].transport;
1530 /* Fail if the server responded with another lower transport mode
1531 * than what we requested. */
1532 if (reply->transports[0].lower_transport != lower_transport) {
1533 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1534 err = AVERROR_INVALIDDATA;
1538 switch(reply->transports[0].lower_transport) {
1539 case RTSP_LOWER_TRANSPORT_TCP:
1540 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1541 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1544 case RTSP_LOWER_TRANSPORT_UDP: {
1545 char url[1024], options[30] = "";
1546 const char *peer = host;
1548 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1549 av_strlcpy(options, "?connect=1", sizeof(options));
1550 /* Use source address if specified */
1551 if (reply->transports[0].source[0])
1552 peer = reply->transports[0].source;
1553 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1554 reply->transports[0].server_port_min, "%s", options);
1555 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1556 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1557 err = AVERROR_INVALIDDATA;
1562 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1563 char url[1024], namebuf[50], optbuf[20] = "";
1564 struct sockaddr_storage addr;
1567 if (reply->transports[0].destination.ss_family) {
1568 addr = reply->transports[0].destination;
1569 port = reply->transports[0].port_min;
1570 ttl = reply->transports[0].ttl;
1572 addr = rtsp_st->sdp_ip;
1573 port = rtsp_st->sdp_port;
1574 ttl = rtsp_st->sdp_ttl;
1577 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1578 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1579 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1580 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1581 port, "%s", optbuf);
1582 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1583 &s->interrupt_callback, NULL) < 0) {
1584 err = AVERROR_INVALIDDATA;
1591 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1595 if (rt->nb_rtsp_streams && reply->timeout > 0)
1596 rt->timeout = reply->timeout;
1598 if (rt->server_type == RTSP_SERVER_REAL)
1599 rt->need_subscription = 1;
1604 ff_rtsp_undo_setup(s, 0);
1608 void ff_rtsp_close_connections(AVFormatContext *s)
1610 RTSPState *rt = s->priv_data;
1611 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1612 ffurl_close(rt->rtsp_hd);
1613 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1616 int ff_rtsp_connect(AVFormatContext *s)
1618 RTSPState *rt = s->priv_data;
1619 char proto[128], host[1024], path[1024];
1620 char tcpname[1024], cmd[2048], auth[128];
1621 const char *lower_rtsp_proto = "tcp";
1622 int port, err, tcp_fd;
1623 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1624 int lower_transport_mask = 0;
1625 int default_port = RTSP_DEFAULT_PORT;
1626 char real_challenge[64] = "";
1627 struct sockaddr_storage peer;
1628 socklen_t peer_len = sizeof(peer);
1630 if (rt->rtp_port_max < rt->rtp_port_min) {
1631 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1632 "than min port %d\n", rt->rtp_port_max,
1634 return AVERROR(EINVAL);
1637 if (!ff_network_init())
1638 return AVERROR(EIO);
1640 if (s->max_delay < 0) /* Not set by the caller */
1641 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1643 rt->control_transport = RTSP_MODE_PLAIN;
1644 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1645 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1646 rt->control_transport = RTSP_MODE_TUNNEL;
1648 /* Only pass through valid flags from here */
1649 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1652 /* extract hostname and port */
1653 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1654 host, sizeof(host), &port, path, sizeof(path), s->filename);
1656 if (!strcmp(proto, "rtsps")) {
1657 lower_rtsp_proto = "tls";
1658 default_port = RTSPS_DEFAULT_PORT;
1659 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1663 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1666 port = default_port;
1668 lower_transport_mask = rt->lower_transport_mask;
1670 if (!lower_transport_mask)
1671 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1674 /* Only UDP or TCP - UDP multicast isn't supported. */
1675 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1676 (1 << RTSP_LOWER_TRANSPORT_TCP);
1677 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1678 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1679 "only UDP and TCP are supported for output.\n");
1680 err = AVERROR(EINVAL);
1685 /* Construct the URI used in request; this is similar to s->filename,
1686 * but with authentication credentials removed and RTSP specific options
1688 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1689 host, port, "%s", path);
1691 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1692 /* set up initial handshake for tunneling */
1693 char httpname[1024];
1694 char sessioncookie[17];
1697 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1698 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1699 av_get_random_seed(), av_get_random_seed());
1702 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1703 &s->interrupt_callback) < 0) {
1708 /* generate GET headers */
1709 snprintf(headers, sizeof(headers),
1710 "x-sessioncookie: %s\r\n"
1711 "Accept: application/x-rtsp-tunnelled\r\n"
1712 "Pragma: no-cache\r\n"
1713 "Cache-Control: no-cache\r\n",
1715 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1717 /* complete the connection */
1718 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1724 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1725 &s->interrupt_callback) < 0 ) {
1730 /* generate POST headers */
1731 snprintf(headers, sizeof(headers),
1732 "x-sessioncookie: %s\r\n"
1733 "Content-Type: application/x-rtsp-tunnelled\r\n"
1734 "Pragma: no-cache\r\n"
1735 "Cache-Control: no-cache\r\n"
1736 "Content-Length: 32767\r\n"
1737 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1739 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1740 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1742 /* Initialize the authentication state for the POST session. The HTTP
1743 * protocol implementation doesn't properly handle multi-pass
1744 * authentication for POST requests, since it would require one of
1746 * - implementing Expect: 100-continue, which many HTTP servers
1747 * don't support anyway, even less the RTSP servers that do HTTP
1749 * - sending the whole POST data until getting a 401 reply specifying
1750 * what authentication method to use, then resending all that data
1751 * - waiting for potential 401 replies directly after sending the
1752 * POST header (waiting for some unspecified time)
1753 * Therefore, we copy the full auth state, which works for both basic
1754 * and digest. (For digest, we would have to synchronize the nonce
1755 * count variable between the two sessions, if we'd do more requests
1756 * with the original session, though.)
1758 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1760 /* complete the connection */
1761 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1767 /* open the tcp connection */
1768 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1770 "?timeout=%d", rt->stimeout);
1771 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1772 &s->interrupt_callback, NULL)) < 0) {
1776 rt->rtsp_hd_out = rt->rtsp_hd;
1780 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1785 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1786 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1787 NULL, 0, NI_NUMERICHOST);
1790 /* request options supported by the server; this also detects server
1792 for (rt->server_type = RTSP_SERVER_RTP;;) {
1794 if (rt->server_type == RTSP_SERVER_REAL)
1797 * The following entries are required for proper
1798 * streaming from a Realmedia server. They are
1799 * interdependent in some way although we currently
1800 * don't quite understand how. Values were copied
1801 * from mplayer SVN r23589.
1802 * ClientChallenge is a 16-byte ID in hex
1803 * CompanyID is a 16-byte ID in base64
1805 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1806 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1807 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1808 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1810 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1811 if (reply->status_code != RTSP_STATUS_OK) {
1812 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1816 /* detect server type if not standard-compliant RTP */
1817 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1818 rt->server_type = RTSP_SERVER_REAL;
1820 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1821 rt->server_type = RTSP_SERVER_WMS;
1822 } else if (rt->server_type == RTSP_SERVER_REAL)
1823 strcpy(real_challenge, reply->real_challenge);
1827 if (CONFIG_RTSP_DEMUXER && s->iformat)
1828 err = ff_rtsp_setup_input_streams(s, reply);
1829 else if (CONFIG_RTSP_MUXER)
1830 err = ff_rtsp_setup_output_streams(s, host);
1837 int lower_transport = ff_log2_tab[lower_transport_mask &
1838 ~(lower_transport_mask - 1)];
1840 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1841 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1842 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1844 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1845 rt->server_type == RTSP_SERVER_REAL ?
1846 real_challenge : NULL);
1849 lower_transport_mask &= ~(1 << lower_transport);
1850 if (lower_transport_mask == 0 && err == 1) {
1851 err = AVERROR(EPROTONOSUPPORT);
1856 rt->lower_transport_mask = lower_transport_mask;
1857 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1858 rt->state = RTSP_STATE_IDLE;
1859 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1862 ff_rtsp_close_streams(s);
1863 ff_rtsp_close_connections(s);
1864 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1865 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1866 rt->session_id[0] = '\0';
1867 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1875 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1878 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1879 uint8_t *buf, int buf_size, int64_t wait_end)
1881 RTSPState *rt = s->priv_data;
1882 RTSPStream *rtsp_st;
1883 int n, i, ret, tcp_fd, timeout_cnt = 0;
1885 struct pollfd *p = rt->p;
1886 int *fds = NULL, fdsnum, fdsidx;
1889 if (ff_check_interrupt(&s->interrupt_callback))
1890 return AVERROR_EXIT;
1891 if (wait_end && wait_end - av_gettime_relative() < 0)
1892 return AVERROR(EAGAIN);
1895 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1896 p[max_p].fd = tcp_fd;
1897 p[max_p++].events = POLLIN;
1901 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1902 rtsp_st = rt->rtsp_streams[i];
1903 if (rtsp_st->rtp_handle) {
1904 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1906 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1910 av_log(s, AV_LOG_ERROR,
1911 "Number of fds %d not supported\n", fdsnum);
1912 return AVERROR_INVALIDDATA;
1914 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1915 p[max_p].fd = fds[fdsidx];
1916 p[max_p++].events = POLLIN;
1921 n = poll(p, max_p, POLL_TIMEOUT_MS);
1923 int j = 1 - (tcp_fd == -1);
1925 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1926 rtsp_st = rt->rtsp_streams[i];
1927 if (rtsp_st->rtp_handle) {
1928 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1929 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1931 *prtsp_st = rtsp_st;
1938 #if CONFIG_RTSP_DEMUXER
1939 if (tcp_fd != -1 && p[0].revents & POLLIN) {
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)
1960 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1961 return AVERROR(ETIMEDOUT);
1962 } else if (n < 0 && errno != EINTR)
1963 return AVERROR(errno);
1967 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1968 const uint8_t *buf, int len)
1970 RTSPState *rt = s->priv_data;
1974 if (rt->nb_rtsp_streams == 1) {
1975 *rtsp_st = rt->rtsp_streams[0];
1978 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1979 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1981 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1982 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1985 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1986 *rtsp_st = rt->rtsp_streams[i];
1993 av_log(s, AV_LOG_WARNING,
1994 "Unable to pick stream for packet - SSRC not known for "
1996 return AVERROR(EAGAIN);
1999 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2000 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2001 *rtsp_st = rt->rtsp_streams[i];
2007 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2008 return AVERROR(EAGAIN);
2011 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2013 RTSPState *rt = s->priv_data;
2015 RTSPStream *rtsp_st, *first_queue_st = NULL;
2016 int64_t wait_end = 0;
2018 if (rt->nb_byes == rt->nb_rtsp_streams)
2021 /* get next frames from the same RTP packet */
2022 if (rt->cur_transport_priv) {
2023 if (rt->transport == RTSP_TRANSPORT_RDT) {
2024 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2025 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2026 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2027 } else if (CONFIG_RTPDEC && rt->ts) {
2028 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2030 rt->recvbuf_pos += ret;
2031 ret = rt->recvbuf_pos < rt->recvbuf_len;
2036 rt->cur_transport_priv = NULL;
2038 } else if (ret == 1) {
2041 rt->cur_transport_priv = NULL;
2045 if (rt->transport == RTSP_TRANSPORT_RTP) {
2047 int64_t first_queue_time = 0;
2048 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2049 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2053 queue_time = ff_rtp_queued_packet_time(rtpctx);
2054 if (queue_time && (queue_time - first_queue_time < 0 ||
2055 !first_queue_time)) {
2056 first_queue_time = queue_time;
2057 first_queue_st = rt->rtsp_streams[i];
2060 if (first_queue_time) {
2061 wait_end = first_queue_time + s->max_delay;
2064 first_queue_st = NULL;
2068 /* read next RTP packet */
2070 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2072 return AVERROR(ENOMEM);
2075 switch(rt->lower_transport) {
2077 #if CONFIG_RTSP_DEMUXER
2078 case RTSP_LOWER_TRANSPORT_TCP:
2079 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2082 case RTSP_LOWER_TRANSPORT_UDP:
2083 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2084 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2085 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2086 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2088 case RTSP_LOWER_TRANSPORT_CUSTOM:
2089 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2090 wait_end && wait_end < av_gettime_relative())
2091 len = AVERROR(EAGAIN);
2093 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2094 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2095 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2096 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2099 if (len == AVERROR(EAGAIN) && first_queue_st &&
2100 rt->transport == RTSP_TRANSPORT_RTP) {
2101 rtsp_st = first_queue_st;
2102 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2109 if (rt->transport == RTSP_TRANSPORT_RDT) {
2110 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2111 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2112 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2113 if (rtsp_st->feedback) {
2114 AVIOContext *pb = NULL;
2115 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2117 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2120 /* Either bad packet, or a RTCP packet. Check if the
2121 * first_rtcp_ntp_time field was initialized. */
2122 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2123 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2124 /* first_rtcp_ntp_time has been initialized for this stream,
2125 * copy the same value to all other uninitialized streams,
2126 * in order to map their timestamp origin to the same ntp time
2129 AVStream *st = NULL;
2130 if (rtsp_st->stream_index >= 0)
2131 st = s->streams[rtsp_st->stream_index];
2132 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2133 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2134 AVStream *st2 = NULL;
2135 if (rt->rtsp_streams[i]->stream_index >= 0)
2136 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2137 if (rtpctx2 && st && st2 &&
2138 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2139 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2140 rtpctx2->rtcp_ts_offset = av_rescale_q(
2141 rtpctx->rtcp_ts_offset, st->time_base,
2145 // Make real NTP start time available in AVFormatContext
2146 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2147 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2149 s->start_time_realtime -=
2150 av_rescale (rtpctx->rtcp_ts_offset,
2151 (uint64_t) rtpctx->st->time_base.num * 1000000,
2152 rtpctx->st->time_base.den);
2156 if (ret == -RTCP_BYE) {
2159 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2160 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2162 if (rt->nb_byes == rt->nb_rtsp_streams)
2166 } else if (CONFIG_RTPDEC && rt->ts) {
2167 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2170 rt->recvbuf_len = len;
2171 rt->recvbuf_pos = ret;
2172 rt->cur_transport_priv = rt->ts;
2179 return AVERROR_INVALIDDATA;
2185 /* more packets may follow, so we save the RTP context */
2186 rt->cur_transport_priv = rtsp_st->transport_priv;
2190 #endif /* CONFIG_RTPDEC */
2192 #if CONFIG_SDP_DEMUXER
2193 static int sdp_probe(AVProbeData *p1)
2195 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2197 /* we look for a line beginning "c=IN IP" */
2198 while (p < p_end && *p != '\0') {
2199 if (sizeof("c=IN IP") - 1 < p_end - p &&
2200 av_strstart(p, "c=IN IP", NULL))
2201 return AVPROBE_SCORE_EXTENSION;
2203 while (p < p_end - 1 && *p != '\n') p++;
2212 static void append_source_addrs(char *buf, int size, const char *name,
2213 int count, struct RTSPSource **addrs)
2218 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2219 for (i = 1; i < count; i++)
2220 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2223 static int sdp_read_header(AVFormatContext *s)
2225 RTSPState *rt = s->priv_data;
2226 RTSPStream *rtsp_st;
2231 if (!ff_network_init())
2232 return AVERROR(EIO);
2234 if (s->max_delay < 0) /* Not set by the caller */
2235 s->max_delay = DEFAULT_REORDERING_DELAY;
2236 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2237 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2239 /* read the whole sdp file */
2240 /* XXX: better loading */
2241 content = av_malloc(SDP_MAX_SIZE);
2242 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2245 return AVERROR_INVALIDDATA;
2247 content[size] ='\0';
2249 err = ff_sdp_parse(s, content);
2253 /* open each RTP stream */
2254 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2256 rtsp_st = rt->rtsp_streams[i];
2258 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2259 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2260 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2261 ff_url_join(url, sizeof(url), "rtp", NULL,
2262 namebuf, rtsp_st->sdp_port,
2263 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2264 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2265 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2266 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2268 append_source_addrs(url, sizeof(url), "sources",
2269 rtsp_st->nb_include_source_addrs,
2270 rtsp_st->include_source_addrs);
2271 append_source_addrs(url, sizeof(url), "block",
2272 rtsp_st->nb_exclude_source_addrs,
2273 rtsp_st->exclude_source_addrs);
2274 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2275 &s->interrupt_callback, NULL) < 0) {
2276 err = AVERROR_INVALIDDATA;
2280 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2285 ff_rtsp_close_streams(s);
2290 static int sdp_read_close(AVFormatContext *s)
2292 ff_rtsp_close_streams(s);
2297 static const AVClass sdp_demuxer_class = {
2298 .class_name = "SDP demuxer",
2299 .item_name = av_default_item_name,
2300 .option = sdp_options,
2301 .version = LIBAVUTIL_VERSION_INT,
2304 AVInputFormat ff_sdp_demuxer = {
2306 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2307 .priv_data_size = sizeof(RTSPState),
2308 .read_probe = sdp_probe,
2309 .read_header = sdp_read_header,
2310 .read_packet = ff_rtsp_fetch_packet,
2311 .read_close = sdp_read_close,
2312 .priv_class = &sdp_demuxer_class,
2314 #endif /* CONFIG_SDP_DEMUXER */
2316 #if CONFIG_RTP_DEMUXER
2317 static int rtp_probe(AVProbeData *p)
2319 if (av_strstart(p->filename, "rtp:", NULL))
2320 return AVPROBE_SCORE_MAX;
2324 static int rtp_read_header(AVFormatContext *s)
2326 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2327 char host[500], sdp[500];
2329 URLContext* in = NULL;
2331 AVCodecContext codec = { 0 };
2332 struct sockaddr_storage addr;
2334 socklen_t addrlen = sizeof(addr);
2335 RTSPState *rt = s->priv_data;
2337 if (!ff_network_init())
2338 return AVERROR(EIO);
2340 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2341 &s->interrupt_callback, NULL);
2346 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2347 if (ret == AVERROR(EAGAIN))
2352 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2356 if ((recvbuf[0] & 0xc0) != 0x80) {
2357 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2362 if (RTP_PT_IS_RTCP(recvbuf[1]))
2365 payload_type = recvbuf[1] & 0x7f;
2368 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2372 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2373 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2374 "without an SDP file describing it\n",
2378 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2379 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2380 "properly you need an SDP file "
2384 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2385 NULL, 0, s->filename);
2387 snprintf(sdp, sizeof(sdp),
2388 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2389 addr.ss_family == AF_INET ? 4 : 6, host,
2390 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2391 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2392 port, payload_type);
2393 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2395 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2398 /* sdp_read_header initializes this again */
2401 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2403 ret = sdp_read_header(s);
2414 static const AVClass rtp_demuxer_class = {
2415 .class_name = "RTP demuxer",
2416 .item_name = av_default_item_name,
2417 .option = rtp_options,
2418 .version = LIBAVUTIL_VERSION_INT,
2421 AVInputFormat ff_rtp_demuxer = {
2423 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2424 .priv_data_size = sizeof(RTSPState),
2425 .read_probe = rtp_probe,
2426 .read_header = rtp_read_header,
2427 .read_packet = ff_rtsp_fetch_packet,
2428 .read_close = sdp_read_close,
2429 .flags = AVFMT_NOFILE,
2430 .priv_class = &rtp_demuxer_class,
2432 #endif /* CONFIG_RTP_DEMUXER */