3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
51 /* Timeout values for socket poll, in ms,
52 * and read_packet(), in seconds */
53 #define POLL_TIMEOUT_MS 100
54 #define READ_PACKET_TIMEOUT_S 10
55 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
56 #define SDP_MAX_SIZE 16384
57 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
58 #define DEFAULT_REORDERING_DELAY 100000
60 #define OFFSET(x) offsetof(RTSPState, x)
61 #define DEC AV_OPT_FLAG_DECODING_PARAM
62 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 #define RTSP_FLAG_OPTS(name, longname) \
65 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
66 { "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" }, \
67 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 #define RTSP_REORDERING_OPTS() \
76 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
78 const AVOption ff_rtsp_options[] = {
79 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
80 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
81 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
82 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
85 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
86 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
87 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
88 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
89 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
90 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
91 RTSP_REORDERING_OPTS(),
95 static const AVOption sdp_options[] = {
96 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
97 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
98 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
99 RTSP_REORDERING_OPTS(),
103 static const AVOption rtp_options[] = {
104 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
105 RTSP_REORDERING_OPTS(),
109 static void get_word_until_chars(char *buf, int buf_size,
110 const char *sep, const char **pp)
116 p += strspn(p, SPACE_CHARS);
118 while (!strchr(sep, *p) && *p != '\0') {
119 if ((q - buf) < buf_size - 1)
128 static void get_word_sep(char *buf, int buf_size, const char *sep,
131 if (**pp == '/') (*pp)++;
132 get_word_until_chars(buf, buf_size, sep, pp);
135 static void get_word(char *buf, int buf_size, const char **pp)
137 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
140 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
142 * Used for seeking in the rtp stream.
144 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
148 p += strspn(p, SPACE_CHARS);
149 if (!av_stristart(p, "npt=", &p))
152 *start = AV_NOPTS_VALUE;
153 *end = AV_NOPTS_VALUE;
155 get_word_sep(buf, sizeof(buf), "-", &p);
156 av_parse_time(start, buf, 1);
159 get_word_sep(buf, sizeof(buf), "-", &p);
160 av_parse_time(end, buf, 1);
164 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
166 struct addrinfo hints = { 0 }, *ai = NULL;
167 hints.ai_flags = AI_NUMERICHOST;
168 if (getaddrinfo(buf, NULL, &hints, &ai))
170 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
176 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
177 RTSPStream *rtsp_st, AVCodecContext *codec)
182 codec->codec_id = handler->codec_id;
183 rtsp_st->dynamic_handler = handler;
184 if (handler->alloc) {
185 rtsp_st->dynamic_protocol_context = handler->alloc();
186 if (!rtsp_st->dynamic_protocol_context)
187 rtsp_st->dynamic_handler = NULL;
191 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
192 static int sdp_parse_rtpmap(AVFormatContext *s,
193 AVStream *st, RTSPStream *rtsp_st,
194 int payload_type, const char *p)
196 AVCodecContext *codec = st->codec;
202 /* See if we can handle this kind of payload.
203 * The space should normally not be there but some Real streams or
204 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
205 * have a trailing space. */
206 get_word_sep(buf, sizeof(buf), "/ ", &p);
207 if (payload_type < RTP_PT_PRIVATE) {
208 /* We are in a standard case
209 * (from http://www.iana.org/assignments/rtp-parameters). */
210 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
213 if (codec->codec_id == AV_CODEC_ID_NONE) {
214 RTPDynamicProtocolHandler *handler =
215 ff_rtp_handler_find_by_name(buf, codec->codec_type);
216 init_rtp_handler(handler, rtsp_st, codec);
217 /* If no dynamic handler was found, check with the list of standard
218 * allocated types, if such a stream for some reason happens to
219 * use a private payload type. This isn't handled in rtpdec.c, since
220 * the format name from the rtpmap line never is passed into rtpdec. */
221 if (!rtsp_st->dynamic_handler)
222 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
225 c = avcodec_find_decoder(codec->codec_id);
231 get_word_sep(buf, sizeof(buf), "/", &p);
233 switch (codec->codec_type) {
234 case AVMEDIA_TYPE_AUDIO:
235 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
236 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
237 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
239 codec->sample_rate = i;
240 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
241 get_word_sep(buf, sizeof(buf), "/", &p);
246 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
248 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
251 case AVMEDIA_TYPE_VIDEO:
252 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
254 avpriv_set_pts_info(st, 32, 1, i);
259 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
260 rtsp_st->dynamic_handler->init(s, st->index,
261 rtsp_st->dynamic_protocol_context);
265 /* parse the attribute line from the fmtp a line of an sdp response. This
266 * is broken out as a function because it is used in rtp_h264.c, which is
268 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
269 char *value, int value_size)
271 *p += strspn(*p, SPACE_CHARS);
273 get_word_sep(attr, attr_size, "=", p);
276 get_word_sep(value, value_size, ";", p);
284 typedef struct SDPParseState {
286 struct sockaddr_storage default_ip;
288 int skip_media; ///< set if an unknown m= line occurs
289 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
290 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
291 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
292 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
295 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
296 struct RTSPSource ***dest, int *dest_count)
298 RTSPSource *rtsp_src, *rtsp_src2;
300 for (i = 0; i < count; i++) {
302 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
305 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
306 dynarray_add(dest, dest_count, rtsp_src2);
310 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
311 int letter, const char *buf)
313 RTSPState *rt = s->priv_data;
314 char buf1[64], st_type[64];
316 enum AVMediaType codec_type;
320 RTSPSource *rtsp_src;
321 struct sockaddr_storage sdp_ip;
324 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
327 if (s1->skip_media && letter != 'm')
331 get_word(buf1, sizeof(buf1), &p);
332 if (strcmp(buf1, "IN") != 0)
334 get_word(buf1, sizeof(buf1), &p);
335 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
337 get_word_sep(buf1, sizeof(buf1), "/", &p);
338 if (get_sockaddr(buf1, &sdp_ip))
343 get_word_sep(buf1, sizeof(buf1), "/", &p);
346 if (s->nb_streams == 0) {
347 s1->default_ip = sdp_ip;
348 s1->default_ttl = ttl;
350 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
351 rtsp_st->sdp_ip = sdp_ip;
352 rtsp_st->sdp_ttl = ttl;
356 av_dict_set(&s->metadata, "title", p, 0);
359 if (s->nb_streams == 0) {
360 av_dict_set(&s->metadata, "comment", p, 0);
367 codec_type = AVMEDIA_TYPE_UNKNOWN;
368 get_word(st_type, sizeof(st_type), &p);
369 if (!strcmp(st_type, "audio")) {
370 codec_type = AVMEDIA_TYPE_AUDIO;
371 } else if (!strcmp(st_type, "video")) {
372 codec_type = AVMEDIA_TYPE_VIDEO;
373 } else if (!strcmp(st_type, "application")) {
374 codec_type = AVMEDIA_TYPE_DATA;
376 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
380 rtsp_st = av_mallocz(sizeof(RTSPStream));
383 rtsp_st->stream_index = -1;
384 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
386 rtsp_st->sdp_ip = s1->default_ip;
387 rtsp_st->sdp_ttl = s1->default_ttl;
389 copy_default_source_addrs(s1->default_include_source_addrs,
390 s1->nb_default_include_source_addrs,
391 &rtsp_st->include_source_addrs,
392 &rtsp_st->nb_include_source_addrs);
393 copy_default_source_addrs(s1->default_exclude_source_addrs,
394 s1->nb_default_exclude_source_addrs,
395 &rtsp_st->exclude_source_addrs,
396 &rtsp_st->nb_exclude_source_addrs);
398 get_word(buf1, sizeof(buf1), &p); /* port */
399 rtsp_st->sdp_port = atoi(buf1);
401 get_word(buf1, sizeof(buf1), &p); /* protocol */
402 if (!strcmp(buf1, "udp"))
403 rt->transport = RTSP_TRANSPORT_RAW;
404 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
405 rtsp_st->feedback = 1;
407 /* XXX: handle list of formats */
408 get_word(buf1, sizeof(buf1), &p); /* format list */
409 rtsp_st->sdp_payload_type = atoi(buf1);
411 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
412 /* no corresponding stream */
413 if (rt->transport == RTSP_TRANSPORT_RAW) {
414 if (!rt->ts && CONFIG_RTPDEC)
415 rt->ts = ff_mpegts_parse_open(s);
417 RTPDynamicProtocolHandler *handler;
418 handler = ff_rtp_handler_find_by_id(
419 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
420 init_rtp_handler(handler, rtsp_st, NULL);
421 if (handler && handler->init)
422 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
424 } else if (rt->server_type == RTSP_SERVER_WMS &&
425 codec_type == AVMEDIA_TYPE_DATA) {
426 /* RTX stream, a stream that carries all the other actual
427 * audio/video streams. Don't expose this to the callers. */
429 st = avformat_new_stream(s, NULL);
432 st->id = rt->nb_rtsp_streams - 1;
433 rtsp_st->stream_index = st->index;
434 st->codec->codec_type = codec_type;
435 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
436 RTPDynamicProtocolHandler *handler;
437 /* if standard payload type, we can find the codec right now */
438 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
439 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
440 st->codec->sample_rate > 0)
441 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
442 /* Even static payload types may need a custom depacketizer */
443 handler = ff_rtp_handler_find_by_id(
444 rtsp_st->sdp_payload_type, st->codec->codec_type);
445 init_rtp_handler(handler, rtsp_st, st->codec);
446 if (handler && handler->init)
447 handler->init(s, st->index,
448 rtsp_st->dynamic_protocol_context);
451 /* put a default control url */
452 av_strlcpy(rtsp_st->control_url, rt->control_uri,
453 sizeof(rtsp_st->control_url));
456 if (av_strstart(p, "control:", &p)) {
457 if (s->nb_streams == 0) {
458 if (!strncmp(p, "rtsp://", 7))
459 av_strlcpy(rt->control_uri, p,
460 sizeof(rt->control_uri));
463 /* get the control url */
464 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
466 /* XXX: may need to add full url resolution */
467 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
469 if (proto[0] == '\0') {
470 /* relative control URL */
471 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
472 av_strlcat(rtsp_st->control_url, "/",
473 sizeof(rtsp_st->control_url));
474 av_strlcat(rtsp_st->control_url, p,
475 sizeof(rtsp_st->control_url));
477 av_strlcpy(rtsp_st->control_url, p,
478 sizeof(rtsp_st->control_url));
480 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
481 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
482 get_word(buf1, sizeof(buf1), &p);
483 payload_type = atoi(buf1);
484 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
485 if (rtsp_st->stream_index >= 0) {
486 st = s->streams[rtsp_st->stream_index];
487 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
489 } else if (av_strstart(p, "fmtp:", &p) ||
490 av_strstart(p, "framesize:", &p)) {
491 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
492 // let dynamic protocol handlers have a stab at the line.
493 get_word(buf1, sizeof(buf1), &p);
494 payload_type = atoi(buf1);
495 for (i = 0; i < rt->nb_rtsp_streams; i++) {
496 rtsp_st = rt->rtsp_streams[i];
497 if (rtsp_st->sdp_payload_type == payload_type &&
498 rtsp_st->dynamic_handler &&
499 rtsp_st->dynamic_handler->parse_sdp_a_line)
500 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
501 rtsp_st->dynamic_protocol_context, buf);
503 } else if (av_strstart(p, "range:", &p)) {
506 // this is so that seeking on a streamed file can work.
507 rtsp_parse_range_npt(p, &start, &end);
508 s->start_time = start;
509 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
510 s->duration = (end == AV_NOPTS_VALUE) ?
511 AV_NOPTS_VALUE : end - start;
512 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
514 rt->transport = RTSP_TRANSPORT_RDT;
515 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
517 st = s->streams[s->nb_streams - 1];
518 st->codec->sample_rate = atoi(p);
519 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
521 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
522 get_word(buf1, sizeof(buf1), &p); // ignore tag
523 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
524 p += strspn(p, SPACE_CHARS);
525 if (av_strstart(p, "inline:", &p))
526 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
527 } else if (av_strstart(p, "source-filter:", &p)) {
529 get_word(buf1, sizeof(buf1), &p);
530 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
532 exclude = !strcmp(buf1, "excl");
534 get_word(buf1, sizeof(buf1), &p);
535 if (strcmp(buf1, "IN") != 0)
537 get_word(buf1, sizeof(buf1), &p);
538 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
540 // not checking that the destination address actually matches or is wildcard
541 get_word(buf1, sizeof(buf1), &p);
544 rtsp_src = av_mallocz(sizeof(*rtsp_src));
547 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
549 if (s->nb_streams == 0) {
550 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
552 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
553 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
556 if (s->nb_streams == 0) {
557 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
559 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
560 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
565 if (rt->server_type == RTSP_SERVER_WMS)
566 ff_wms_parse_sdp_a_line(s, p);
567 if (s->nb_streams > 0) {
568 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
570 if (rt->server_type == RTSP_SERVER_REAL)
571 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
573 if (rtsp_st->dynamic_handler &&
574 rtsp_st->dynamic_handler->parse_sdp_a_line)
575 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
576 rtsp_st->stream_index,
577 rtsp_st->dynamic_protocol_context, buf);
584 int ff_sdp_parse(AVFormatContext *s, const char *content)
586 RTSPState *rt = s->priv_data;
589 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
590 * contain long SDP lines containing complete ASF Headers (several
591 * kB) or arrays of MDPR (RM stream descriptor) headers plus
592 * "rulebooks" describing their properties. Therefore, the SDP line
595 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
596 * in rtpdec_xiph.c. */
598 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
602 p += strspn(p, SPACE_CHARS);
610 /* get the content */
612 while (*p != '\n' && *p != '\r' && *p != '\0') {
613 if ((q - buf) < sizeof(buf) - 1)
618 sdp_parse_line(s, s1, letter, buf);
620 while (*p != '\n' && *p != '\0')
626 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
627 av_free(s1->default_include_source_addrs[i]);
628 av_freep(&s1->default_include_source_addrs);
629 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
630 av_free(s1->default_exclude_source_addrs[i]);
631 av_freep(&s1->default_exclude_source_addrs);
633 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
634 if (!rt->p) return AVERROR(ENOMEM);
637 #endif /* CONFIG_RTPDEC */
639 void ff_rtsp_undo_setup(AVFormatContext *s)
641 RTSPState *rt = s->priv_data;
644 for (i = 0; i < rt->nb_rtsp_streams; i++) {
645 RTSPStream *rtsp_st = rt->rtsp_streams[i];
648 if (rtsp_st->transport_priv) {
650 AVFormatContext *rtpctx = rtsp_st->transport_priv;
651 av_write_trailer(rtpctx);
652 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
654 avio_close_dyn_buf(rtpctx->pb, &ptr);
657 avio_close(rtpctx->pb);
659 avformat_free_context(rtpctx);
660 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
661 ff_rdt_parse_close(rtsp_st->transport_priv);
662 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
663 ff_rtp_parse_close(rtsp_st->transport_priv);
665 rtsp_st->transport_priv = NULL;
666 if (rtsp_st->rtp_handle)
667 ffurl_close(rtsp_st->rtp_handle);
668 rtsp_st->rtp_handle = NULL;
672 /* close and free RTSP streams */
673 void ff_rtsp_close_streams(AVFormatContext *s)
675 RTSPState *rt = s->priv_data;
679 ff_rtsp_undo_setup(s);
680 for (i = 0; i < rt->nb_rtsp_streams; i++) {
681 rtsp_st = rt->rtsp_streams[i];
683 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
684 rtsp_st->dynamic_handler->free(
685 rtsp_st->dynamic_protocol_context);
686 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
687 av_free(rtsp_st->include_source_addrs[j]);
688 av_freep(&rtsp_st->include_source_addrs);
689 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
690 av_free(rtsp_st->exclude_source_addrs[j]);
691 av_freep(&rtsp_st->exclude_source_addrs);
696 av_free(rt->rtsp_streams);
698 avformat_close_input(&rt->asf_ctx);
700 if (rt->ts && CONFIG_RTPDEC)
701 ff_mpegts_parse_close(rt->ts);
703 av_free(rt->recvbuf);
706 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
708 RTSPState *rt = s->priv_data;
710 int reordering_queue_size = rt->reordering_queue_size;
711 if (reordering_queue_size < 0) {
712 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
713 reordering_queue_size = 0;
715 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
718 /* open the RTP context */
719 if (rtsp_st->stream_index >= 0)
720 st = s->streams[rtsp_st->stream_index];
722 s->ctx_flags |= AVFMTCTX_NOHEADER;
724 if (s->oformat && CONFIG_RTSP_MUXER) {
725 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
727 RTSP_TCP_MAX_PACKET_SIZE,
728 rtsp_st->stream_index);
729 /* Ownership of rtp_handle is passed to the rtp mux context */
730 rtsp_st->rtp_handle = NULL;
733 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
734 return 0; // Don't need to open any parser here
735 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
736 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
737 rtsp_st->dynamic_protocol_context,
738 rtsp_st->dynamic_handler);
739 else if (CONFIG_RTPDEC)
740 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
741 rtsp_st->sdp_payload_type,
742 reordering_queue_size);
744 if (!rtsp_st->transport_priv) {
745 return AVERROR(ENOMEM);
746 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
747 if (rtsp_st->dynamic_handler) {
748 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
749 rtsp_st->dynamic_protocol_context,
750 rtsp_st->dynamic_handler);
752 if (rtsp_st->crypto_suite[0])
753 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
754 rtsp_st->crypto_suite,
755 rtsp_st->crypto_params);
761 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
762 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
769 q += strspn(q, SPACE_CHARS);
770 v = strtol(q, &p, 10);
774 v = strtol(p, &p, 10);
783 /* XXX: only one transport specification is parsed */
784 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
786 char transport_protocol[16];
788 char lower_transport[16];
790 RTSPTransportField *th;
793 reply->nb_transports = 0;
796 p += strspn(p, SPACE_CHARS);
800 th = &reply->transports[reply->nb_transports];
802 get_word_sep(transport_protocol, sizeof(transport_protocol),
804 if (!av_strcasecmp (transport_protocol, "rtp")) {
805 get_word_sep(profile, sizeof(profile), "/;,", &p);
806 lower_transport[0] = '\0';
807 /* rtp/avp/<protocol> */
809 get_word_sep(lower_transport, sizeof(lower_transport),
812 th->transport = RTSP_TRANSPORT_RTP;
813 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
814 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
815 /* x-pn-tng/<protocol> */
816 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
818 th->transport = RTSP_TRANSPORT_RDT;
819 } else if (!av_strcasecmp(transport_protocol, "raw")) {
820 get_word_sep(profile, sizeof(profile), "/;,", &p);
821 lower_transport[0] = '\0';
822 /* raw/raw/<protocol> */
824 get_word_sep(lower_transport, sizeof(lower_transport),
827 th->transport = RTSP_TRANSPORT_RAW;
829 if (!av_strcasecmp(lower_transport, "TCP"))
830 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
832 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
836 /* get each parameter */
837 while (*p != '\0' && *p != ',') {
838 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
839 if (!strcmp(parameter, "port")) {
842 rtsp_parse_range(&th->port_min, &th->port_max, &p);
844 } else if (!strcmp(parameter, "client_port")) {
847 rtsp_parse_range(&th->client_port_min,
848 &th->client_port_max, &p);
850 } else if (!strcmp(parameter, "server_port")) {
853 rtsp_parse_range(&th->server_port_min,
854 &th->server_port_max, &p);
856 } else if (!strcmp(parameter, "interleaved")) {
859 rtsp_parse_range(&th->interleaved_min,
860 &th->interleaved_max, &p);
862 } else if (!strcmp(parameter, "multicast")) {
863 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
864 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
865 } else if (!strcmp(parameter, "ttl")) {
869 th->ttl = strtol(p, &end, 10);
872 } else if (!strcmp(parameter, "destination")) {
875 get_word_sep(buf, sizeof(buf), ";,", &p);
876 get_sockaddr(buf, &th->destination);
878 } else if (!strcmp(parameter, "source")) {
881 get_word_sep(buf, sizeof(buf), ";,", &p);
882 av_strlcpy(th->source, buf, sizeof(th->source));
884 } else if (!strcmp(parameter, "mode")) {
887 get_word_sep(buf, sizeof(buf), ";, ", &p);
888 if (!strcmp(buf, "record") ||
889 !strcmp(buf, "receive"))
894 while (*p != ';' && *p != '\0' && *p != ',')
902 reply->nb_transports++;
906 static void handle_rtp_info(RTSPState *rt, const char *url,
907 uint32_t seq, uint32_t rtptime)
910 if (!rtptime || !url[0])
912 if (rt->transport != RTSP_TRANSPORT_RTP)
914 for (i = 0; i < rt->nb_rtsp_streams; i++) {
915 RTSPStream *rtsp_st = rt->rtsp_streams[i];
916 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
919 if (!strcmp(rtsp_st->control_url, url)) {
920 rtpctx->base_timestamp = rtptime;
926 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
929 char key[20], value[1024], url[1024] = "";
930 uint32_t seq = 0, rtptime = 0;
933 p += strspn(p, SPACE_CHARS);
936 get_word_sep(key, sizeof(key), "=", &p);
940 get_word_sep(value, sizeof(value), ";, ", &p);
942 if (!strcmp(key, "url"))
943 av_strlcpy(url, value, sizeof(url));
944 else if (!strcmp(key, "seq"))
945 seq = strtoul(value, NULL, 10);
946 else if (!strcmp(key, "rtptime"))
947 rtptime = strtoul(value, NULL, 10);
949 handle_rtp_info(rt, url, seq, rtptime);
958 handle_rtp_info(rt, url, seq, rtptime);
961 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
962 RTSPState *rt, const char *method)
966 /* NOTE: we do case independent match for broken servers */
968 if (av_stristart(p, "Session:", &p)) {
970 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
971 if (av_stristart(p, ";timeout=", &p) &&
972 (t = strtol(p, NULL, 10)) > 0) {
975 } else if (av_stristart(p, "Content-Length:", &p)) {
976 reply->content_length = strtol(p, NULL, 10);
977 } else if (av_stristart(p, "Transport:", &p)) {
978 rtsp_parse_transport(reply, p);
979 } else if (av_stristart(p, "CSeq:", &p)) {
980 reply->seq = strtol(p, NULL, 10);
981 } else if (av_stristart(p, "Range:", &p)) {
982 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
983 } else if (av_stristart(p, "RealChallenge1:", &p)) {
984 p += strspn(p, SPACE_CHARS);
985 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
986 } else if (av_stristart(p, "Server:", &p)) {
987 p += strspn(p, SPACE_CHARS);
988 av_strlcpy(reply->server, p, sizeof(reply->server));
989 } else if (av_stristart(p, "Notice:", &p) ||
990 av_stristart(p, "X-Notice:", &p)) {
991 reply->notice = strtol(p, NULL, 10);
992 } else if (av_stristart(p, "Location:", &p)) {
993 p += strspn(p, SPACE_CHARS);
994 av_strlcpy(reply->location, p , sizeof(reply->location));
995 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
996 p += strspn(p, SPACE_CHARS);
997 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
998 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
999 p += strspn(p, SPACE_CHARS);
1000 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1001 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1002 p += strspn(p, SPACE_CHARS);
1003 if (method && !strcmp(method, "DESCRIBE"))
1004 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1005 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1006 p += strspn(p, SPACE_CHARS);
1007 if (method && !strcmp(method, "PLAY"))
1008 rtsp_parse_rtp_info(rt, p);
1009 } else if (av_stristart(p, "Public:", &p) && rt) {
1010 if (strstr(p, "GET_PARAMETER") &&
1011 method && !strcmp(method, "OPTIONS"))
1012 rt->get_parameter_supported = 1;
1013 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1014 p += strspn(p, SPACE_CHARS);
1015 rt->accept_dynamic_rate = atoi(p);
1016 } else if (av_stristart(p, "Content-Type:", &p)) {
1017 p += strspn(p, SPACE_CHARS);
1018 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1022 /* skip a RTP/TCP interleaved packet */
1023 void ff_rtsp_skip_packet(AVFormatContext *s)
1025 RTSPState *rt = s->priv_data;
1029 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1032 len = AV_RB16(buf + 1);
1034 av_dlog(s, "skipping RTP packet len=%d\n", len);
1039 if (len1 > sizeof(buf))
1041 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1048 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1049 unsigned char **content_ptr,
1050 int return_on_interleaved_data, const char *method)
1052 RTSPState *rt = s->priv_data;
1053 char buf[4096], buf1[1024], *q;
1056 int ret, content_length, line_count = 0, request = 0;
1057 unsigned char *content = NULL;
1063 memset(reply, 0, sizeof(*reply));
1065 /* parse reply (XXX: use buffers) */
1066 rt->last_reply[0] = '\0';
1070 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1071 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1077 /* XXX: only parse it if first char on line ? */
1078 if (return_on_interleaved_data) {
1081 ff_rtsp_skip_packet(s);
1082 } else if (ch != '\r') {
1083 if ((q - buf) < sizeof(buf) - 1)
1089 av_dlog(s, "line='%s'\n", buf);
1091 /* test if last line */
1095 if (line_count == 0) {
1096 /* get reply code */
1097 get_word(buf1, sizeof(buf1), &p);
1098 if (!strncmp(buf1, "RTSP/", 5)) {
1099 get_word(buf1, sizeof(buf1), &p);
1100 reply->status_code = atoi(buf1);
1101 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1103 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1104 get_word(buf1, sizeof(buf1), &p); // object
1108 ff_rtsp_parse_line(reply, p, rt, method);
1109 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1110 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1115 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1116 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1118 content_length = reply->content_length;
1119 if (content_length > 0) {
1120 /* leave some room for a trailing '\0' (useful for simple parsing) */
1121 content = av_malloc(content_length + 1);
1122 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1123 content[content_length] = '\0';
1126 *content_ptr = content;
1132 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1133 const char* ptr = buf;
1135 if (!strcmp(reply->reason, "OPTIONS")) {
1136 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1138 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1139 if (reply->session_id[0])
1140 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1143 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1145 av_strlcat(buf, "\r\n", sizeof(buf));
1147 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1148 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1151 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1153 rt->last_cmd_time = av_gettime();
1154 /* Even if the request from the server had data, it is not the data
1155 * that the caller wants or expects. The memory could also be leaked
1156 * if the actual following reply has content data. */
1158 av_freep(content_ptr);
1159 /* If method is set, this is called from ff_rtsp_send_cmd,
1160 * where a reply to exactly this request is awaited. For
1161 * callers from within packet receiving, we just want to
1162 * return to the caller and go back to receiving packets. */
1168 if (rt->seq != reply->seq) {
1169 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1170 rt->seq, reply->seq);
1174 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1175 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1176 reply->notice == 2306 /* Continuous Feed Terminated */) {
1177 rt->state = RTSP_STATE_IDLE;
1178 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1179 return AVERROR(EIO); /* data or server error */
1180 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1181 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1182 return AVERROR(EPERM);
1188 * Send a command to the RTSP server without waiting for the reply.
1190 * @param s RTSP (de)muxer context
1191 * @param method the method for the request
1192 * @param url the target url for the request
1193 * @param headers extra header lines to include in the request
1194 * @param send_content if non-null, the data to send as request body content
1195 * @param send_content_length the length of the send_content data, or 0 if
1196 * send_content is null
1198 * @return zero if success, nonzero otherwise
1200 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1201 const char *method, const char *url,
1202 const char *headers,
1203 const unsigned char *send_content,
1204 int send_content_length)
1206 RTSPState *rt = s->priv_data;
1207 char buf[4096], *out_buf;
1208 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1210 /* Add in RTSP headers */
1213 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1215 av_strlcat(buf, headers, sizeof(buf));
1216 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1217 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1218 if (rt->session_id[0] != '\0' && (!headers ||
1219 !strstr(headers, "\nIf-Match:"))) {
1220 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1223 char *str = ff_http_auth_create_response(&rt->auth_state,
1224 rt->auth, url, method);
1226 av_strlcat(buf, str, sizeof(buf));
1229 if (send_content_length > 0 && send_content)
1230 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1231 av_strlcat(buf, "\r\n", sizeof(buf));
1233 /* base64 encode rtsp if tunneling */
1234 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1235 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1236 out_buf = base64buf;
1239 av_dlog(s, "Sending:\n%s--\n", buf);
1241 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1242 if (send_content_length > 0 && send_content) {
1243 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1244 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1245 "with content data not supported\n");
1246 return AVERROR_PATCHWELCOME;
1248 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1250 rt->last_cmd_time = av_gettime();
1255 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1256 const char *url, const char *headers)
1258 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1261 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1262 const char *headers, RTSPMessageHeader *reply,
1263 unsigned char **content_ptr)
1265 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1266 content_ptr, NULL, 0);
1269 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1270 const char *method, const char *url,
1272 RTSPMessageHeader *reply,
1273 unsigned char **content_ptr,
1274 const unsigned char *send_content,
1275 int send_content_length)
1277 RTSPState *rt = s->priv_data;
1278 HTTPAuthType cur_auth_type;
1279 int ret, attempts = 0;
1282 cur_auth_type = rt->auth_state.auth_type;
1283 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1285 send_content_length)))
1288 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1292 if (reply->status_code == 401 &&
1293 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1294 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1297 if (reply->status_code > 400){
1298 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1302 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1308 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1309 int lower_transport, const char *real_challenge)
1311 RTSPState *rt = s->priv_data;
1312 int rtx = 0, j, i, err, interleave = 0, port_off;
1313 RTSPStream *rtsp_st;
1314 RTSPMessageHeader reply1, *reply = &reply1;
1316 const char *trans_pref;
1318 if (rt->transport == RTSP_TRANSPORT_RDT)
1319 trans_pref = "x-pn-tng";
1320 else if (rt->transport == RTSP_TRANSPORT_RAW)
1321 trans_pref = "RAW/RAW";
1323 trans_pref = "RTP/AVP";
1325 /* default timeout: 1 minute */
1328 /* for each stream, make the setup request */
1329 /* XXX: we assume the same server is used for the control of each
1332 /* Choose a random starting offset within the first half of the
1333 * port range, to allow for a number of ports to try even if the offset
1334 * happens to be at the end of the random range. */
1335 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1336 /* even random offset */
1337 port_off -= port_off & 0x01;
1339 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1340 char transport[2048];
1343 * WMS serves all UDP data over a single connection, the RTX, which
1344 * isn't necessarily the first in the SDP but has to be the first
1345 * to be set up, else the second/third SETUP will fail with a 461.
1347 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1348 rt->server_type == RTSP_SERVER_WMS) {
1351 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1352 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1354 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1358 if (rtx == rt->nb_rtsp_streams)
1359 return -1; /* no RTX found */
1360 rtsp_st = rt->rtsp_streams[rtx];
1362 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1364 rtsp_st = rt->rtsp_streams[i];
1367 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1370 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1371 port = reply->transports[0].client_port_min;
1375 /* first try in specified port range */
1376 while (j <= rt->rtp_port_max) {
1377 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1378 "?localport=%d", j);
1379 /* we will use two ports per rtp stream (rtp and rtcp) */
1381 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1382 &s->interrupt_callback, NULL))
1386 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1391 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1393 snprintf(transport, sizeof(transport) - 1,
1394 "%s/UDP;", trans_pref);
1395 if (rt->server_type != RTSP_SERVER_REAL)
1396 av_strlcat(transport, "unicast;", sizeof(transport));
1397 av_strlcatf(transport, sizeof(transport),
1398 "client_port=%d", port);
1399 if (rt->transport == RTSP_TRANSPORT_RTP &&
1400 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1401 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1405 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1406 /* For WMS streams, the application streams are only used for
1407 * UDP. When trying to set it up for TCP streams, the server
1408 * will return an error. Therefore, we skip those streams. */
1409 if (rt->server_type == RTSP_SERVER_WMS &&
1410 (rtsp_st->stream_index < 0 ||
1411 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1414 snprintf(transport, sizeof(transport) - 1,
1415 "%s/TCP;", trans_pref);
1416 if (rt->transport != RTSP_TRANSPORT_RDT)
1417 av_strlcat(transport, "unicast;", sizeof(transport));
1418 av_strlcatf(transport, sizeof(transport),
1419 "interleaved=%d-%d",
1420 interleave, interleave + 1);
1424 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1425 snprintf(transport, sizeof(transport) - 1,
1426 "%s/UDP;multicast", trans_pref);
1429 av_strlcat(transport, ";mode=record", sizeof(transport));
1430 } else if (rt->server_type == RTSP_SERVER_REAL ||
1431 rt->server_type == RTSP_SERVER_WMS)
1432 av_strlcat(transport, ";mode=play", sizeof(transport));
1433 snprintf(cmd, sizeof(cmd),
1434 "Transport: %s\r\n",
1436 if (rt->accept_dynamic_rate)
1437 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1438 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1439 char real_res[41], real_csum[9];
1440 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1442 av_strlcatf(cmd, sizeof(cmd),
1444 "RealChallenge2: %s, sd=%s\r\n",
1445 rt->session_id, real_res, real_csum);
1447 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1448 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1451 } else if (reply->status_code != RTSP_STATUS_OK ||
1452 reply->nb_transports != 1) {
1453 err = AVERROR_INVALIDDATA;
1457 /* XXX: same protocol for all streams is required */
1459 if (reply->transports[0].lower_transport != rt->lower_transport ||
1460 reply->transports[0].transport != rt->transport) {
1461 err = AVERROR_INVALIDDATA;
1465 rt->lower_transport = reply->transports[0].lower_transport;
1466 rt->transport = reply->transports[0].transport;
1469 /* Fail if the server responded with another lower transport mode
1470 * than what we requested. */
1471 if (reply->transports[0].lower_transport != lower_transport) {
1472 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1473 err = AVERROR_INVALIDDATA;
1477 switch(reply->transports[0].lower_transport) {
1478 case RTSP_LOWER_TRANSPORT_TCP:
1479 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1480 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1483 case RTSP_LOWER_TRANSPORT_UDP: {
1484 char url[1024], options[30] = "";
1486 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1487 av_strlcpy(options, "?connect=1", sizeof(options));
1488 /* Use source address if specified */
1489 if (reply->transports[0].source[0]) {
1490 ff_url_join(url, sizeof(url), "rtp", NULL,
1491 reply->transports[0].source,
1492 reply->transports[0].server_port_min, "%s", options);
1494 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1495 reply->transports[0].server_port_min, "%s", options);
1497 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1498 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1499 err = AVERROR_INVALIDDATA;
1502 /* Try to initialize the connection state in a
1503 * potential NAT router by sending dummy packets.
1504 * RTP/RTCP dummy packets are used for RDT, too.
1506 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1508 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1511 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1512 char url[1024], namebuf[50], optbuf[20] = "";
1513 struct sockaddr_storage addr;
1516 if (reply->transports[0].destination.ss_family) {
1517 addr = reply->transports[0].destination;
1518 port = reply->transports[0].port_min;
1519 ttl = reply->transports[0].ttl;
1521 addr = rtsp_st->sdp_ip;
1522 port = rtsp_st->sdp_port;
1523 ttl = rtsp_st->sdp_ttl;
1526 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1527 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1528 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1529 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1530 port, "%s", optbuf);
1531 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1532 &s->interrupt_callback, NULL) < 0) {
1533 err = AVERROR_INVALIDDATA;
1540 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1544 if (rt->nb_rtsp_streams && reply->timeout > 0)
1545 rt->timeout = reply->timeout;
1547 if (rt->server_type == RTSP_SERVER_REAL)
1548 rt->need_subscription = 1;
1553 ff_rtsp_undo_setup(s);
1557 void ff_rtsp_close_connections(AVFormatContext *s)
1559 RTSPState *rt = s->priv_data;
1560 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1561 ffurl_close(rt->rtsp_hd);
1562 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1565 int ff_rtsp_connect(AVFormatContext *s)
1567 RTSPState *rt = s->priv_data;
1568 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1569 int port, err, tcp_fd;
1570 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1571 int lower_transport_mask = 0;
1572 char real_challenge[64] = "";
1573 struct sockaddr_storage peer;
1574 socklen_t peer_len = sizeof(peer);
1576 if (rt->rtp_port_max < rt->rtp_port_min) {
1577 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1578 "than min port %d\n", rt->rtp_port_max,
1580 return AVERROR(EINVAL);
1583 if (!ff_network_init())
1584 return AVERROR(EIO);
1586 if (s->max_delay < 0) /* Not set by the caller */
1587 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1589 rt->control_transport = RTSP_MODE_PLAIN;
1590 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1591 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1592 rt->control_transport = RTSP_MODE_TUNNEL;
1594 /* Only pass through valid flags from here */
1595 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1598 lower_transport_mask = rt->lower_transport_mask;
1599 /* extract hostname and port */
1600 av_url_split(NULL, 0, auth, sizeof(auth),
1601 host, sizeof(host), &port, path, sizeof(path), s->filename);
1603 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1606 port = RTSP_DEFAULT_PORT;
1608 if (!lower_transport_mask)
1609 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1612 /* Only UDP or TCP - UDP multicast isn't supported. */
1613 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1614 (1 << RTSP_LOWER_TRANSPORT_TCP);
1615 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1616 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1617 "only UDP and TCP are supported for output.\n");
1618 err = AVERROR(EINVAL);
1623 /* Construct the URI used in request; this is similar to s->filename,
1624 * but with authentication credentials removed and RTSP specific options
1626 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1627 host, port, "%s", path);
1629 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1630 /* set up initial handshake for tunneling */
1631 char httpname[1024];
1632 char sessioncookie[17];
1635 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1636 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1637 av_get_random_seed(), av_get_random_seed());
1640 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1641 &s->interrupt_callback) < 0) {
1646 /* generate GET headers */
1647 snprintf(headers, sizeof(headers),
1648 "x-sessioncookie: %s\r\n"
1649 "Accept: application/x-rtsp-tunnelled\r\n"
1650 "Pragma: no-cache\r\n"
1651 "Cache-Control: no-cache\r\n",
1653 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1655 /* complete the connection */
1656 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1662 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1663 &s->interrupt_callback) < 0 ) {
1668 /* generate POST headers */
1669 snprintf(headers, sizeof(headers),
1670 "x-sessioncookie: %s\r\n"
1671 "Content-Type: application/x-rtsp-tunnelled\r\n"
1672 "Pragma: no-cache\r\n"
1673 "Cache-Control: no-cache\r\n"
1674 "Content-Length: 32767\r\n"
1675 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1677 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1678 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1680 /* Initialize the authentication state for the POST session. The HTTP
1681 * protocol implementation doesn't properly handle multi-pass
1682 * authentication for POST requests, since it would require one of
1684 * - implementing Expect: 100-continue, which many HTTP servers
1685 * don't support anyway, even less the RTSP servers that do HTTP
1687 * - sending the whole POST data until getting a 401 reply specifying
1688 * what authentication method to use, then resending all that data
1689 * - waiting for potential 401 replies directly after sending the
1690 * POST header (waiting for some unspecified time)
1691 * Therefore, we copy the full auth state, which works for both basic
1692 * and digest. (For digest, we would have to synchronize the nonce
1693 * count variable between the two sessions, if we'd do more requests
1694 * with the original session, though.)
1696 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1698 /* complete the connection */
1699 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1704 /* open the tcp connection */
1705 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1706 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1707 &s->interrupt_callback, NULL) < 0) {
1711 rt->rtsp_hd_out = rt->rtsp_hd;
1715 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1716 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1717 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1718 NULL, 0, NI_NUMERICHOST);
1721 /* request options supported by the server; this also detects server
1723 for (rt->server_type = RTSP_SERVER_RTP;;) {
1725 if (rt->server_type == RTSP_SERVER_REAL)
1728 * The following entries are required for proper
1729 * streaming from a Realmedia server. They are
1730 * interdependent in some way although we currently
1731 * don't quite understand how. Values were copied
1732 * from mplayer SVN r23589.
1733 * ClientChallenge is a 16-byte ID in hex
1734 * CompanyID is a 16-byte ID in base64
1736 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1737 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1738 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1739 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1741 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1742 if (reply->status_code != RTSP_STATUS_OK) {
1743 err = AVERROR_INVALIDDATA;
1747 /* detect server type if not standard-compliant RTP */
1748 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1749 rt->server_type = RTSP_SERVER_REAL;
1751 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1752 rt->server_type = RTSP_SERVER_WMS;
1753 } else if (rt->server_type == RTSP_SERVER_REAL)
1754 strcpy(real_challenge, reply->real_challenge);
1758 if (s->iformat && CONFIG_RTSP_DEMUXER)
1759 err = ff_rtsp_setup_input_streams(s, reply);
1760 else if (CONFIG_RTSP_MUXER)
1761 err = ff_rtsp_setup_output_streams(s, host);
1766 int lower_transport = ff_log2_tab[lower_transport_mask &
1767 ~(lower_transport_mask - 1)];
1769 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1770 rt->server_type == RTSP_SERVER_REAL ?
1771 real_challenge : NULL);
1774 lower_transport_mask &= ~(1 << lower_transport);
1775 if (lower_transport_mask == 0 && err == 1) {
1776 err = AVERROR(EPROTONOSUPPORT);
1781 rt->lower_transport_mask = lower_transport_mask;
1782 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1783 rt->state = RTSP_STATE_IDLE;
1784 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1787 ff_rtsp_close_streams(s);
1788 ff_rtsp_close_connections(s);
1789 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1790 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1791 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1799 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1802 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1803 uint8_t *buf, int buf_size, int64_t wait_end)
1805 RTSPState *rt = s->priv_data;
1806 RTSPStream *rtsp_st;
1807 int n, i, ret, tcp_fd, timeout_cnt = 0;
1809 struct pollfd *p = rt->p;
1810 int *fds = NULL, fdsnum, fdsidx;
1813 if (ff_check_interrupt(&s->interrupt_callback))
1814 return AVERROR_EXIT;
1815 if (wait_end && wait_end - av_gettime() < 0)
1816 return AVERROR(EAGAIN);
1819 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1820 p[max_p].fd = tcp_fd;
1821 p[max_p++].events = POLLIN;
1825 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1826 rtsp_st = rt->rtsp_streams[i];
1827 if (rtsp_st->rtp_handle) {
1828 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1830 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1834 av_log(s, AV_LOG_ERROR,
1835 "Number of fds %d not supported\n", fdsnum);
1836 return AVERROR_INVALIDDATA;
1838 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1839 p[max_p].fd = fds[fdsidx];
1840 p[max_p++].events = POLLIN;
1845 n = poll(p, max_p, POLL_TIMEOUT_MS);
1847 int j = 1 - (tcp_fd == -1);
1849 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1850 rtsp_st = rt->rtsp_streams[i];
1851 if (rtsp_st->rtp_handle) {
1852 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1853 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1855 *prtsp_st = rtsp_st;
1862 #if CONFIG_RTSP_DEMUXER
1863 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1864 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1865 if (rt->state == RTSP_STATE_STREAMING) {
1866 if (!ff_rtsp_parse_streaming_commands(s))
1869 av_log(s, AV_LOG_WARNING,
1870 "Unable to answer to TEARDOWN\n");
1874 RTSPMessageHeader reply;
1875 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1878 /* XXX: parse message */
1879 if (rt->state != RTSP_STATE_STREAMING)
1884 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1885 return AVERROR(ETIMEDOUT);
1886 } else if (n < 0 && errno != EINTR)
1887 return AVERROR(errno);
1891 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1892 const uint8_t *buf, int len)
1894 RTSPState *rt = s->priv_data;
1898 if (rt->nb_rtsp_streams == 1) {
1899 *rtsp_st = rt->rtsp_streams[0];
1902 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1903 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1905 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1906 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1909 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1910 *rtsp_st = rt->rtsp_streams[i];
1917 av_log(s, AV_LOG_WARNING,
1918 "Unable to pick stream for packet - SSRC not known for "
1920 return AVERROR(EAGAIN);
1923 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1924 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1925 *rtsp_st = rt->rtsp_streams[i];
1931 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1932 return AVERROR(EAGAIN);
1935 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1937 RTSPState *rt = s->priv_data;
1939 RTSPStream *rtsp_st, *first_queue_st = NULL;
1940 int64_t wait_end = 0;
1942 if (rt->nb_byes == rt->nb_rtsp_streams)
1945 /* get next frames from the same RTP packet */
1946 if (rt->cur_transport_priv) {
1947 if (rt->transport == RTSP_TRANSPORT_RDT) {
1948 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1949 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1950 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1951 } else if (rt->ts && CONFIG_RTPDEC) {
1952 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1954 rt->recvbuf_pos += ret;
1955 ret = rt->recvbuf_pos < rt->recvbuf_len;
1960 rt->cur_transport_priv = NULL;
1962 } else if (ret == 1) {
1965 rt->cur_transport_priv = NULL;
1969 if (rt->transport == RTSP_TRANSPORT_RTP) {
1971 int64_t first_queue_time = 0;
1972 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1977 queue_time = ff_rtp_queued_packet_time(rtpctx);
1978 if (queue_time && (queue_time - first_queue_time < 0 ||
1979 !first_queue_time)) {
1980 first_queue_time = queue_time;
1981 first_queue_st = rt->rtsp_streams[i];
1984 if (first_queue_time) {
1985 wait_end = first_queue_time + s->max_delay;
1988 first_queue_st = NULL;
1992 /* read next RTP packet */
1994 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1996 return AVERROR(ENOMEM);
1999 switch(rt->lower_transport) {
2001 #if CONFIG_RTSP_DEMUXER
2002 case RTSP_LOWER_TRANSPORT_TCP:
2003 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2006 case RTSP_LOWER_TRANSPORT_UDP:
2007 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2008 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2009 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2010 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2012 case RTSP_LOWER_TRANSPORT_CUSTOM:
2013 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2014 wait_end && wait_end < av_gettime())
2015 len = AVERROR(EAGAIN);
2017 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2018 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2019 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2020 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2023 if (len == AVERROR(EAGAIN) && first_queue_st &&
2024 rt->transport == RTSP_TRANSPORT_RTP) {
2025 rtsp_st = first_queue_st;
2026 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2033 if (rt->transport == RTSP_TRANSPORT_RDT) {
2034 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2035 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2036 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2037 if (rtsp_st->feedback) {
2038 AVIOContext *pb = NULL;
2039 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2041 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2044 /* Either bad packet, or a RTCP packet. Check if the
2045 * first_rtcp_ntp_time field was initialized. */
2046 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2047 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2048 /* first_rtcp_ntp_time has been initialized for this stream,
2049 * copy the same value to all other uninitialized streams,
2050 * in order to map their timestamp origin to the same ntp time
2053 AVStream *st = NULL;
2054 if (rtsp_st->stream_index >= 0)
2055 st = s->streams[rtsp_st->stream_index];
2056 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2057 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2058 AVStream *st2 = NULL;
2059 if (rt->rtsp_streams[i]->stream_index >= 0)
2060 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2061 if (rtpctx2 && st && st2 &&
2062 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2063 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2064 rtpctx2->rtcp_ts_offset = av_rescale_q(
2065 rtpctx->rtcp_ts_offset, st->time_base,
2070 if (ret == -RTCP_BYE) {
2073 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2074 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2076 if (rt->nb_byes == rt->nb_rtsp_streams)
2080 } else if (rt->ts && CONFIG_RTPDEC) {
2081 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2084 rt->recvbuf_len = len;
2085 rt->recvbuf_pos = ret;
2086 rt->cur_transport_priv = rt->ts;
2093 return AVERROR_INVALIDDATA;
2099 /* more packets may follow, so we save the RTP context */
2100 rt->cur_transport_priv = rtsp_st->transport_priv;
2104 #endif /* CONFIG_RTPDEC */
2106 #if CONFIG_SDP_DEMUXER
2107 static int sdp_probe(AVProbeData *p1)
2109 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2111 /* we look for a line beginning "c=IN IP" */
2112 while (p < p_end && *p != '\0') {
2113 if (p + sizeof("c=IN IP") - 1 < p_end &&
2114 av_strstart(p, "c=IN IP", NULL))
2115 return AVPROBE_SCORE_EXTENSION;
2117 while (p < p_end - 1 && *p != '\n') p++;
2126 static void append_source_addrs(char *buf, int size, const char *name,
2127 int count, struct RTSPSource **addrs)
2132 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2133 for (i = 1; i < count; i++)
2134 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2137 static int sdp_read_header(AVFormatContext *s)
2139 RTSPState *rt = s->priv_data;
2140 RTSPStream *rtsp_st;
2145 if (!ff_network_init())
2146 return AVERROR(EIO);
2148 if (s->max_delay < 0) /* Not set by the caller */
2149 s->max_delay = DEFAULT_REORDERING_DELAY;
2150 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2151 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2153 /* read the whole sdp file */
2154 /* XXX: better loading */
2155 content = av_malloc(SDP_MAX_SIZE);
2156 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2159 return AVERROR_INVALIDDATA;
2161 content[size] ='\0';
2163 err = ff_sdp_parse(s, content);
2167 /* open each RTP stream */
2168 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2170 rtsp_st = rt->rtsp_streams[i];
2172 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2173 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2174 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2175 ff_url_join(url, sizeof(url), "rtp", NULL,
2176 namebuf, rtsp_st->sdp_port,
2177 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2179 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2181 append_source_addrs(url, sizeof(url), "sources",
2182 rtsp_st->nb_include_source_addrs,
2183 rtsp_st->include_source_addrs);
2184 append_source_addrs(url, sizeof(url), "block",
2185 rtsp_st->nb_exclude_source_addrs,
2186 rtsp_st->exclude_source_addrs);
2187 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2188 &s->interrupt_callback, NULL) < 0) {
2189 err = AVERROR_INVALIDDATA;
2193 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2198 ff_rtsp_close_streams(s);
2203 static int sdp_read_close(AVFormatContext *s)
2205 ff_rtsp_close_streams(s);
2210 static const AVClass sdp_demuxer_class = {
2211 .class_name = "SDP demuxer",
2212 .item_name = av_default_item_name,
2213 .option = sdp_options,
2214 .version = LIBAVUTIL_VERSION_INT,
2217 AVInputFormat ff_sdp_demuxer = {
2219 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2220 .priv_data_size = sizeof(RTSPState),
2221 .read_probe = sdp_probe,
2222 .read_header = sdp_read_header,
2223 .read_packet = ff_rtsp_fetch_packet,
2224 .read_close = sdp_read_close,
2225 .priv_class = &sdp_demuxer_class,
2227 #endif /* CONFIG_SDP_DEMUXER */
2229 #if CONFIG_RTP_DEMUXER
2230 static int rtp_probe(AVProbeData *p)
2232 if (av_strstart(p->filename, "rtp:", NULL))
2233 return AVPROBE_SCORE_MAX;
2237 static int rtp_read_header(AVFormatContext *s)
2239 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2240 char host[500], sdp[500];
2242 URLContext* in = NULL;
2244 AVCodecContext codec = { 0 };
2245 struct sockaddr_storage addr;
2247 socklen_t addrlen = sizeof(addr);
2248 RTSPState *rt = s->priv_data;
2250 if (!ff_network_init())
2251 return AVERROR(EIO);
2253 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2254 &s->interrupt_callback, NULL);
2259 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2260 if (ret == AVERROR(EAGAIN))
2265 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2269 if ((recvbuf[0] & 0xc0) != 0x80) {
2270 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2275 if (RTP_PT_IS_RTCP(recvbuf[1]))
2278 payload_type = recvbuf[1] & 0x7f;
2281 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2285 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2286 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2287 "without an SDP file describing it\n",
2291 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2292 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2293 "properly you need an SDP file "
2297 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2298 NULL, 0, s->filename);
2300 snprintf(sdp, sizeof(sdp),
2301 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2302 addr.ss_family == AF_INET ? 4 : 6, host,
2303 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2304 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2305 port, payload_type);
2306 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2308 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2311 /* sdp_read_header initializes this again */
2314 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2316 ret = sdp_read_header(s);
2327 static const AVClass rtp_demuxer_class = {
2328 .class_name = "RTP demuxer",
2329 .item_name = av_default_item_name,
2330 .option = rtp_options,
2331 .version = LIBAVUTIL_VERSION_INT,
2334 AVInputFormat ff_rtp_demuxer = {
2336 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2337 .priv_data_size = sizeof(RTSPState),
2338 .read_probe = rtp_probe,
2339 .read_header = rtp_read_header,
2340 .read_packet = ff_rtsp_fetch_packet,
2341 .read_close = sdp_read_close,
2342 .flags = AVFMT_NOFILE,
2343 .priv_class = &rtp_demuxer_class,
2345 #endif /* CONFIG_RTP_DEMUXER */