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"
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" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { 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" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
77 #define RTSP_REORDERING_OPTS() \
78 { "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 }
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "Don't 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", "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", "RTSP flags"),
89 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
90 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
91 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
92 { "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 },
93 RTSP_REORDERING_OPTS(),
97 static const AVOption sdp_options[] = {
98 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
99 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
100 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
101 RTSP_REORDERING_OPTS(),
105 static const AVOption rtp_options[] = {
106 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
107 RTSP_REORDERING_OPTS(),
111 static void get_word_until_chars(char *buf, int buf_size,
112 const char *sep, const char **pp)
118 p += strspn(p, SPACE_CHARS);
120 while (!strchr(sep, *p) && *p != '\0') {
121 if ((q - buf) < buf_size - 1)
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
133 if (**pp == '/') (*pp)++;
134 get_word_until_chars(buf, buf_size, sep, pp);
137 static void get_word(char *buf, int buf_size, const char **pp)
139 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
142 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
144 * Used for seeking in the rtp stream.
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
150 p += strspn(p, SPACE_CHARS);
151 if (!av_stristart(p, "npt=", &p))
154 *start = AV_NOPTS_VALUE;
155 *end = AV_NOPTS_VALUE;
157 get_word_sep(buf, sizeof(buf), "-", &p);
158 av_parse_time(start, buf, 1);
161 get_word_sep(buf, sizeof(buf), "-", &p);
162 av_parse_time(end, buf, 1);
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
168 struct addrinfo hints = { 0 }, *ai = NULL;
169 hints.ai_flags = AI_NUMERICHOST;
170 if (getaddrinfo(buf, NULL, &hints, &ai))
172 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179 RTSPStream *rtsp_st, AVCodecContext *codec)
183 codec->codec_id = handler->codec_id;
184 rtsp_st->dynamic_handler = handler;
185 if (handler->alloc) {
186 rtsp_st->dynamic_protocol_context = handler->alloc();
187 if (!rtsp_st->dynamic_protocol_context)
188 rtsp_st->dynamic_handler = NULL;
192 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
193 static int sdp_parse_rtpmap(AVFormatContext *s,
194 AVStream *st, RTSPStream *rtsp_st,
195 int payload_type, const char *p)
197 AVCodecContext *codec = st->codec;
203 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
204 * see if we can handle this kind of payload.
205 * The space should normally not be there but some Real streams or
206 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207 * have a trailing space. */
208 get_word_sep(buf, sizeof(buf), "/ ", &p);
209 if (payload_type < RTP_PT_PRIVATE) {
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 /* search into AVRtpPayloadTypes[] */
213 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
216 if (codec->codec_id == AV_CODEC_ID_NONE) {
217 RTPDynamicProtocolHandler *handler =
218 ff_rtp_handler_find_by_name(buf, codec->codec_type);
219 init_rtp_handler(handler, rtsp_st, codec);
220 /* If no dynamic handler was found, check with the list of standard
221 * allocated types, if such a stream for some reason happens to
222 * use a private payload type. This isn't handled in rtpdec.c, since
223 * the format name from the rtpmap line never is passed into rtpdec. */
224 if (!rtsp_st->dynamic_handler)
225 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
228 c = avcodec_find_decoder(codec->codec_id);
234 get_word_sep(buf, sizeof(buf), "/", &p);
236 switch (codec->codec_type) {
237 case AVMEDIA_TYPE_AUDIO:
238 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
239 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
240 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
242 codec->sample_rate = i;
243 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
244 get_word_sep(buf, sizeof(buf), "/", &p);
248 // TODO: there is a bug here; if it is a mono stream, and
249 // less than 22000Hz, faad upconverts to stereo and twice
250 // the frequency. No problem, but the sample rate is being
251 // set here by the sdp line. Patch on its way. (rdm)
253 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
255 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
258 case AVMEDIA_TYPE_VIDEO:
259 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
261 avpriv_set_pts_info(st, 32, 1, i);
266 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
267 rtsp_st->dynamic_handler->init(s, st->index,
268 rtsp_st->dynamic_protocol_context);
272 /* parse the attribute line from the fmtp a line of an sdp response. This
273 * is broken out as a function because it is used in rtp_h264.c, which is
275 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
276 char *value, int value_size)
278 *p += strspn(*p, SPACE_CHARS);
280 get_word_sep(attr, attr_size, "=", p);
283 get_word_sep(value, value_size, ";", p);
291 typedef struct SDPParseState {
293 struct sockaddr_storage default_ip;
295 int skip_media; ///< set if an unknown m= line occurs
298 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
299 int letter, const char *buf)
301 RTSPState *rt = s->priv_data;
302 char buf1[64], st_type[64];
304 enum AVMediaType codec_type;
308 struct sockaddr_storage sdp_ip;
311 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
314 if (s1->skip_media && letter != 'm')
318 get_word(buf1, sizeof(buf1), &p);
319 if (strcmp(buf1, "IN") != 0)
321 get_word(buf1, sizeof(buf1), &p);
322 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
324 get_word_sep(buf1, sizeof(buf1), "/", &p);
325 if (get_sockaddr(buf1, &sdp_ip))
330 get_word_sep(buf1, sizeof(buf1), "/", &p);
333 if (s->nb_streams == 0) {
334 s1->default_ip = sdp_ip;
335 s1->default_ttl = ttl;
337 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
338 rtsp_st->sdp_ip = sdp_ip;
339 rtsp_st->sdp_ttl = ttl;
343 av_dict_set(&s->metadata, "title", p, 0);
346 if (s->nb_streams == 0) {
347 av_dict_set(&s->metadata, "comment", p, 0);
354 codec_type = AVMEDIA_TYPE_UNKNOWN;
355 get_word(st_type, sizeof(st_type), &p);
356 if (!strcmp(st_type, "audio")) {
357 codec_type = AVMEDIA_TYPE_AUDIO;
358 } else if (!strcmp(st_type, "video")) {
359 codec_type = AVMEDIA_TYPE_VIDEO;
360 } else if (!strcmp(st_type, "application")) {
361 codec_type = AVMEDIA_TYPE_DATA;
363 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
367 rtsp_st = av_mallocz(sizeof(RTSPStream));
370 rtsp_st->stream_index = -1;
371 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
373 rtsp_st->sdp_ip = s1->default_ip;
374 rtsp_st->sdp_ttl = s1->default_ttl;
376 get_word(buf1, sizeof(buf1), &p); /* port */
377 rtsp_st->sdp_port = atoi(buf1);
379 get_word(buf1, sizeof(buf1), &p); /* protocol */
380 if (!strcmp(buf1, "udp"))
381 rt->transport = RTSP_TRANSPORT_RAW;
382 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
383 rtsp_st->feedback = 1;
385 /* XXX: handle list of formats */
386 get_word(buf1, sizeof(buf1), &p); /* format list */
387 rtsp_st->sdp_payload_type = atoi(buf1);
389 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
390 /* no corresponding stream */
391 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
392 rt->ts = ff_mpegts_parse_open(s);
393 } else if (rt->server_type == RTSP_SERVER_WMS &&
394 codec_type == AVMEDIA_TYPE_DATA) {
395 /* RTX stream, a stream that carries all the other actual
396 * audio/video streams. Don't expose this to the callers. */
398 st = avformat_new_stream(s, NULL);
401 st->id = rt->nb_rtsp_streams - 1;
402 rtsp_st->stream_index = st->index;
403 st->codec->codec_type = codec_type;
404 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
405 RTPDynamicProtocolHandler *handler;
406 /* if standard payload type, we can find the codec right now */
407 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
408 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
409 st->codec->sample_rate > 0)
410 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
411 /* Even static payload types may need a custom depacketizer */
412 handler = ff_rtp_handler_find_by_id(
413 rtsp_st->sdp_payload_type, st->codec->codec_type);
414 init_rtp_handler(handler, rtsp_st, st->codec);
415 if (handler && handler->init)
416 handler->init(s, st->index,
417 rtsp_st->dynamic_protocol_context);
420 /* put a default control url */
421 av_strlcpy(rtsp_st->control_url, rt->control_uri,
422 sizeof(rtsp_st->control_url));
425 if (av_strstart(p, "control:", &p)) {
426 if (s->nb_streams == 0) {
427 if (!strncmp(p, "rtsp://", 7))
428 av_strlcpy(rt->control_uri, p,
429 sizeof(rt->control_uri));
432 /* get the control url */
433 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
435 /* XXX: may need to add full url resolution */
436 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
438 if (proto[0] == '\0') {
439 /* relative control URL */
440 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
441 av_strlcat(rtsp_st->control_url, "/",
442 sizeof(rtsp_st->control_url));
443 av_strlcat(rtsp_st->control_url, p,
444 sizeof(rtsp_st->control_url));
446 av_strlcpy(rtsp_st->control_url, p,
447 sizeof(rtsp_st->control_url));
449 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
450 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
451 get_word(buf1, sizeof(buf1), &p);
452 payload_type = atoi(buf1);
453 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
454 if (rtsp_st->stream_index >= 0) {
455 st = s->streams[rtsp_st->stream_index];
456 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
458 } else if (av_strstart(p, "fmtp:", &p) ||
459 av_strstart(p, "framesize:", &p)) {
460 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
461 // let dynamic protocol handlers have a stab at the line.
462 get_word(buf1, sizeof(buf1), &p);
463 payload_type = atoi(buf1);
464 for (i = 0; i < rt->nb_rtsp_streams; i++) {
465 rtsp_st = rt->rtsp_streams[i];
466 if (rtsp_st->sdp_payload_type == payload_type &&
467 rtsp_st->dynamic_handler &&
468 rtsp_st->dynamic_handler->parse_sdp_a_line)
469 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
470 rtsp_st->dynamic_protocol_context, buf);
472 } else if (av_strstart(p, "range:", &p)) {
475 // this is so that seeking on a streamed file can work.
476 rtsp_parse_range_npt(p, &start, &end);
477 s->start_time = start;
478 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
479 s->duration = (end == AV_NOPTS_VALUE) ?
480 AV_NOPTS_VALUE : end - start;
481 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
483 rt->transport = RTSP_TRANSPORT_RDT;
484 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
486 st = s->streams[s->nb_streams - 1];
487 st->codec->sample_rate = atoi(p);
489 if (rt->server_type == RTSP_SERVER_WMS)
490 ff_wms_parse_sdp_a_line(s, p);
491 if (s->nb_streams > 0) {
492 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
494 if (rt->server_type == RTSP_SERVER_REAL)
495 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
497 if (rtsp_st->dynamic_handler &&
498 rtsp_st->dynamic_handler->parse_sdp_a_line)
499 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
500 rtsp_st->stream_index,
501 rtsp_st->dynamic_protocol_context, buf);
508 int ff_sdp_parse(AVFormatContext *s, const char *content)
510 RTSPState *rt = s->priv_data;
513 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
514 * contain long SDP lines containing complete ASF Headers (several
515 * kB) or arrays of MDPR (RM stream descriptor) headers plus
516 * "rulebooks" describing their properties. Therefore, the SDP line
519 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
520 * in rtpdec_xiph.c. */
522 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
526 p += strspn(p, SPACE_CHARS);
534 /* get the content */
536 while (*p != '\n' && *p != '\r' && *p != '\0') {
537 if ((q - buf) < sizeof(buf) - 1)
542 sdp_parse_line(s, s1, letter, buf);
544 while (*p != '\n' && *p != '\0')
549 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
550 if (!rt->p) return AVERROR(ENOMEM);
553 #endif /* CONFIG_RTPDEC */
555 void ff_rtsp_undo_setup(AVFormatContext *s)
557 RTSPState *rt = s->priv_data;
560 for (i = 0; i < rt->nb_rtsp_streams; i++) {
561 RTSPStream *rtsp_st = rt->rtsp_streams[i];
564 if (rtsp_st->transport_priv) {
566 AVFormatContext *rtpctx = rtsp_st->transport_priv;
567 av_write_trailer(rtpctx);
568 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
570 avio_close_dyn_buf(rtpctx->pb, &ptr);
573 avio_close(rtpctx->pb);
575 avformat_free_context(rtpctx);
576 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
577 ff_rdt_parse_close(rtsp_st->transport_priv);
578 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
579 ff_rtp_parse_close(rtsp_st->transport_priv);
581 rtsp_st->transport_priv = NULL;
582 if (rtsp_st->rtp_handle)
583 ffurl_close(rtsp_st->rtp_handle);
584 rtsp_st->rtp_handle = NULL;
588 /* close and free RTSP streams */
589 void ff_rtsp_close_streams(AVFormatContext *s)
591 RTSPState *rt = s->priv_data;
595 ff_rtsp_undo_setup(s);
596 for (i = 0; i < rt->nb_rtsp_streams; i++) {
597 rtsp_st = rt->rtsp_streams[i];
599 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
600 rtsp_st->dynamic_handler->free(
601 rtsp_st->dynamic_protocol_context);
605 av_free(rt->rtsp_streams);
607 avformat_close_input(&rt->asf_ctx);
609 if (rt->ts && CONFIG_RTPDEC)
610 ff_mpegts_parse_close(rt->ts);
612 av_free(rt->recvbuf);
615 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
617 RTSPState *rt = s->priv_data;
619 int reordering_queue_size = rt->reordering_queue_size;
620 if (reordering_queue_size < 0) {
621 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
622 reordering_queue_size = 0;
624 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
627 /* open the RTP context */
628 if (rtsp_st->stream_index >= 0)
629 st = s->streams[rtsp_st->stream_index];
631 s->ctx_flags |= AVFMTCTX_NOHEADER;
633 if (s->oformat && CONFIG_RTSP_MUXER) {
634 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
636 RTSP_TCP_MAX_PACKET_SIZE,
637 rtsp_st->stream_index);
638 /* Ownership of rtp_handle is passed to the rtp mux context */
639 rtsp_st->rtp_handle = NULL;
642 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
643 return 0; // Don't need to open any parser here
644 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
645 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
646 rtsp_st->dynamic_protocol_context,
647 rtsp_st->dynamic_handler);
648 else if (CONFIG_RTPDEC)
649 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
650 rtsp_st->sdp_payload_type,
651 reordering_queue_size);
653 if (!rtsp_st->transport_priv) {
654 return AVERROR(ENOMEM);
655 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
656 if (rtsp_st->dynamic_handler) {
657 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
658 rtsp_st->dynamic_protocol_context,
659 rtsp_st->dynamic_handler);
666 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
667 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
674 q += strspn(q, SPACE_CHARS);
675 v = strtol(q, &p, 10);
679 v = strtol(p, &p, 10);
688 /* XXX: only one transport specification is parsed */
689 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
691 char transport_protocol[16];
693 char lower_transport[16];
695 RTSPTransportField *th;
698 reply->nb_transports = 0;
701 p += strspn(p, SPACE_CHARS);
705 th = &reply->transports[reply->nb_transports];
707 get_word_sep(transport_protocol, sizeof(transport_protocol),
709 if (!av_strcasecmp (transport_protocol, "rtp")) {
710 get_word_sep(profile, sizeof(profile), "/;,", &p);
711 lower_transport[0] = '\0';
712 /* rtp/avp/<protocol> */
714 get_word_sep(lower_transport, sizeof(lower_transport),
717 th->transport = RTSP_TRANSPORT_RTP;
718 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
719 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
720 /* x-pn-tng/<protocol> */
721 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
723 th->transport = RTSP_TRANSPORT_RDT;
724 } else if (!av_strcasecmp(transport_protocol, "raw")) {
725 get_word_sep(profile, sizeof(profile), "/;,", &p);
726 lower_transport[0] = '\0';
727 /* raw/raw/<protocol> */
729 get_word_sep(lower_transport, sizeof(lower_transport),
732 th->transport = RTSP_TRANSPORT_RAW;
734 if (!av_strcasecmp(lower_transport, "TCP"))
735 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
737 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
741 /* get each parameter */
742 while (*p != '\0' && *p != ',') {
743 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
744 if (!strcmp(parameter, "port")) {
747 rtsp_parse_range(&th->port_min, &th->port_max, &p);
749 } else if (!strcmp(parameter, "client_port")) {
752 rtsp_parse_range(&th->client_port_min,
753 &th->client_port_max, &p);
755 } else if (!strcmp(parameter, "server_port")) {
758 rtsp_parse_range(&th->server_port_min,
759 &th->server_port_max, &p);
761 } else if (!strcmp(parameter, "interleaved")) {
764 rtsp_parse_range(&th->interleaved_min,
765 &th->interleaved_max, &p);
767 } else if (!strcmp(parameter, "multicast")) {
768 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
769 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
770 } else if (!strcmp(parameter, "ttl")) {
774 th->ttl = strtol(p, &end, 10);
777 } else if (!strcmp(parameter, "destination")) {
780 get_word_sep(buf, sizeof(buf), ";,", &p);
781 get_sockaddr(buf, &th->destination);
783 } else if (!strcmp(parameter, "source")) {
786 get_word_sep(buf, sizeof(buf), ";,", &p);
787 av_strlcpy(th->source, buf, sizeof(th->source));
789 } else if (!strcmp(parameter, "mode")) {
792 get_word_sep(buf, sizeof(buf), ";, ", &p);
793 if (!strcmp(buf, "record") ||
794 !strcmp(buf, "receive"))
799 while (*p != ';' && *p != '\0' && *p != ',')
807 reply->nb_transports++;
811 static void handle_rtp_info(RTSPState *rt, const char *url,
812 uint32_t seq, uint32_t rtptime)
815 if (!rtptime || !url[0])
817 if (rt->transport != RTSP_TRANSPORT_RTP)
819 for (i = 0; i < rt->nb_rtsp_streams; i++) {
820 RTSPStream *rtsp_st = rt->rtsp_streams[i];
821 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
824 if (!strcmp(rtsp_st->control_url, url)) {
825 rtpctx->base_timestamp = rtptime;
831 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
834 char key[20], value[1024], url[1024] = "";
835 uint32_t seq = 0, rtptime = 0;
838 p += strspn(p, SPACE_CHARS);
841 get_word_sep(key, sizeof(key), "=", &p);
845 get_word_sep(value, sizeof(value), ";, ", &p);
847 if (!strcmp(key, "url"))
848 av_strlcpy(url, value, sizeof(url));
849 else if (!strcmp(key, "seq"))
850 seq = strtoul(value, NULL, 10);
851 else if (!strcmp(key, "rtptime"))
852 rtptime = strtoul(value, NULL, 10);
854 handle_rtp_info(rt, url, seq, rtptime);
863 handle_rtp_info(rt, url, seq, rtptime);
866 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
867 RTSPState *rt, const char *method)
871 /* NOTE: we do case independent match for broken servers */
873 if (av_stristart(p, "Session:", &p)) {
875 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
876 if (av_stristart(p, ";timeout=", &p) &&
877 (t = strtol(p, NULL, 10)) > 0) {
880 } else if (av_stristart(p, "Content-Length:", &p)) {
881 reply->content_length = strtol(p, NULL, 10);
882 } else if (av_stristart(p, "Transport:", &p)) {
883 rtsp_parse_transport(reply, p);
884 } else if (av_stristart(p, "CSeq:", &p)) {
885 reply->seq = strtol(p, NULL, 10);
886 } else if (av_stristart(p, "Range:", &p)) {
887 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
888 } else if (av_stristart(p, "RealChallenge1:", &p)) {
889 p += strspn(p, SPACE_CHARS);
890 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
891 } else if (av_stristart(p, "Server:", &p)) {
892 p += strspn(p, SPACE_CHARS);
893 av_strlcpy(reply->server, p, sizeof(reply->server));
894 } else if (av_stristart(p, "Notice:", &p) ||
895 av_stristart(p, "X-Notice:", &p)) {
896 reply->notice = strtol(p, NULL, 10);
897 } else if (av_stristart(p, "Location:", &p)) {
898 p += strspn(p, SPACE_CHARS);
899 av_strlcpy(reply->location, p , sizeof(reply->location));
900 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
901 p += strspn(p, SPACE_CHARS);
902 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
903 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
904 p += strspn(p, SPACE_CHARS);
905 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
906 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
907 p += strspn(p, SPACE_CHARS);
908 if (method && !strcmp(method, "DESCRIBE"))
909 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
910 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
911 p += strspn(p, SPACE_CHARS);
912 if (method && !strcmp(method, "PLAY"))
913 rtsp_parse_rtp_info(rt, p);
914 } else if (av_stristart(p, "Public:", &p) && rt) {
915 if (strstr(p, "GET_PARAMETER") &&
916 method && !strcmp(method, "OPTIONS"))
917 rt->get_parameter_supported = 1;
918 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
919 p += strspn(p, SPACE_CHARS);
920 rt->accept_dynamic_rate = atoi(p);
921 } else if (av_stristart(p, "Content-Type:", &p)) {
922 p += strspn(p, SPACE_CHARS);
923 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
927 /* skip a RTP/TCP interleaved packet */
928 void ff_rtsp_skip_packet(AVFormatContext *s)
930 RTSPState *rt = s->priv_data;
934 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
937 len = AV_RB16(buf + 1);
939 av_dlog(s, "skipping RTP packet len=%d\n", len);
944 if (len1 > sizeof(buf))
946 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
953 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
954 unsigned char **content_ptr,
955 int return_on_interleaved_data, const char *method)
957 RTSPState *rt = s->priv_data;
958 char buf[4096], buf1[1024], *q;
961 int ret, content_length, line_count = 0, request = 0;
962 unsigned char *content = NULL;
968 memset(reply, 0, sizeof(*reply));
970 /* parse reply (XXX: use buffers) */
971 rt->last_reply[0] = '\0';
975 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
976 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
982 /* XXX: only parse it if first char on line ? */
983 if (return_on_interleaved_data) {
986 ff_rtsp_skip_packet(s);
987 } else if (ch != '\r') {
988 if ((q - buf) < sizeof(buf) - 1)
994 av_dlog(s, "line='%s'\n", buf);
996 /* test if last line */
1000 if (line_count == 0) {
1001 /* get reply code */
1002 get_word(buf1, sizeof(buf1), &p);
1003 if (!strncmp(buf1, "RTSP/", 5)) {
1004 get_word(buf1, sizeof(buf1), &p);
1005 reply->status_code = atoi(buf1);
1006 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1008 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1009 get_word(buf1, sizeof(buf1), &p); // object
1013 ff_rtsp_parse_line(reply, p, rt, method);
1014 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1015 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1020 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1021 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1023 content_length = reply->content_length;
1024 if (content_length > 0) {
1025 /* leave some room for a trailing '\0' (useful for simple parsing) */
1026 content = av_malloc(content_length + 1);
1027 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1028 content[content_length] = '\0';
1031 *content_ptr = content;
1037 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1038 const char* ptr = buf;
1040 if (!strcmp(reply->reason, "OPTIONS")) {
1041 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1043 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1044 if (reply->session_id[0])
1045 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1048 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1050 av_strlcat(buf, "\r\n", sizeof(buf));
1052 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1053 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1056 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1058 rt->last_cmd_time = av_gettime();
1059 /* Even if the request from the server had data, it is not the data
1060 * that the caller wants or expects. The memory could also be leaked
1061 * if the actual following reply has content data. */
1063 av_freep(content_ptr);
1064 /* If method is set, this is called from ff_rtsp_send_cmd,
1065 * where a reply to exactly this request is awaited. For
1066 * callers from within packet receiving, we just want to
1067 * return to the caller and go back to receiving packets. */
1073 if (rt->seq != reply->seq) {
1074 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1075 rt->seq, reply->seq);
1079 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1080 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1081 reply->notice == 2306 /* Continuous Feed Terminated */) {
1082 rt->state = RTSP_STATE_IDLE;
1083 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1084 return AVERROR(EIO); /* data or server error */
1085 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1086 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1087 return AVERROR(EPERM);
1093 * Send a command to the RTSP server without waiting for the reply.
1095 * @param s RTSP (de)muxer context
1096 * @param method the method for the request
1097 * @param url the target url for the request
1098 * @param headers extra header lines to include in the request
1099 * @param send_content if non-null, the data to send as request body content
1100 * @param send_content_length the length of the send_content data, or 0 if
1101 * send_content is null
1103 * @return zero if success, nonzero otherwise
1105 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1106 const char *method, const char *url,
1107 const char *headers,
1108 const unsigned char *send_content,
1109 int send_content_length)
1111 RTSPState *rt = s->priv_data;
1112 char buf[4096], *out_buf;
1113 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1115 /* Add in RTSP headers */
1118 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1120 av_strlcat(buf, headers, sizeof(buf));
1121 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1122 if (rt->session_id[0] != '\0' && (!headers ||
1123 !strstr(headers, "\nIf-Match:"))) {
1124 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1127 char *str = ff_http_auth_create_response(&rt->auth_state,
1128 rt->auth, url, method);
1130 av_strlcat(buf, str, sizeof(buf));
1133 if (send_content_length > 0 && send_content)
1134 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1135 av_strlcat(buf, "\r\n", sizeof(buf));
1137 /* base64 encode rtsp if tunneling */
1138 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1139 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1140 out_buf = base64buf;
1143 av_dlog(s, "Sending:\n%s--\n", buf);
1145 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1146 if (send_content_length > 0 && send_content) {
1147 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1148 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1149 "with content data not supported\n");
1150 return AVERROR_PATCHWELCOME;
1152 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1154 rt->last_cmd_time = av_gettime();
1159 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1160 const char *url, const char *headers)
1162 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1165 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1166 const char *headers, RTSPMessageHeader *reply,
1167 unsigned char **content_ptr)
1169 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1170 content_ptr, NULL, 0);
1173 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1174 const char *method, const char *url,
1176 RTSPMessageHeader *reply,
1177 unsigned char **content_ptr,
1178 const unsigned char *send_content,
1179 int send_content_length)
1181 RTSPState *rt = s->priv_data;
1182 HTTPAuthType cur_auth_type;
1183 int ret, attempts = 0;
1186 cur_auth_type = rt->auth_state.auth_type;
1187 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1189 send_content_length)))
1192 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1196 if (reply->status_code == 401 &&
1197 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1198 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1201 if (reply->status_code > 400){
1202 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1206 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1212 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1213 int lower_transport, const char *real_challenge)
1215 RTSPState *rt = s->priv_data;
1216 int rtx = 0, j, i, err, interleave = 0, port_off;
1217 RTSPStream *rtsp_st;
1218 RTSPMessageHeader reply1, *reply = &reply1;
1220 const char *trans_pref;
1222 if (rt->transport == RTSP_TRANSPORT_RDT)
1223 trans_pref = "x-pn-tng";
1224 else if (rt->transport == RTSP_TRANSPORT_RAW)
1225 trans_pref = "RAW/RAW";
1227 trans_pref = "RTP/AVP";
1229 /* default timeout: 1 minute */
1232 /* for each stream, make the setup request */
1233 /* XXX: we assume the same server is used for the control of each
1236 /* Choose a random starting offset within the first half of the
1237 * port range, to allow for a number of ports to try even if the offset
1238 * happens to be at the end of the random range. */
1239 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1240 /* even random offset */
1241 port_off -= port_off & 0x01;
1243 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1244 char transport[2048];
1247 * WMS serves all UDP data over a single connection, the RTX, which
1248 * isn't necessarily the first in the SDP but has to be the first
1249 * to be set up, else the second/third SETUP will fail with a 461.
1251 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1252 rt->server_type == RTSP_SERVER_WMS) {
1255 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1256 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1258 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1262 if (rtx == rt->nb_rtsp_streams)
1263 return -1; /* no RTX found */
1264 rtsp_st = rt->rtsp_streams[rtx];
1266 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1268 rtsp_st = rt->rtsp_streams[i];
1271 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1274 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1275 port = reply->transports[0].client_port_min;
1279 /* first try in specified port range */
1280 while (j <= rt->rtp_port_max) {
1281 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1282 "?localport=%d", j);
1283 /* we will use two ports per rtp stream (rtp and rtcp) */
1285 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1286 &s->interrupt_callback, NULL))
1290 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1295 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1297 snprintf(transport, sizeof(transport) - 1,
1298 "%s/UDP;", trans_pref);
1299 if (rt->server_type != RTSP_SERVER_REAL)
1300 av_strlcat(transport, "unicast;", sizeof(transport));
1301 av_strlcatf(transport, sizeof(transport),
1302 "client_port=%d", port);
1303 if (rt->transport == RTSP_TRANSPORT_RTP &&
1304 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1305 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1309 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1310 /* For WMS streams, the application streams are only used for
1311 * UDP. When trying to set it up for TCP streams, the server
1312 * will return an error. Therefore, we skip those streams. */
1313 if (rt->server_type == RTSP_SERVER_WMS &&
1314 (rtsp_st->stream_index < 0 ||
1315 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1318 snprintf(transport, sizeof(transport) - 1,
1319 "%s/TCP;", trans_pref);
1320 if (rt->transport != RTSP_TRANSPORT_RDT)
1321 av_strlcat(transport, "unicast;", sizeof(transport));
1322 av_strlcatf(transport, sizeof(transport),
1323 "interleaved=%d-%d",
1324 interleave, interleave + 1);
1328 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1329 snprintf(transport, sizeof(transport) - 1,
1330 "%s/UDP;multicast", trans_pref);
1333 av_strlcat(transport, ";mode=record", sizeof(transport));
1334 } else if (rt->server_type == RTSP_SERVER_REAL ||
1335 rt->server_type == RTSP_SERVER_WMS)
1336 av_strlcat(transport, ";mode=play", sizeof(transport));
1337 snprintf(cmd, sizeof(cmd),
1338 "Transport: %s\r\n",
1340 if (rt->accept_dynamic_rate)
1341 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1342 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1343 char real_res[41], real_csum[9];
1344 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1346 av_strlcatf(cmd, sizeof(cmd),
1348 "RealChallenge2: %s, sd=%s\r\n",
1349 rt->session_id, real_res, real_csum);
1351 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1352 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1355 } else if (reply->status_code != RTSP_STATUS_OK ||
1356 reply->nb_transports != 1) {
1357 err = AVERROR_INVALIDDATA;
1361 /* XXX: same protocol for all streams is required */
1363 if (reply->transports[0].lower_transport != rt->lower_transport ||
1364 reply->transports[0].transport != rt->transport) {
1365 err = AVERROR_INVALIDDATA;
1369 rt->lower_transport = reply->transports[0].lower_transport;
1370 rt->transport = reply->transports[0].transport;
1373 /* Fail if the server responded with another lower transport mode
1374 * than what we requested. */
1375 if (reply->transports[0].lower_transport != lower_transport) {
1376 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1377 err = AVERROR_INVALIDDATA;
1381 switch(reply->transports[0].lower_transport) {
1382 case RTSP_LOWER_TRANSPORT_TCP:
1383 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1384 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1387 case RTSP_LOWER_TRANSPORT_UDP: {
1388 char url[1024], options[30] = "";
1390 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1391 av_strlcpy(options, "?connect=1", sizeof(options));
1392 /* Use source address if specified */
1393 if (reply->transports[0].source[0]) {
1394 ff_url_join(url, sizeof(url), "rtp", NULL,
1395 reply->transports[0].source,
1396 reply->transports[0].server_port_min, "%s", options);
1398 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1399 reply->transports[0].server_port_min, "%s", options);
1401 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1402 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1403 err = AVERROR_INVALIDDATA;
1406 /* Try to initialize the connection state in a
1407 * potential NAT router by sending dummy packets.
1408 * RTP/RTCP dummy packets are used for RDT, too.
1410 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1412 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1415 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1416 char url[1024], namebuf[50], optbuf[20] = "";
1417 struct sockaddr_storage addr;
1420 if (reply->transports[0].destination.ss_family) {
1421 addr = reply->transports[0].destination;
1422 port = reply->transports[0].port_min;
1423 ttl = reply->transports[0].ttl;
1425 addr = rtsp_st->sdp_ip;
1426 port = rtsp_st->sdp_port;
1427 ttl = rtsp_st->sdp_ttl;
1430 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1431 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1432 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1433 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1434 port, "%s", optbuf);
1435 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1436 &s->interrupt_callback, NULL) < 0) {
1437 err = AVERROR_INVALIDDATA;
1444 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1448 if (rt->nb_rtsp_streams && reply->timeout > 0)
1449 rt->timeout = reply->timeout;
1451 if (rt->server_type == RTSP_SERVER_REAL)
1452 rt->need_subscription = 1;
1457 ff_rtsp_undo_setup(s);
1461 void ff_rtsp_close_connections(AVFormatContext *s)
1463 RTSPState *rt = s->priv_data;
1464 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1465 ffurl_close(rt->rtsp_hd);
1466 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1469 int ff_rtsp_connect(AVFormatContext *s)
1471 RTSPState *rt = s->priv_data;
1472 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1473 int port, err, tcp_fd;
1474 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1475 int lower_transport_mask = 0;
1476 char real_challenge[64] = "";
1477 struct sockaddr_storage peer;
1478 socklen_t peer_len = sizeof(peer);
1480 if (rt->rtp_port_max < rt->rtp_port_min) {
1481 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1482 "than min port %d\n", rt->rtp_port_max,
1484 return AVERROR(EINVAL);
1487 if (!ff_network_init())
1488 return AVERROR(EIO);
1490 if (s->max_delay < 0) /* Not set by the caller */
1491 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1493 rt->control_transport = RTSP_MODE_PLAIN;
1494 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1495 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1496 rt->control_transport = RTSP_MODE_TUNNEL;
1498 /* Only pass through valid flags from here */
1499 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1502 lower_transport_mask = rt->lower_transport_mask;
1503 /* extract hostname and port */
1504 av_url_split(NULL, 0, auth, sizeof(auth),
1505 host, sizeof(host), &port, path, sizeof(path), s->filename);
1507 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1510 port = RTSP_DEFAULT_PORT;
1512 if (!lower_transport_mask)
1513 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1516 /* Only UDP or TCP - UDP multicast isn't supported. */
1517 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1518 (1 << RTSP_LOWER_TRANSPORT_TCP);
1519 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1520 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1521 "only UDP and TCP are supported for output.\n");
1522 err = AVERROR(EINVAL);
1527 /* Construct the URI used in request; this is similar to s->filename,
1528 * but with authentication credentials removed and RTSP specific options
1530 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1531 host, port, "%s", path);
1533 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1534 /* set up initial handshake for tunneling */
1535 char httpname[1024];
1536 char sessioncookie[17];
1539 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1540 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1541 av_get_random_seed(), av_get_random_seed());
1544 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1545 &s->interrupt_callback) < 0) {
1550 /* generate GET headers */
1551 snprintf(headers, sizeof(headers),
1552 "x-sessioncookie: %s\r\n"
1553 "Accept: application/x-rtsp-tunnelled\r\n"
1554 "Pragma: no-cache\r\n"
1555 "Cache-Control: no-cache\r\n",
1557 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1559 /* complete the connection */
1560 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1566 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1567 &s->interrupt_callback) < 0 ) {
1572 /* generate POST headers */
1573 snprintf(headers, sizeof(headers),
1574 "x-sessioncookie: %s\r\n"
1575 "Content-Type: application/x-rtsp-tunnelled\r\n"
1576 "Pragma: no-cache\r\n"
1577 "Cache-Control: no-cache\r\n"
1578 "Content-Length: 32767\r\n"
1579 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1581 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1582 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1584 /* Initialize the authentication state for the POST session. The HTTP
1585 * protocol implementation doesn't properly handle multi-pass
1586 * authentication for POST requests, since it would require one of
1588 * - implementing Expect: 100-continue, which many HTTP servers
1589 * don't support anyway, even less the RTSP servers that do HTTP
1591 * - sending the whole POST data until getting a 401 reply specifying
1592 * what authentication method to use, then resending all that data
1593 * - waiting for potential 401 replies directly after sending the
1594 * POST header (waiting for some unspecified time)
1595 * Therefore, we copy the full auth state, which works for both basic
1596 * and digest. (For digest, we would have to synchronize the nonce
1597 * count variable between the two sessions, if we'd do more requests
1598 * with the original session, though.)
1600 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1602 /* complete the connection */
1603 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1608 /* open the tcp connection */
1609 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1610 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1611 &s->interrupt_callback, NULL) < 0) {
1615 rt->rtsp_hd_out = rt->rtsp_hd;
1619 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1620 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1621 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1622 NULL, 0, NI_NUMERICHOST);
1625 /* request options supported by the server; this also detects server
1627 for (rt->server_type = RTSP_SERVER_RTP;;) {
1629 if (rt->server_type == RTSP_SERVER_REAL)
1632 * The following entries are required for proper
1633 * streaming from a Realmedia server. They are
1634 * interdependent in some way although we currently
1635 * don't quite understand how. Values were copied
1636 * from mplayer SVN r23589.
1637 * ClientChallenge is a 16-byte ID in hex
1638 * CompanyID is a 16-byte ID in base64
1640 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1641 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1642 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1643 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1645 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1646 if (reply->status_code != RTSP_STATUS_OK) {
1647 err = AVERROR_INVALIDDATA;
1651 /* detect server type if not standard-compliant RTP */
1652 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1653 rt->server_type = RTSP_SERVER_REAL;
1655 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1656 rt->server_type = RTSP_SERVER_WMS;
1657 } else if (rt->server_type == RTSP_SERVER_REAL)
1658 strcpy(real_challenge, reply->real_challenge);
1662 if (s->iformat && CONFIG_RTSP_DEMUXER)
1663 err = ff_rtsp_setup_input_streams(s, reply);
1664 else if (CONFIG_RTSP_MUXER)
1665 err = ff_rtsp_setup_output_streams(s, host);
1670 int lower_transport = ff_log2_tab[lower_transport_mask &
1671 ~(lower_transport_mask - 1)];
1673 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1674 rt->server_type == RTSP_SERVER_REAL ?
1675 real_challenge : NULL);
1678 lower_transport_mask &= ~(1 << lower_transport);
1679 if (lower_transport_mask == 0 && err == 1) {
1680 err = AVERROR(EPROTONOSUPPORT);
1685 rt->lower_transport_mask = lower_transport_mask;
1686 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1687 rt->state = RTSP_STATE_IDLE;
1688 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1691 ff_rtsp_close_streams(s);
1692 ff_rtsp_close_connections(s);
1693 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1694 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1695 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1703 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1706 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1707 uint8_t *buf, int buf_size, int64_t wait_end)
1709 RTSPState *rt = s->priv_data;
1710 RTSPStream *rtsp_st;
1711 int n, i, ret, tcp_fd, timeout_cnt = 0;
1713 struct pollfd *p = rt->p;
1714 int *fds = NULL, fdsnum, fdsidx;
1717 if (ff_check_interrupt(&s->interrupt_callback))
1718 return AVERROR_EXIT;
1719 if (wait_end && wait_end - av_gettime() < 0)
1720 return AVERROR(EAGAIN);
1723 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1724 p[max_p].fd = tcp_fd;
1725 p[max_p++].events = POLLIN;
1729 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1730 rtsp_st = rt->rtsp_streams[i];
1731 if (rtsp_st->rtp_handle) {
1732 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1734 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1738 av_log(s, AV_LOG_ERROR,
1739 "Number of fds %d not supported\n", fdsnum);
1740 return AVERROR_INVALIDDATA;
1742 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1743 p[max_p].fd = fds[fdsidx];
1744 p[max_p++].events = POLLIN;
1749 n = poll(p, max_p, POLL_TIMEOUT_MS);
1751 int j = 1 - (tcp_fd == -1);
1753 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1754 rtsp_st = rt->rtsp_streams[i];
1755 if (rtsp_st->rtp_handle) {
1756 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1757 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1759 *prtsp_st = rtsp_st;
1766 #if CONFIG_RTSP_DEMUXER
1767 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1768 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1769 if (rt->state == RTSP_STATE_STREAMING) {
1770 if (!ff_rtsp_parse_streaming_commands(s))
1773 av_log(s, AV_LOG_WARNING,
1774 "Unable to answer to TEARDOWN\n");
1778 RTSPMessageHeader reply;
1779 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1782 /* XXX: parse message */
1783 if (rt->state != RTSP_STATE_STREAMING)
1788 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1789 return AVERROR(ETIMEDOUT);
1790 } else if (n < 0 && errno != EINTR)
1791 return AVERROR(errno);
1795 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1796 const uint8_t *buf, int len)
1798 RTSPState *rt = s->priv_data;
1802 if (rt->nb_rtsp_streams == 1) {
1803 *rtsp_st = rt->rtsp_streams[0];
1806 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1807 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1809 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1810 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1813 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1814 *rtsp_st = rt->rtsp_streams[i];
1821 av_log(s, AV_LOG_WARNING,
1822 "Unable to pick stream for packet - SSRC not known for "
1824 return AVERROR(EAGAIN);
1827 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1828 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1829 *rtsp_st = rt->rtsp_streams[i];
1835 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1836 return AVERROR(EAGAIN);
1839 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1841 RTSPState *rt = s->priv_data;
1843 RTSPStream *rtsp_st, *first_queue_st = NULL;
1844 int64_t wait_end = 0;
1846 if (rt->nb_byes == rt->nb_rtsp_streams)
1849 /* get next frames from the same RTP packet */
1850 if (rt->cur_transport_priv) {
1851 if (rt->transport == RTSP_TRANSPORT_RDT) {
1852 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1853 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1854 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1855 } else if (rt->ts && CONFIG_RTPDEC) {
1856 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1858 rt->recvbuf_pos += ret;
1859 ret = rt->recvbuf_pos < rt->recvbuf_len;
1864 rt->cur_transport_priv = NULL;
1866 } else if (ret == 1) {
1869 rt->cur_transport_priv = NULL;
1873 if (rt->transport == RTSP_TRANSPORT_RTP) {
1875 int64_t first_queue_time = 0;
1876 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1877 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1881 queue_time = ff_rtp_queued_packet_time(rtpctx);
1882 if (queue_time && (queue_time - first_queue_time < 0 ||
1883 !first_queue_time)) {
1884 first_queue_time = queue_time;
1885 first_queue_st = rt->rtsp_streams[i];
1888 if (first_queue_time) {
1889 wait_end = first_queue_time + s->max_delay;
1892 first_queue_st = NULL;
1896 /* read next RTP packet */
1898 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1900 return AVERROR(ENOMEM);
1903 switch(rt->lower_transport) {
1905 #if CONFIG_RTSP_DEMUXER
1906 case RTSP_LOWER_TRANSPORT_TCP:
1907 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1910 case RTSP_LOWER_TRANSPORT_UDP:
1911 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1912 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1913 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1914 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1916 case RTSP_LOWER_TRANSPORT_CUSTOM:
1917 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1918 wait_end && wait_end < av_gettime())
1919 len = AVERROR(EAGAIN);
1921 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1922 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1923 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1924 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1927 if (len == AVERROR(EAGAIN) && first_queue_st &&
1928 rt->transport == RTSP_TRANSPORT_RTP) {
1929 rtsp_st = first_queue_st;
1930 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1937 if (rt->transport == RTSP_TRANSPORT_RDT) {
1938 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1939 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1940 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1941 if (rtsp_st->feedback) {
1942 AVIOContext *pb = NULL;
1943 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1945 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1948 /* Either bad packet, or a RTCP packet. Check if the
1949 * first_rtcp_ntp_time field was initialized. */
1950 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1951 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1952 /* first_rtcp_ntp_time has been initialized for this stream,
1953 * copy the same value to all other uninitialized streams,
1954 * in order to map their timestamp origin to the same ntp time
1957 AVStream *st = NULL;
1958 if (rtsp_st->stream_index >= 0)
1959 st = s->streams[rtsp_st->stream_index];
1960 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1961 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1962 AVStream *st2 = NULL;
1963 if (rt->rtsp_streams[i]->stream_index >= 0)
1964 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1965 if (rtpctx2 && st && st2 &&
1966 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1967 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1968 rtpctx2->rtcp_ts_offset = av_rescale_q(
1969 rtpctx->rtcp_ts_offset, st->time_base,
1974 if (ret == -RTCP_BYE) {
1977 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1978 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1980 if (rt->nb_byes == rt->nb_rtsp_streams)
1984 } else if (rt->ts && CONFIG_RTPDEC) {
1985 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1988 rt->recvbuf_len = len;
1989 rt->recvbuf_pos = ret;
1990 rt->cur_transport_priv = rt->ts;
1997 return AVERROR_INVALIDDATA;
2003 /* more packets may follow, so we save the RTP context */
2004 rt->cur_transport_priv = rtsp_st->transport_priv;
2008 #endif /* CONFIG_RTPDEC */
2010 #if CONFIG_SDP_DEMUXER
2011 static int sdp_probe(AVProbeData *p1)
2013 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2015 /* we look for a line beginning "c=IN IP" */
2016 while (p < p_end && *p != '\0') {
2017 if (p + sizeof("c=IN IP") - 1 < p_end &&
2018 av_strstart(p, "c=IN IP", NULL))
2019 return AVPROBE_SCORE_MAX / 2;
2021 while (p < p_end - 1 && *p != '\n') p++;
2030 static int sdp_read_header(AVFormatContext *s)
2032 RTSPState *rt = s->priv_data;
2033 RTSPStream *rtsp_st;
2038 if (!ff_network_init())
2039 return AVERROR(EIO);
2041 if (s->max_delay < 0) /* Not set by the caller */
2042 s->max_delay = DEFAULT_REORDERING_DELAY;
2043 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2044 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2046 /* read the whole sdp file */
2047 /* XXX: better loading */
2048 content = av_malloc(SDP_MAX_SIZE);
2049 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2052 return AVERROR_INVALIDDATA;
2054 content[size] ='\0';
2056 err = ff_sdp_parse(s, content);
2060 /* open each RTP stream */
2061 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2063 rtsp_st = rt->rtsp_streams[i];
2065 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2066 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2067 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2068 ff_url_join(url, sizeof(url), "rtp", NULL,
2069 namebuf, rtsp_st->sdp_port,
2070 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2072 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2073 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2074 &s->interrupt_callback, NULL) < 0) {
2075 err = AVERROR_INVALIDDATA;
2079 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2084 ff_rtsp_close_streams(s);
2089 static int sdp_read_close(AVFormatContext *s)
2091 ff_rtsp_close_streams(s);
2096 static const AVClass sdp_demuxer_class = {
2097 .class_name = "SDP demuxer",
2098 .item_name = av_default_item_name,
2099 .option = sdp_options,
2100 .version = LIBAVUTIL_VERSION_INT,
2103 AVInputFormat ff_sdp_demuxer = {
2105 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2106 .priv_data_size = sizeof(RTSPState),
2107 .read_probe = sdp_probe,
2108 .read_header = sdp_read_header,
2109 .read_packet = ff_rtsp_fetch_packet,
2110 .read_close = sdp_read_close,
2111 .priv_class = &sdp_demuxer_class,
2113 #endif /* CONFIG_SDP_DEMUXER */
2115 #if CONFIG_RTP_DEMUXER
2116 static int rtp_probe(AVProbeData *p)
2118 if (av_strstart(p->filename, "rtp:", NULL))
2119 return AVPROBE_SCORE_MAX;
2123 static int rtp_read_header(AVFormatContext *s)
2125 uint8_t recvbuf[1500];
2126 char host[500], sdp[500];
2128 URLContext* in = NULL;
2130 AVCodecContext codec = { 0 };
2131 struct sockaddr_storage addr;
2133 socklen_t addrlen = sizeof(addr);
2134 RTSPState *rt = s->priv_data;
2136 if (!ff_network_init())
2137 return AVERROR(EIO);
2139 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2140 &s->interrupt_callback, NULL);
2145 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2146 if (ret == AVERROR(EAGAIN))
2151 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2155 if ((recvbuf[0] & 0xc0) != 0x80) {
2156 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2161 if (RTP_PT_IS_RTCP(recvbuf[1]))
2164 payload_type = recvbuf[1] & 0x7f;
2167 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2171 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2172 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2173 "without an SDP file describing it\n",
2177 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2178 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2179 "properly you need an SDP file "
2183 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2184 NULL, 0, s->filename);
2186 snprintf(sdp, sizeof(sdp),
2187 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2188 addr.ss_family == AF_INET ? 4 : 6, host,
2189 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2190 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2191 port, payload_type);
2192 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2194 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2197 /* sdp_read_header initializes this again */
2200 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2202 ret = sdp_read_header(s);
2213 static const AVClass rtp_demuxer_class = {
2214 .class_name = "RTP demuxer",
2215 .item_name = av_default_item_name,
2216 .option = rtp_options,
2217 .version = LIBAVUTIL_VERSION_INT,
2220 AVInputFormat ff_rtp_demuxer = {
2222 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2223 .priv_data_size = sizeof(RTSPState),
2224 .read_probe = rtp_probe,
2225 .read_header = rtp_read_header,
2226 .read_packet = ff_rtsp_fetch_packet,
2227 .read_close = sdp_read_close,
2228 .flags = AVFMT_NOFILE,
2229 .priv_class = &rtp_demuxer_class,
2231 #endif /* CONFIG_RTP_DEMUXER */