3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
70 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
72 #define RTSP_MEDIATYPE_OPTS(name, longname) \
73 { 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" }, \
74 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
76 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
78 #define RTSP_REORDERING_OPTS() \
79 { "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 }
81 const AVOption ff_rtsp_options[] = {
82 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
83 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
84 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
85 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
88 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "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 },
94 RTSP_REORDERING_OPTS(),
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP 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;
383 /* XXX: handle list of formats */
384 get_word(buf1, sizeof(buf1), &p); /* format list */
385 rtsp_st->sdp_payload_type = atoi(buf1);
387 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
388 /* no corresponding stream */
389 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
390 rt->ts = ff_mpegts_parse_open(s);
391 } else if (rt->server_type == RTSP_SERVER_WMS &&
392 codec_type == AVMEDIA_TYPE_DATA) {
393 /* RTX stream, a stream that carries all the other actual
394 * audio/video streams. Don't expose this to the callers. */
396 st = avformat_new_stream(s, NULL);
399 st->id = rt->nb_rtsp_streams - 1;
400 rtsp_st->stream_index = st->index;
401 st->codec->codec_type = codec_type;
402 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
403 RTPDynamicProtocolHandler *handler;
404 /* if standard payload type, we can find the codec right now */
405 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
406 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
407 st->codec->sample_rate > 0)
408 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
409 /* Even static payload types may need a custom depacketizer */
410 handler = ff_rtp_handler_find_by_id(
411 rtsp_st->sdp_payload_type, st->codec->codec_type);
412 init_rtp_handler(handler, rtsp_st, st->codec);
413 if (handler && handler->init)
414 handler->init(s, st->index,
415 rtsp_st->dynamic_protocol_context);
418 /* put a default control url */
419 av_strlcpy(rtsp_st->control_url, rt->control_uri,
420 sizeof(rtsp_st->control_url));
423 if (av_strstart(p, "control:", &p)) {
424 if (s->nb_streams == 0) {
425 if (!strncmp(p, "rtsp://", 7))
426 av_strlcpy(rt->control_uri, p,
427 sizeof(rt->control_uri));
430 /* get the control url */
431 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
433 /* XXX: may need to add full url resolution */
434 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
436 if (proto[0] == '\0') {
437 /* relative control URL */
438 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
439 av_strlcat(rtsp_st->control_url, "/",
440 sizeof(rtsp_st->control_url));
441 av_strlcat(rtsp_st->control_url, p,
442 sizeof(rtsp_st->control_url));
444 av_strlcpy(rtsp_st->control_url, p,
445 sizeof(rtsp_st->control_url));
447 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
448 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
449 get_word(buf1, sizeof(buf1), &p);
450 payload_type = atoi(buf1);
451 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
452 if (rtsp_st->stream_index >= 0) {
453 st = s->streams[rtsp_st->stream_index];
454 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
456 } else if (av_strstart(p, "fmtp:", &p) ||
457 av_strstart(p, "framesize:", &p)) {
458 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
459 // let dynamic protocol handlers have a stab at the line.
460 get_word(buf1, sizeof(buf1), &p);
461 payload_type = atoi(buf1);
462 for (i = 0; i < rt->nb_rtsp_streams; i++) {
463 rtsp_st = rt->rtsp_streams[i];
464 if (rtsp_st->sdp_payload_type == payload_type &&
465 rtsp_st->dynamic_handler &&
466 rtsp_st->dynamic_handler->parse_sdp_a_line)
467 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
468 rtsp_st->dynamic_protocol_context, buf);
470 } else if (av_strstart(p, "range:", &p)) {
473 // this is so that seeking on a streamed file can work.
474 rtsp_parse_range_npt(p, &start, &end);
475 s->start_time = start;
476 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
477 s->duration = (end == AV_NOPTS_VALUE) ?
478 AV_NOPTS_VALUE : end - start;
479 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
481 rt->transport = RTSP_TRANSPORT_RDT;
482 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
484 st = s->streams[s->nb_streams - 1];
485 st->codec->sample_rate = atoi(p);
487 if (rt->server_type == RTSP_SERVER_WMS)
488 ff_wms_parse_sdp_a_line(s, p);
489 if (s->nb_streams > 0) {
490 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
492 if (rt->server_type == RTSP_SERVER_REAL)
493 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
495 if (rtsp_st->dynamic_handler &&
496 rtsp_st->dynamic_handler->parse_sdp_a_line)
497 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
498 rtsp_st->stream_index,
499 rtsp_st->dynamic_protocol_context, buf);
506 int ff_sdp_parse(AVFormatContext *s, const char *content)
508 RTSPState *rt = s->priv_data;
511 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
512 * contain long SDP lines containing complete ASF Headers (several
513 * kB) or arrays of MDPR (RM stream descriptor) headers plus
514 * "rulebooks" describing their properties. Therefore, the SDP line
517 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
518 * in rtpdec_xiph.c. */
520 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
524 p += strspn(p, SPACE_CHARS);
532 /* get the content */
534 while (*p != '\n' && *p != '\r' && *p != '\0') {
535 if ((q - buf) < sizeof(buf) - 1)
540 sdp_parse_line(s, s1, letter, buf);
542 while (*p != '\n' && *p != '\0')
547 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
548 if (!rt->p) return AVERROR(ENOMEM);
551 #endif /* CONFIG_RTPDEC */
553 void ff_rtsp_undo_setup(AVFormatContext *s)
555 RTSPState *rt = s->priv_data;
558 for (i = 0; i < rt->nb_rtsp_streams; i++) {
559 RTSPStream *rtsp_st = rt->rtsp_streams[i];
562 if (rtsp_st->transport_priv) {
564 AVFormatContext *rtpctx = rtsp_st->transport_priv;
565 av_write_trailer(rtpctx);
566 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
568 avio_close_dyn_buf(rtpctx->pb, &ptr);
571 avio_close(rtpctx->pb);
573 avformat_free_context(rtpctx);
574 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
575 ff_rdt_parse_close(rtsp_st->transport_priv);
576 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
577 ff_rtp_parse_close(rtsp_st->transport_priv);
579 rtsp_st->transport_priv = NULL;
580 if (rtsp_st->rtp_handle)
581 ffurl_close(rtsp_st->rtp_handle);
582 rtsp_st->rtp_handle = NULL;
586 /* close and free RTSP streams */
587 void ff_rtsp_close_streams(AVFormatContext *s)
589 RTSPState *rt = s->priv_data;
593 ff_rtsp_undo_setup(s);
594 for (i = 0; i < rt->nb_rtsp_streams; i++) {
595 rtsp_st = rt->rtsp_streams[i];
597 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
598 rtsp_st->dynamic_handler->free(
599 rtsp_st->dynamic_protocol_context);
603 av_free(rt->rtsp_streams);
605 avformat_close_input(&rt->asf_ctx);
607 if (rt->ts && CONFIG_RTPDEC)
608 ff_mpegts_parse_close(rt->ts);
610 av_free(rt->recvbuf);
613 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
615 RTSPState *rt = s->priv_data;
617 int reordering_queue_size = rt->reordering_queue_size;
618 if (reordering_queue_size < 0) {
619 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
620 reordering_queue_size = 0;
622 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
625 /* open the RTP context */
626 if (rtsp_st->stream_index >= 0)
627 st = s->streams[rtsp_st->stream_index];
629 s->ctx_flags |= AVFMTCTX_NOHEADER;
631 if (s->oformat && CONFIG_RTSP_MUXER) {
632 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
634 RTSP_TCP_MAX_PACKET_SIZE);
635 /* Ownership of rtp_handle is passed to the rtp mux context */
636 rtsp_st->rtp_handle = NULL;
639 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
640 return 0; // Don't need to open any parser here
641 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
642 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
643 rtsp_st->dynamic_protocol_context,
644 rtsp_st->dynamic_handler);
645 else if (CONFIG_RTPDEC)
646 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
647 rtsp_st->sdp_payload_type,
648 reordering_queue_size);
650 if (!rtsp_st->transport_priv) {
651 return AVERROR(ENOMEM);
652 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
653 if (rtsp_st->dynamic_handler) {
654 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
655 rtsp_st->dynamic_protocol_context,
656 rtsp_st->dynamic_handler);
663 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
664 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
671 q += strspn(q, SPACE_CHARS);
672 v = strtol(q, &p, 10);
676 v = strtol(p, &p, 10);
685 /* XXX: only one transport specification is parsed */
686 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
688 char transport_protocol[16];
690 char lower_transport[16];
692 RTSPTransportField *th;
695 reply->nb_transports = 0;
698 p += strspn(p, SPACE_CHARS);
702 th = &reply->transports[reply->nb_transports];
704 get_word_sep(transport_protocol, sizeof(transport_protocol),
706 if (!av_strcasecmp (transport_protocol, "rtp")) {
707 get_word_sep(profile, sizeof(profile), "/;,", &p);
708 lower_transport[0] = '\0';
709 /* rtp/avp/<protocol> */
711 get_word_sep(lower_transport, sizeof(lower_transport),
714 th->transport = RTSP_TRANSPORT_RTP;
715 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
716 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
717 /* x-pn-tng/<protocol> */
718 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
720 th->transport = RTSP_TRANSPORT_RDT;
721 } else if (!av_strcasecmp(transport_protocol, "raw")) {
722 get_word_sep(profile, sizeof(profile), "/;,", &p);
723 lower_transport[0] = '\0';
724 /* raw/raw/<protocol> */
726 get_word_sep(lower_transport, sizeof(lower_transport),
729 th->transport = RTSP_TRANSPORT_RAW;
731 if (!av_strcasecmp(lower_transport, "TCP"))
732 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
734 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
738 /* get each parameter */
739 while (*p != '\0' && *p != ',') {
740 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
741 if (!strcmp(parameter, "port")) {
744 rtsp_parse_range(&th->port_min, &th->port_max, &p);
746 } else if (!strcmp(parameter, "client_port")) {
749 rtsp_parse_range(&th->client_port_min,
750 &th->client_port_max, &p);
752 } else if (!strcmp(parameter, "server_port")) {
755 rtsp_parse_range(&th->server_port_min,
756 &th->server_port_max, &p);
758 } else if (!strcmp(parameter, "interleaved")) {
761 rtsp_parse_range(&th->interleaved_min,
762 &th->interleaved_max, &p);
764 } else if (!strcmp(parameter, "multicast")) {
765 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
766 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
767 } else if (!strcmp(parameter, "ttl")) {
771 th->ttl = strtol(p, &end, 10);
774 } else if (!strcmp(parameter, "destination")) {
777 get_word_sep(buf, sizeof(buf), ";,", &p);
778 get_sockaddr(buf, &th->destination);
780 } else if (!strcmp(parameter, "source")) {
783 get_word_sep(buf, sizeof(buf), ";,", &p);
784 av_strlcpy(th->source, buf, sizeof(th->source));
786 } else if (!strcmp(parameter, "mode")) {
789 get_word_sep(buf, sizeof(buf), ";, ", &p);
790 if (!strcmp(buf, "record") ||
791 !strcmp(buf, "receive"))
796 while (*p != ';' && *p != '\0' && *p != ',')
804 reply->nb_transports++;
808 static void handle_rtp_info(RTSPState *rt, const char *url,
809 uint32_t seq, uint32_t rtptime)
812 if (!rtptime || !url[0])
814 if (rt->transport != RTSP_TRANSPORT_RTP)
816 for (i = 0; i < rt->nb_rtsp_streams; i++) {
817 RTSPStream *rtsp_st = rt->rtsp_streams[i];
818 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
821 if (!strcmp(rtsp_st->control_url, url)) {
822 rtpctx->base_timestamp = rtptime;
828 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
831 char key[20], value[1024], url[1024] = "";
832 uint32_t seq = 0, rtptime = 0;
835 p += strspn(p, SPACE_CHARS);
838 get_word_sep(key, sizeof(key), "=", &p);
842 get_word_sep(value, sizeof(value), ";, ", &p);
844 if (!strcmp(key, "url"))
845 av_strlcpy(url, value, sizeof(url));
846 else if (!strcmp(key, "seq"))
847 seq = strtoul(value, NULL, 10);
848 else if (!strcmp(key, "rtptime"))
849 rtptime = strtoul(value, NULL, 10);
851 handle_rtp_info(rt, url, seq, rtptime);
860 handle_rtp_info(rt, url, seq, rtptime);
863 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
864 RTSPState *rt, const char *method)
868 /* NOTE: we do case independent match for broken servers */
870 if (av_stristart(p, "Session:", &p)) {
872 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
873 if (av_stristart(p, ";timeout=", &p) &&
874 (t = strtol(p, NULL, 10)) > 0) {
877 } else if (av_stristart(p, "Content-Length:", &p)) {
878 reply->content_length = strtol(p, NULL, 10);
879 } else if (av_stristart(p, "Transport:", &p)) {
880 rtsp_parse_transport(reply, p);
881 } else if (av_stristart(p, "CSeq:", &p)) {
882 reply->seq = strtol(p, NULL, 10);
883 } else if (av_stristart(p, "Range:", &p)) {
884 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
885 } else if (av_stristart(p, "RealChallenge1:", &p)) {
886 p += strspn(p, SPACE_CHARS);
887 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
888 } else if (av_stristart(p, "Server:", &p)) {
889 p += strspn(p, SPACE_CHARS);
890 av_strlcpy(reply->server, p, sizeof(reply->server));
891 } else if (av_stristart(p, "Notice:", &p) ||
892 av_stristart(p, "X-Notice:", &p)) {
893 reply->notice = strtol(p, NULL, 10);
894 } else if (av_stristart(p, "Location:", &p)) {
895 p += strspn(p, SPACE_CHARS);
896 av_strlcpy(reply->location, p , sizeof(reply->location));
897 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
898 p += strspn(p, SPACE_CHARS);
899 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
900 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
901 p += strspn(p, SPACE_CHARS);
902 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
903 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
904 p += strspn(p, SPACE_CHARS);
905 if (method && !strcmp(method, "DESCRIBE"))
906 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
907 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
908 p += strspn(p, SPACE_CHARS);
909 if (method && !strcmp(method, "PLAY"))
910 rtsp_parse_rtp_info(rt, p);
911 } else if (av_stristart(p, "Public:", &p) && rt) {
912 if (strstr(p, "GET_PARAMETER") &&
913 method && !strcmp(method, "OPTIONS"))
914 rt->get_parameter_supported = 1;
915 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
916 p += strspn(p, SPACE_CHARS);
917 rt->accept_dynamic_rate = atoi(p);
918 } else if (av_stristart(p, "Content-Type:", &p)) {
919 p += strspn(p, SPACE_CHARS);
920 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
924 /* skip a RTP/TCP interleaved packet */
925 void ff_rtsp_skip_packet(AVFormatContext *s)
927 RTSPState *rt = s->priv_data;
931 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
934 len = AV_RB16(buf + 1);
936 av_dlog(s, "skipping RTP packet len=%d\n", len);
941 if (len1 > sizeof(buf))
943 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
950 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
951 unsigned char **content_ptr,
952 int return_on_interleaved_data, const char *method)
954 RTSPState *rt = s->priv_data;
955 char buf[4096], buf1[1024], *q;
958 int ret, content_length, line_count = 0, request = 0;
959 unsigned char *content = NULL;
965 memset(reply, 0, sizeof(*reply));
967 /* parse reply (XXX: use buffers) */
968 rt->last_reply[0] = '\0';
972 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
973 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
979 /* XXX: only parse it if first char on line ? */
980 if (return_on_interleaved_data) {
983 ff_rtsp_skip_packet(s);
984 } else if (ch != '\r') {
985 if ((q - buf) < sizeof(buf) - 1)
991 av_dlog(s, "line='%s'\n", buf);
993 /* test if last line */
997 if (line_count == 0) {
999 get_word(buf1, sizeof(buf1), &p);
1000 if (!strncmp(buf1, "RTSP/", 5)) {
1001 get_word(buf1, sizeof(buf1), &p);
1002 reply->status_code = atoi(buf1);
1003 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1005 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1006 get_word(buf1, sizeof(buf1), &p); // object
1010 ff_rtsp_parse_line(reply, p, rt, method);
1011 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1012 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1017 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1018 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1020 content_length = reply->content_length;
1021 if (content_length > 0) {
1022 /* leave some room for a trailing '\0' (useful for simple parsing) */
1023 content = av_malloc(content_length + 1);
1024 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1025 content[content_length] = '\0';
1028 *content_ptr = content;
1034 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1035 const char* ptr = buf;
1037 if (!strcmp(reply->reason, "OPTIONS")) {
1038 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1040 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1041 if (reply->session_id[0])
1042 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1045 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1047 av_strlcat(buf, "\r\n", sizeof(buf));
1049 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1050 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1053 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1055 rt->last_cmd_time = av_gettime();
1056 /* Even if the request from the server had data, it is not the data
1057 * that the caller wants or expects. The memory could also be leaked
1058 * if the actual following reply has content data. */
1060 av_freep(content_ptr);
1061 /* If method is set, this is called from ff_rtsp_send_cmd,
1062 * where a reply to exactly this request is awaited. For
1063 * callers from within packet receiving, we just want to
1064 * return to the caller and go back to receiving packets. */
1070 if (rt->seq != reply->seq) {
1071 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1072 rt->seq, reply->seq);
1076 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1077 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1078 reply->notice == 2306 /* Continuous Feed Terminated */) {
1079 rt->state = RTSP_STATE_IDLE;
1080 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1081 return AVERROR(EIO); /* data or server error */
1082 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1083 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1084 return AVERROR(EPERM);
1090 * Send a command to the RTSP server without waiting for the reply.
1092 * @param s RTSP (de)muxer context
1093 * @param method the method for the request
1094 * @param url the target url for the request
1095 * @param headers extra header lines to include in the request
1096 * @param send_content if non-null, the data to send as request body content
1097 * @param send_content_length the length of the send_content data, or 0 if
1098 * send_content is null
1100 * @return zero if success, nonzero otherwise
1102 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1103 const char *method, const char *url,
1104 const char *headers,
1105 const unsigned char *send_content,
1106 int send_content_length)
1108 RTSPState *rt = s->priv_data;
1109 char buf[4096], *out_buf;
1110 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1112 /* Add in RTSP headers */
1115 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1117 av_strlcat(buf, headers, sizeof(buf));
1118 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1119 if (rt->session_id[0] != '\0' && (!headers ||
1120 !strstr(headers, "\nIf-Match:"))) {
1121 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1124 char *str = ff_http_auth_create_response(&rt->auth_state,
1125 rt->auth, url, method);
1127 av_strlcat(buf, str, sizeof(buf));
1130 if (send_content_length > 0 && send_content)
1131 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1132 av_strlcat(buf, "\r\n", sizeof(buf));
1134 /* base64 encode rtsp if tunneling */
1135 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1136 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1137 out_buf = base64buf;
1140 av_dlog(s, "Sending:\n%s--\n", buf);
1142 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1143 if (send_content_length > 0 && send_content) {
1144 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1145 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1146 "with content data not supported\n");
1147 return AVERROR_PATCHWELCOME;
1149 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1151 rt->last_cmd_time = av_gettime();
1156 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1157 const char *url, const char *headers)
1159 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1162 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1163 const char *headers, RTSPMessageHeader *reply,
1164 unsigned char **content_ptr)
1166 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1167 content_ptr, NULL, 0);
1170 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1171 const char *method, const char *url,
1173 RTSPMessageHeader *reply,
1174 unsigned char **content_ptr,
1175 const unsigned char *send_content,
1176 int send_content_length)
1178 RTSPState *rt = s->priv_data;
1179 HTTPAuthType cur_auth_type;
1180 int ret, attempts = 0;
1183 cur_auth_type = rt->auth_state.auth_type;
1184 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1186 send_content_length)))
1189 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1193 if (reply->status_code == 401 &&
1194 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1195 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1198 if (reply->status_code > 400){
1199 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1203 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1209 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1210 int lower_transport, const char *real_challenge)
1212 RTSPState *rt = s->priv_data;
1213 int rtx = 0, j, i, err, interleave = 0, port_off;
1214 RTSPStream *rtsp_st;
1215 RTSPMessageHeader reply1, *reply = &reply1;
1217 const char *trans_pref;
1219 if (rt->transport == RTSP_TRANSPORT_RDT)
1220 trans_pref = "x-pn-tng";
1221 else if (rt->transport == RTSP_TRANSPORT_RAW)
1222 trans_pref = "RAW/RAW";
1224 trans_pref = "RTP/AVP";
1226 /* default timeout: 1 minute */
1229 /* Choose a random starting offset within the first half of the
1230 * port range, to allow for a number of ports to try even if the offset
1231 * happens to be at the end of the random range. */
1232 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1233 /* even random offset */
1234 port_off -= port_off & 0x01;
1236 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1237 char transport[2048];
1240 * WMS serves all UDP data over a single connection, the RTX, which
1241 * isn't necessarily the first in the SDP but has to be the first
1242 * to be set up, else the second/third SETUP will fail with a 461.
1244 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1245 rt->server_type == RTSP_SERVER_WMS) {
1248 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1249 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1251 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1255 if (rtx == rt->nb_rtsp_streams)
1256 return -1; /* no RTX found */
1257 rtsp_st = rt->rtsp_streams[rtx];
1259 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1261 rtsp_st = rt->rtsp_streams[i];
1264 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1267 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1268 port = reply->transports[0].client_port_min;
1272 /* first try in specified port range */
1273 while (j <= rt->rtp_port_max) {
1274 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1275 "?localport=%d", j);
1276 /* we will use two ports per rtp stream (rtp and rtcp) */
1278 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1279 &s->interrupt_callback, NULL))
1282 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1287 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1289 snprintf(transport, sizeof(transport) - 1,
1290 "%s/UDP;", trans_pref);
1291 if (rt->server_type != RTSP_SERVER_REAL)
1292 av_strlcat(transport, "unicast;", sizeof(transport));
1293 av_strlcatf(transport, sizeof(transport),
1294 "client_port=%d", port);
1295 if (rt->transport == RTSP_TRANSPORT_RTP &&
1296 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1297 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1301 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1302 /* For WMS streams, the application streams are only used for
1303 * UDP. When trying to set it up for TCP streams, the server
1304 * will return an error. Therefore, we skip those streams. */
1305 if (rt->server_type == RTSP_SERVER_WMS &&
1306 (rtsp_st->stream_index < 0 ||
1307 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1310 snprintf(transport, sizeof(transport) - 1,
1311 "%s/TCP;", trans_pref);
1312 if (rt->transport != RTSP_TRANSPORT_RDT)
1313 av_strlcat(transport, "unicast;", sizeof(transport));
1314 av_strlcatf(transport, sizeof(transport),
1315 "interleaved=%d-%d",
1316 interleave, interleave + 1);
1320 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1321 snprintf(transport, sizeof(transport) - 1,
1322 "%s/UDP;multicast", trans_pref);
1325 av_strlcat(transport, ";mode=record", sizeof(transport));
1326 } else if (rt->server_type == RTSP_SERVER_REAL ||
1327 rt->server_type == RTSP_SERVER_WMS)
1328 av_strlcat(transport, ";mode=play", sizeof(transport));
1329 snprintf(cmd, sizeof(cmd),
1330 "Transport: %s\r\n",
1332 if (rt->accept_dynamic_rate)
1333 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1334 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1335 char real_res[41], real_csum[9];
1336 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1338 av_strlcatf(cmd, sizeof(cmd),
1340 "RealChallenge2: %s, sd=%s\r\n",
1341 rt->session_id, real_res, real_csum);
1343 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1344 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1347 } else if (reply->status_code != RTSP_STATUS_OK ||
1348 reply->nb_transports != 1) {
1349 err = AVERROR_INVALIDDATA;
1353 /* XXX: same protocol for all streams is required */
1355 if (reply->transports[0].lower_transport != rt->lower_transport ||
1356 reply->transports[0].transport != rt->transport) {
1357 err = AVERROR_INVALIDDATA;
1361 rt->lower_transport = reply->transports[0].lower_transport;
1362 rt->transport = reply->transports[0].transport;
1365 /* Fail if the server responded with another lower transport mode
1366 * than what we requested. */
1367 if (reply->transports[0].lower_transport != lower_transport) {
1368 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1369 err = AVERROR_INVALIDDATA;
1373 switch(reply->transports[0].lower_transport) {
1374 case RTSP_LOWER_TRANSPORT_TCP:
1375 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1376 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1379 case RTSP_LOWER_TRANSPORT_UDP: {
1380 char url[1024], options[30] = "";
1382 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1383 av_strlcpy(options, "?connect=1", sizeof(options));
1384 /* Use source address if specified */
1385 if (reply->transports[0].source[0]) {
1386 ff_url_join(url, sizeof(url), "rtp", NULL,
1387 reply->transports[0].source,
1388 reply->transports[0].server_port_min, "%s", options);
1390 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1391 reply->transports[0].server_port_min, "%s", options);
1393 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1394 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1395 err = AVERROR_INVALIDDATA;
1398 /* Try to initialize the connection state in a
1399 * potential NAT router by sending dummy packets.
1400 * RTP/RTCP dummy packets are used for RDT, too.
1402 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1404 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1407 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1408 char url[1024], namebuf[50], optbuf[20] = "";
1409 struct sockaddr_storage addr;
1412 if (reply->transports[0].destination.ss_family) {
1413 addr = reply->transports[0].destination;
1414 port = reply->transports[0].port_min;
1415 ttl = reply->transports[0].ttl;
1417 addr = rtsp_st->sdp_ip;
1418 port = rtsp_st->sdp_port;
1419 ttl = rtsp_st->sdp_ttl;
1422 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1423 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1424 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1425 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1426 port, "%s", optbuf);
1427 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1428 &s->interrupt_callback, NULL) < 0) {
1429 err = AVERROR_INVALIDDATA;
1436 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1440 if (rt->nb_rtsp_streams && reply->timeout > 0)
1441 rt->timeout = reply->timeout;
1443 if (rt->server_type == RTSP_SERVER_REAL)
1444 rt->need_subscription = 1;
1449 ff_rtsp_undo_setup(s);
1453 void ff_rtsp_close_connections(AVFormatContext *s)
1455 RTSPState *rt = s->priv_data;
1456 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1457 ffurl_close(rt->rtsp_hd);
1458 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1461 int ff_rtsp_connect(AVFormatContext *s)
1463 RTSPState *rt = s->priv_data;
1464 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1465 int port, err, tcp_fd;
1466 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1467 int lower_transport_mask = 0;
1468 char real_challenge[64] = "";
1469 struct sockaddr_storage peer;
1470 socklen_t peer_len = sizeof(peer);
1472 if (rt->rtp_port_max < rt->rtp_port_min) {
1473 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1474 "than min port %d\n", rt->rtp_port_max,
1476 return AVERROR(EINVAL);
1479 if (!ff_network_init())
1480 return AVERROR(EIO);
1482 if (s->max_delay < 0) /* Not set by the caller */
1483 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1485 rt->control_transport = RTSP_MODE_PLAIN;
1486 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1487 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1488 rt->control_transport = RTSP_MODE_TUNNEL;
1490 /* Only pass through valid flags from here */
1491 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1494 lower_transport_mask = rt->lower_transport_mask;
1495 /* extract hostname and port */
1496 av_url_split(NULL, 0, auth, sizeof(auth),
1497 host, sizeof(host), &port, path, sizeof(path), s->filename);
1499 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1502 port = RTSP_DEFAULT_PORT;
1504 if (!lower_transport_mask)
1505 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1508 /* Only UDP or TCP - UDP multicast isn't supported. */
1509 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1510 (1 << RTSP_LOWER_TRANSPORT_TCP);
1511 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1512 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1513 "only UDP and TCP are supported for output.\n");
1514 err = AVERROR(EINVAL);
1519 /* Construct the URI used in request; this is similar to s->filename,
1520 * but with authentication credentials removed and RTSP specific options
1522 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1523 host, port, "%s", path);
1525 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1526 /* set up initial handshake for tunneling */
1527 char httpname[1024];
1528 char sessioncookie[17];
1531 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1532 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1533 av_get_random_seed(), av_get_random_seed());
1536 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1537 &s->interrupt_callback) < 0) {
1542 /* generate GET headers */
1543 snprintf(headers, sizeof(headers),
1544 "x-sessioncookie: %s\r\n"
1545 "Accept: application/x-rtsp-tunnelled\r\n"
1546 "Pragma: no-cache\r\n"
1547 "Cache-Control: no-cache\r\n",
1549 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1551 /* complete the connection */
1552 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1558 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1559 &s->interrupt_callback) < 0 ) {
1564 /* generate POST headers */
1565 snprintf(headers, sizeof(headers),
1566 "x-sessioncookie: %s\r\n"
1567 "Content-Type: application/x-rtsp-tunnelled\r\n"
1568 "Pragma: no-cache\r\n"
1569 "Cache-Control: no-cache\r\n"
1570 "Content-Length: 32767\r\n"
1571 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1573 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1574 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1576 /* Initialize the authentication state for the POST session. The HTTP
1577 * protocol implementation doesn't properly handle multi-pass
1578 * authentication for POST requests, since it would require one of
1580 * - implementing Expect: 100-continue, which many HTTP servers
1581 * don't support anyway, even less the RTSP servers that do HTTP
1583 * - sending the whole POST data until getting a 401 reply specifying
1584 * what authentication method to use, then resending all that data
1585 * - waiting for potential 401 replies directly after sending the
1586 * POST header (waiting for some unspecified time)
1587 * Therefore, we copy the full auth state, which works for both basic
1588 * and digest. (For digest, we would have to synchronize the nonce
1589 * count variable between the two sessions, if we'd do more requests
1590 * with the original session, though.)
1592 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1594 /* complete the connection */
1595 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1600 /* open the tcp connection */
1601 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1602 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1603 &s->interrupt_callback, NULL) < 0) {
1607 rt->rtsp_hd_out = rt->rtsp_hd;
1611 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1612 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1613 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1614 NULL, 0, NI_NUMERICHOST);
1617 /* request options supported by the server; this also detects server
1619 for (rt->server_type = RTSP_SERVER_RTP;;) {
1621 if (rt->server_type == RTSP_SERVER_REAL)
1624 * The following entries are required for proper
1625 * streaming from a Realmedia server. They are
1626 * interdependent in some way although we currently
1627 * don't quite understand how. Values were copied
1628 * from mplayer SVN r23589.
1629 * ClientChallenge is a 16-byte ID in hex
1630 * CompanyID is a 16-byte ID in base64
1632 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1633 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1634 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1635 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1637 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1638 if (reply->status_code != RTSP_STATUS_OK) {
1639 err = AVERROR_INVALIDDATA;
1643 /* detect server type if not standard-compliant RTP */
1644 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1645 rt->server_type = RTSP_SERVER_REAL;
1647 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1648 rt->server_type = RTSP_SERVER_WMS;
1649 } else if (rt->server_type == RTSP_SERVER_REAL)
1650 strcpy(real_challenge, reply->real_challenge);
1654 if (s->iformat && CONFIG_RTSP_DEMUXER)
1655 err = ff_rtsp_setup_input_streams(s, reply);
1656 else if (CONFIG_RTSP_MUXER)
1657 err = ff_rtsp_setup_output_streams(s, host);
1662 int lower_transport = ff_log2_tab[lower_transport_mask &
1663 ~(lower_transport_mask - 1)];
1665 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1666 rt->server_type == RTSP_SERVER_REAL ?
1667 real_challenge : NULL);
1670 lower_transport_mask &= ~(1 << lower_transport);
1671 if (lower_transport_mask == 0 && err == 1) {
1672 err = AVERROR(EPROTONOSUPPORT);
1677 rt->lower_transport_mask = lower_transport_mask;
1678 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1679 rt->state = RTSP_STATE_IDLE;
1680 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1683 ff_rtsp_close_streams(s);
1684 ff_rtsp_close_connections(s);
1685 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1686 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1687 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1695 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1698 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1699 uint8_t *buf, int buf_size, int64_t wait_end)
1701 RTSPState *rt = s->priv_data;
1702 RTSPStream *rtsp_st;
1703 int n, i, ret, tcp_fd, timeout_cnt = 0;
1705 struct pollfd *p = rt->p;
1706 int *fds = NULL, fdsnum, fdsidx;
1709 if (ff_check_interrupt(&s->interrupt_callback))
1710 return AVERROR_EXIT;
1711 if (wait_end && wait_end - av_gettime() < 0)
1712 return AVERROR(EAGAIN);
1715 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1716 p[max_p].fd = tcp_fd;
1717 p[max_p++].events = POLLIN;
1721 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1722 rtsp_st = rt->rtsp_streams[i];
1723 if (rtsp_st->rtp_handle) {
1724 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1726 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1730 av_log(s, AV_LOG_ERROR,
1731 "Number of fds %d not supported\n", fdsnum);
1732 return AVERROR_INVALIDDATA;
1734 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1735 p[max_p].fd = fds[fdsidx];
1736 p[max_p++].events = POLLIN;
1741 n = poll(p, max_p, POLL_TIMEOUT_MS);
1743 int j = 1 - (tcp_fd == -1);
1745 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1746 rtsp_st = rt->rtsp_streams[i];
1747 if (rtsp_st->rtp_handle) {
1748 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1749 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1751 *prtsp_st = rtsp_st;
1758 #if CONFIG_RTSP_DEMUXER
1759 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1760 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1761 if (rt->state == RTSP_STATE_STREAMING) {
1762 if (!ff_rtsp_parse_streaming_commands(s))
1765 av_log(s, AV_LOG_WARNING,
1766 "Unable to answer to TEARDOWN\n");
1770 RTSPMessageHeader reply;
1771 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1774 /* XXX: parse message */
1775 if (rt->state != RTSP_STATE_STREAMING)
1780 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1781 return AVERROR(ETIMEDOUT);
1782 } else if (n < 0 && errno != EINTR)
1783 return AVERROR(errno);
1787 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1789 RTSPState *rt = s->priv_data;
1791 RTSPStream *rtsp_st, *first_queue_st = NULL;
1792 int64_t wait_end = 0;
1794 if (rt->nb_byes == rt->nb_rtsp_streams)
1797 /* get next frames from the same RTP packet */
1798 if (rt->cur_transport_priv) {
1799 if (rt->transport == RTSP_TRANSPORT_RDT) {
1800 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1801 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1802 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1803 } else if (rt->ts && CONFIG_RTPDEC) {
1804 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1806 rt->recvbuf_pos += ret;
1807 ret = rt->recvbuf_pos < rt->recvbuf_len;
1812 rt->cur_transport_priv = NULL;
1814 } else if (ret == 1) {
1817 rt->cur_transport_priv = NULL;
1820 if (rt->transport == RTSP_TRANSPORT_RTP) {
1822 int64_t first_queue_time = 0;
1823 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1824 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1828 queue_time = ff_rtp_queued_packet_time(rtpctx);
1829 if (queue_time && (queue_time - first_queue_time < 0 ||
1830 !first_queue_time)) {
1831 first_queue_time = queue_time;
1832 first_queue_st = rt->rtsp_streams[i];
1835 if (first_queue_time)
1836 wait_end = first_queue_time + s->max_delay;
1839 /* read next RTP packet */
1842 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1844 return AVERROR(ENOMEM);
1847 switch(rt->lower_transport) {
1849 #if CONFIG_RTSP_DEMUXER
1850 case RTSP_LOWER_TRANSPORT_TCP:
1851 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1854 case RTSP_LOWER_TRANSPORT_UDP:
1855 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1856 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1857 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1858 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1861 if (len == AVERROR(EAGAIN) && first_queue_st &&
1862 rt->transport == RTSP_TRANSPORT_RTP) {
1863 rtsp_st = first_queue_st;
1864 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1871 if (rt->transport == RTSP_TRANSPORT_RDT) {
1872 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1873 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1874 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1876 /* Either bad packet, or a RTCP packet. Check if the
1877 * first_rtcp_ntp_time field was initialized. */
1878 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1879 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1880 /* first_rtcp_ntp_time has been initialized for this stream,
1881 * copy the same value to all other uninitialized streams,
1882 * in order to map their timestamp origin to the same ntp time
1885 AVStream *st = NULL;
1886 if (rtsp_st->stream_index >= 0)
1887 st = s->streams[rtsp_st->stream_index];
1888 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1889 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1890 AVStream *st2 = NULL;
1891 if (rt->rtsp_streams[i]->stream_index >= 0)
1892 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1893 if (rtpctx2 && st && st2 &&
1894 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1895 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1896 rtpctx2->rtcp_ts_offset = av_rescale_q(
1897 rtpctx->rtcp_ts_offset, st->time_base,
1902 if (ret == -RTCP_BYE) {
1905 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1906 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1908 if (rt->nb_byes == rt->nb_rtsp_streams)
1912 } else if (rt->ts && CONFIG_RTPDEC) {
1913 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1916 rt->recvbuf_len = len;
1917 rt->recvbuf_pos = ret;
1918 rt->cur_transport_priv = rt->ts;
1925 return AVERROR_INVALIDDATA;
1931 /* more packets may follow, so we save the RTP context */
1932 rt->cur_transport_priv = rtsp_st->transport_priv;
1936 #endif /* CONFIG_RTPDEC */
1938 #if CONFIG_SDP_DEMUXER
1939 static int sdp_probe(AVProbeData *p1)
1941 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1943 /* we look for a line beginning "c=IN IP" */
1944 while (p < p_end && *p != '\0') {
1945 if (p + sizeof("c=IN IP") - 1 < p_end &&
1946 av_strstart(p, "c=IN IP", NULL))
1947 return AVPROBE_SCORE_MAX / 2;
1949 while (p < p_end - 1 && *p != '\n') p++;
1958 static int sdp_read_header(AVFormatContext *s)
1960 RTSPState *rt = s->priv_data;
1961 RTSPStream *rtsp_st;
1966 if (!ff_network_init())
1967 return AVERROR(EIO);
1969 if (s->max_delay < 0) /* Not set by the caller */
1970 s->max_delay = DEFAULT_REORDERING_DELAY;
1972 /* read the whole sdp file */
1973 /* XXX: better loading */
1974 content = av_malloc(SDP_MAX_SIZE);
1975 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1978 return AVERROR_INVALIDDATA;
1980 content[size] ='\0';
1982 err = ff_sdp_parse(s, content);
1986 /* open each RTP stream */
1987 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1989 rtsp_st = rt->rtsp_streams[i];
1991 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1992 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1993 ff_url_join(url, sizeof(url), "rtp", NULL,
1994 namebuf, rtsp_st->sdp_port,
1995 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1997 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1998 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1999 &s->interrupt_callback, NULL) < 0) {
2000 err = AVERROR_INVALIDDATA;
2003 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2008 ff_rtsp_close_streams(s);
2013 static int sdp_read_close(AVFormatContext *s)
2015 ff_rtsp_close_streams(s);
2020 static const AVClass sdp_demuxer_class = {
2021 .class_name = "SDP demuxer",
2022 .item_name = av_default_item_name,
2023 .option = sdp_options,
2024 .version = LIBAVUTIL_VERSION_INT,
2027 AVInputFormat ff_sdp_demuxer = {
2029 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2030 .priv_data_size = sizeof(RTSPState),
2031 .read_probe = sdp_probe,
2032 .read_header = sdp_read_header,
2033 .read_packet = ff_rtsp_fetch_packet,
2034 .read_close = sdp_read_close,
2035 .priv_class = &sdp_demuxer_class,
2037 #endif /* CONFIG_SDP_DEMUXER */
2039 #if CONFIG_RTP_DEMUXER
2040 static int rtp_probe(AVProbeData *p)
2042 if (av_strstart(p->filename, "rtp:", NULL))
2043 return AVPROBE_SCORE_MAX;
2047 static int rtp_read_header(AVFormatContext *s)
2049 uint8_t recvbuf[1500];
2050 char host[500], sdp[500];
2052 URLContext* in = NULL;
2054 AVCodecContext codec = { 0 };
2055 struct sockaddr_storage addr;
2057 socklen_t addrlen = sizeof(addr);
2058 RTSPState *rt = s->priv_data;
2060 if (!ff_network_init())
2061 return AVERROR(EIO);
2063 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2064 &s->interrupt_callback, NULL);
2069 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2070 if (ret == AVERROR(EAGAIN))
2075 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2079 if ((recvbuf[0] & 0xc0) != 0x80) {
2080 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2085 if (RTP_PT_IS_RTCP(recvbuf[1]))
2088 payload_type = recvbuf[1] & 0x7f;
2091 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2095 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2096 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2097 "without an SDP file describing it\n",
2101 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2102 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2103 "properly you need an SDP file "
2107 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2108 NULL, 0, s->filename);
2110 snprintf(sdp, sizeof(sdp),
2111 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2112 addr.ss_family == AF_INET ? 4 : 6, host,
2113 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2114 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2115 port, payload_type);
2116 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2118 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2121 /* sdp_read_header initializes this again */
2124 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2126 ret = sdp_read_header(s);
2137 static const AVClass rtp_demuxer_class = {
2138 .class_name = "RTP demuxer",
2139 .item_name = av_default_item_name,
2140 .option = rtp_options,
2141 .version = LIBAVUTIL_VERSION_INT,
2144 AVInputFormat ff_rtp_demuxer = {
2146 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2147 .priv_data_size = sizeof(RTSPState),
2148 .read_probe = rtp_probe,
2149 .read_header = rtp_read_header,
2150 .read_packet = ff_rtsp_fetch_packet,
2151 .read_close = sdp_read_close,
2152 .flags = AVFMT_NOFILE,
2153 .priv_class = &rtp_demuxer_class,
2155 #endif /* CONFIG_RTP_DEMUXER */