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 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
100 RTSP_REORDERING_OPTS(),
104 static const AVOption rtp_options[] = {
105 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
106 RTSP_REORDERING_OPTS(),
110 static void get_word_until_chars(char *buf, int buf_size,
111 const char *sep, const char **pp)
117 p += strspn(p, SPACE_CHARS);
119 while (!strchr(sep, *p) && *p != '\0') {
120 if ((q - buf) < buf_size - 1)
129 static void get_word_sep(char *buf, int buf_size, const char *sep,
132 if (**pp == '/') (*pp)++;
133 get_word_until_chars(buf, buf_size, sep, pp);
136 static void get_word(char *buf, int buf_size, const char **pp)
138 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
141 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
143 * Used for seeking in the rtp stream.
145 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
149 p += strspn(p, SPACE_CHARS);
150 if (!av_stristart(p, "npt=", &p))
153 *start = AV_NOPTS_VALUE;
154 *end = AV_NOPTS_VALUE;
156 get_word_sep(buf, sizeof(buf), "-", &p);
157 av_parse_time(start, buf, 1);
160 get_word_sep(buf, sizeof(buf), "-", &p);
161 av_parse_time(end, buf, 1);
165 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
167 struct addrinfo hints = { 0 }, *ai = NULL;
168 hints.ai_flags = AI_NUMERICHOST;
169 if (getaddrinfo(buf, NULL, &hints, &ai))
171 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
177 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
178 RTSPStream *rtsp_st, AVCodecContext *codec)
182 codec->codec_id = handler->codec_id;
183 rtsp_st->dynamic_handler = handler;
184 if (handler->alloc) {
185 rtsp_st->dynamic_protocol_context = handler->alloc();
186 if (!rtsp_st->dynamic_protocol_context)
187 rtsp_st->dynamic_handler = NULL;
191 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
192 static int sdp_parse_rtpmap(AVFormatContext *s,
193 AVStream *st, RTSPStream *rtsp_st,
194 int payload_type, const char *p)
196 AVCodecContext *codec = st->codec;
202 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
203 * see if we can handle this kind of payload.
204 * The space should normally not be there but some Real streams or
205 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
206 * have a trailing space. */
207 get_word_sep(buf, sizeof(buf), "/ ", &p);
208 if (payload_type < RTP_PT_PRIVATE) {
209 /* We are in a standard case
210 * (from http://www.iana.org/assignments/rtp-parameters). */
211 /* search into AVRtpPayloadTypes[] */
212 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
215 if (codec->codec_id == AV_CODEC_ID_NONE) {
216 RTPDynamicProtocolHandler *handler =
217 ff_rtp_handler_find_by_name(buf, codec->codec_type);
218 init_rtp_handler(handler, rtsp_st, codec);
219 /* If no dynamic handler was found, check with the list of standard
220 * allocated types, if such a stream for some reason happens to
221 * use a private payload type. This isn't handled in rtpdec.c, since
222 * the format name from the rtpmap line never is passed into rtpdec. */
223 if (!rtsp_st->dynamic_handler)
224 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
227 c = avcodec_find_decoder(codec->codec_id);
233 get_word_sep(buf, sizeof(buf), "/", &p);
235 switch (codec->codec_type) {
236 case AVMEDIA_TYPE_AUDIO:
237 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
238 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
239 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
241 codec->sample_rate = i;
242 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243 get_word_sep(buf, sizeof(buf), "/", &p);
247 // TODO: there is a bug here; if it is a mono stream, and
248 // less than 22000Hz, faad upconverts to stereo and twice
249 // the frequency. No problem, but the sample rate is being
250 // set here by the sdp line. Patch on its way. (rdm)
252 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
254 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
257 case AVMEDIA_TYPE_VIDEO:
258 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
260 avpriv_set_pts_info(st, 32, 1, i);
265 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
266 rtsp_st->dynamic_handler->init(s, st->index,
267 rtsp_st->dynamic_protocol_context);
271 /* parse the attribute line from the fmtp a line of an sdp response. This
272 * is broken out as a function because it is used in rtp_h264.c, which is
274 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
275 char *value, int value_size)
277 *p += strspn(*p, SPACE_CHARS);
279 get_word_sep(attr, attr_size, "=", p);
282 get_word_sep(value, value_size, ";", p);
290 typedef struct SDPParseState {
292 struct sockaddr_storage default_ip;
294 int skip_media; ///< set if an unknown m= line occurs
297 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
298 int letter, const char *buf)
300 RTSPState *rt = s->priv_data;
301 char buf1[64], st_type[64];
303 enum AVMediaType codec_type;
307 struct sockaddr_storage sdp_ip;
310 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
313 if (s1->skip_media && letter != 'm')
317 get_word(buf1, sizeof(buf1), &p);
318 if (strcmp(buf1, "IN") != 0)
320 get_word(buf1, sizeof(buf1), &p);
321 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
323 get_word_sep(buf1, sizeof(buf1), "/", &p);
324 if (get_sockaddr(buf1, &sdp_ip))
329 get_word_sep(buf1, sizeof(buf1), "/", &p);
332 if (s->nb_streams == 0) {
333 s1->default_ip = sdp_ip;
334 s1->default_ttl = ttl;
336 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
337 rtsp_st->sdp_ip = sdp_ip;
338 rtsp_st->sdp_ttl = ttl;
342 av_dict_set(&s->metadata, "title", p, 0);
345 if (s->nb_streams == 0) {
346 av_dict_set(&s->metadata, "comment", p, 0);
353 codec_type = AVMEDIA_TYPE_UNKNOWN;
354 get_word(st_type, sizeof(st_type), &p);
355 if (!strcmp(st_type, "audio")) {
356 codec_type = AVMEDIA_TYPE_AUDIO;
357 } else if (!strcmp(st_type, "video")) {
358 codec_type = AVMEDIA_TYPE_VIDEO;
359 } else if (!strcmp(st_type, "application")) {
360 codec_type = AVMEDIA_TYPE_DATA;
362 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
366 rtsp_st = av_mallocz(sizeof(RTSPStream));
369 rtsp_st->stream_index = -1;
370 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
372 rtsp_st->sdp_ip = s1->default_ip;
373 rtsp_st->sdp_ttl = s1->default_ttl;
375 get_word(buf1, sizeof(buf1), &p); /* port */
376 rtsp_st->sdp_port = atoi(buf1);
378 get_word(buf1, sizeof(buf1), &p); /* protocol */
379 if (!strcmp(buf1, "udp"))
380 rt->transport = RTSP_TRANSPORT_RAW;
382 /* XXX: handle list of formats */
383 get_word(buf1, sizeof(buf1), &p); /* format list */
384 rtsp_st->sdp_payload_type = atoi(buf1);
386 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
387 /* no corresponding stream */
388 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
389 rt->ts = ff_mpegts_parse_open(s);
390 } else if (rt->server_type == RTSP_SERVER_WMS &&
391 codec_type == AVMEDIA_TYPE_DATA) {
392 /* RTX stream, a stream that carries all the other actual
393 * audio/video streams. Don't expose this to the callers. */
395 st = avformat_new_stream(s, NULL);
398 st->id = rt->nb_rtsp_streams - 1;
399 rtsp_st->stream_index = st->index;
400 st->codec->codec_type = codec_type;
401 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
402 RTPDynamicProtocolHandler *handler;
403 /* if standard payload type, we can find the codec right now */
404 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
405 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
406 st->codec->sample_rate > 0)
407 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
408 /* Even static payload types may need a custom depacketizer */
409 handler = ff_rtp_handler_find_by_id(
410 rtsp_st->sdp_payload_type, st->codec->codec_type);
411 init_rtp_handler(handler, rtsp_st, st->codec);
412 if (handler && handler->init)
413 handler->init(s, st->index,
414 rtsp_st->dynamic_protocol_context);
417 /* put a default control url */
418 av_strlcpy(rtsp_st->control_url, rt->control_uri,
419 sizeof(rtsp_st->control_url));
422 if (av_strstart(p, "control:", &p)) {
423 if (s->nb_streams == 0) {
424 if (!strncmp(p, "rtsp://", 7))
425 av_strlcpy(rt->control_uri, p,
426 sizeof(rt->control_uri));
429 /* get the control url */
430 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
432 /* XXX: may need to add full url resolution */
433 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
435 if (proto[0] == '\0') {
436 /* relative control URL */
437 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
438 av_strlcat(rtsp_st->control_url, "/",
439 sizeof(rtsp_st->control_url));
440 av_strlcat(rtsp_st->control_url, p,
441 sizeof(rtsp_st->control_url));
443 av_strlcpy(rtsp_st->control_url, p,
444 sizeof(rtsp_st->control_url));
446 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
447 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
448 get_word(buf1, sizeof(buf1), &p);
449 payload_type = atoi(buf1);
450 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
451 if (rtsp_st->stream_index >= 0) {
452 st = s->streams[rtsp_st->stream_index];
453 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
455 } else if (av_strstart(p, "fmtp:", &p) ||
456 av_strstart(p, "framesize:", &p)) {
457 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
458 // let dynamic protocol handlers have a stab at the line.
459 get_word(buf1, sizeof(buf1), &p);
460 payload_type = atoi(buf1);
461 for (i = 0; i < rt->nb_rtsp_streams; i++) {
462 rtsp_st = rt->rtsp_streams[i];
463 if (rtsp_st->sdp_payload_type == payload_type &&
464 rtsp_st->dynamic_handler &&
465 rtsp_st->dynamic_handler->parse_sdp_a_line)
466 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
467 rtsp_st->dynamic_protocol_context, buf);
469 } else if (av_strstart(p, "range:", &p)) {
472 // this is so that seeking on a streamed file can work.
473 rtsp_parse_range_npt(p, &start, &end);
474 s->start_time = start;
475 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
476 s->duration = (end == AV_NOPTS_VALUE) ?
477 AV_NOPTS_VALUE : end - start;
478 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
480 rt->transport = RTSP_TRANSPORT_RDT;
481 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
483 st = s->streams[s->nb_streams - 1];
484 st->codec->sample_rate = atoi(p);
486 if (rt->server_type == RTSP_SERVER_WMS)
487 ff_wms_parse_sdp_a_line(s, p);
488 if (s->nb_streams > 0) {
489 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
491 if (rt->server_type == RTSP_SERVER_REAL)
492 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
494 if (rtsp_st->dynamic_handler &&
495 rtsp_st->dynamic_handler->parse_sdp_a_line)
496 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
497 rtsp_st->stream_index,
498 rtsp_st->dynamic_protocol_context, buf);
505 int ff_sdp_parse(AVFormatContext *s, const char *content)
507 RTSPState *rt = s->priv_data;
510 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
511 * contain long SDP lines containing complete ASF Headers (several
512 * kB) or arrays of MDPR (RM stream descriptor) headers plus
513 * "rulebooks" describing their properties. Therefore, the SDP line
516 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
517 * in rtpdec_xiph.c. */
519 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
523 p += strspn(p, SPACE_CHARS);
531 /* get the content */
533 while (*p != '\n' && *p != '\r' && *p != '\0') {
534 if ((q - buf) < sizeof(buf) - 1)
539 sdp_parse_line(s, s1, letter, buf);
541 while (*p != '\n' && *p != '\0')
546 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
547 if (!rt->p) return AVERROR(ENOMEM);
550 #endif /* CONFIG_RTPDEC */
552 void ff_rtsp_undo_setup(AVFormatContext *s)
554 RTSPState *rt = s->priv_data;
557 for (i = 0; i < rt->nb_rtsp_streams; i++) {
558 RTSPStream *rtsp_st = rt->rtsp_streams[i];
561 if (rtsp_st->transport_priv) {
563 AVFormatContext *rtpctx = rtsp_st->transport_priv;
564 av_write_trailer(rtpctx);
565 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
567 avio_close_dyn_buf(rtpctx->pb, &ptr);
570 avio_close(rtpctx->pb);
572 avformat_free_context(rtpctx);
573 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
574 ff_rdt_parse_close(rtsp_st->transport_priv);
575 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
576 ff_rtp_parse_close(rtsp_st->transport_priv);
578 rtsp_st->transport_priv = NULL;
579 if (rtsp_st->rtp_handle)
580 ffurl_close(rtsp_st->rtp_handle);
581 rtsp_st->rtp_handle = NULL;
585 /* close and free RTSP streams */
586 void ff_rtsp_close_streams(AVFormatContext *s)
588 RTSPState *rt = s->priv_data;
592 ff_rtsp_undo_setup(s);
593 for (i = 0; i < rt->nb_rtsp_streams; i++) {
594 rtsp_st = rt->rtsp_streams[i];
596 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
597 rtsp_st->dynamic_handler->free(
598 rtsp_st->dynamic_protocol_context);
602 av_free(rt->rtsp_streams);
604 avformat_close_input(&rt->asf_ctx);
606 if (rt->ts && CONFIG_RTPDEC)
607 ff_mpegts_parse_close(rt->ts);
609 av_free(rt->recvbuf);
612 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
614 RTSPState *rt = s->priv_data;
616 int reordering_queue_size = rt->reordering_queue_size;
617 if (reordering_queue_size < 0) {
618 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
619 reordering_queue_size = 0;
621 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
624 /* open the RTP context */
625 if (rtsp_st->stream_index >= 0)
626 st = s->streams[rtsp_st->stream_index];
628 s->ctx_flags |= AVFMTCTX_NOHEADER;
630 if (s->oformat && CONFIG_RTSP_MUXER) {
631 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
633 RTSP_TCP_MAX_PACKET_SIZE);
634 /* Ownership of rtp_handle is passed to the rtp mux context */
635 rtsp_st->rtp_handle = NULL;
638 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
639 return 0; // Don't need to open any parser here
640 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
641 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
642 rtsp_st->dynamic_protocol_context,
643 rtsp_st->dynamic_handler);
644 else if (CONFIG_RTPDEC)
645 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
646 rtsp_st->sdp_payload_type,
647 reordering_queue_size);
649 if (!rtsp_st->transport_priv) {
650 return AVERROR(ENOMEM);
651 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
652 if (rtsp_st->dynamic_handler) {
653 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
654 rtsp_st->dynamic_protocol_context,
655 rtsp_st->dynamic_handler);
662 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
663 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
670 q += strspn(q, SPACE_CHARS);
671 v = strtol(q, &p, 10);
675 v = strtol(p, &p, 10);
684 /* XXX: only one transport specification is parsed */
685 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
687 char transport_protocol[16];
689 char lower_transport[16];
691 RTSPTransportField *th;
694 reply->nb_transports = 0;
697 p += strspn(p, SPACE_CHARS);
701 th = &reply->transports[reply->nb_transports];
703 get_word_sep(transport_protocol, sizeof(transport_protocol),
705 if (!av_strcasecmp (transport_protocol, "rtp")) {
706 get_word_sep(profile, sizeof(profile), "/;,", &p);
707 lower_transport[0] = '\0';
708 /* rtp/avp/<protocol> */
710 get_word_sep(lower_transport, sizeof(lower_transport),
713 th->transport = RTSP_TRANSPORT_RTP;
714 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
715 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
716 /* x-pn-tng/<protocol> */
717 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
719 th->transport = RTSP_TRANSPORT_RDT;
720 } else if (!av_strcasecmp(transport_protocol, "raw")) {
721 get_word_sep(profile, sizeof(profile), "/;,", &p);
722 lower_transport[0] = '\0';
723 /* raw/raw/<protocol> */
725 get_word_sep(lower_transport, sizeof(lower_transport),
728 th->transport = RTSP_TRANSPORT_RAW;
730 if (!av_strcasecmp(lower_transport, "TCP"))
731 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
733 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
737 /* get each parameter */
738 while (*p != '\0' && *p != ',') {
739 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
740 if (!strcmp(parameter, "port")) {
743 rtsp_parse_range(&th->port_min, &th->port_max, &p);
745 } else if (!strcmp(parameter, "client_port")) {
748 rtsp_parse_range(&th->client_port_min,
749 &th->client_port_max, &p);
751 } else if (!strcmp(parameter, "server_port")) {
754 rtsp_parse_range(&th->server_port_min,
755 &th->server_port_max, &p);
757 } else if (!strcmp(parameter, "interleaved")) {
760 rtsp_parse_range(&th->interleaved_min,
761 &th->interleaved_max, &p);
763 } else if (!strcmp(parameter, "multicast")) {
764 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
765 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
766 } else if (!strcmp(parameter, "ttl")) {
769 th->ttl = strtol(p, (char **)&p, 10);
771 } else if (!strcmp(parameter, "destination")) {
774 get_word_sep(buf, sizeof(buf), ";,", &p);
775 get_sockaddr(buf, &th->destination);
777 } else if (!strcmp(parameter, "source")) {
780 get_word_sep(buf, sizeof(buf), ";,", &p);
781 av_strlcpy(th->source, buf, sizeof(th->source));
783 } else if (!strcmp(parameter, "mode")) {
786 get_word_sep(buf, sizeof(buf), ";, ", &p);
787 if (!strcmp(buf, "record") ||
788 !strcmp(buf, "receive"))
793 while (*p != ';' && *p != '\0' && *p != ',')
801 reply->nb_transports++;
805 static void handle_rtp_info(RTSPState *rt, const char *url,
806 uint32_t seq, uint32_t rtptime)
809 if (!rtptime || !url[0])
811 if (rt->transport != RTSP_TRANSPORT_RTP)
813 for (i = 0; i < rt->nb_rtsp_streams; i++) {
814 RTSPStream *rtsp_st = rt->rtsp_streams[i];
815 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
818 if (!strcmp(rtsp_st->control_url, url)) {
819 rtpctx->base_timestamp = rtptime;
825 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
828 char key[20], value[1024], url[1024] = "";
829 uint32_t seq = 0, rtptime = 0;
832 p += strspn(p, SPACE_CHARS);
835 get_word_sep(key, sizeof(key), "=", &p);
839 get_word_sep(value, sizeof(value), ";, ", &p);
841 if (!strcmp(key, "url"))
842 av_strlcpy(url, value, sizeof(url));
843 else if (!strcmp(key, "seq"))
844 seq = strtoul(value, NULL, 10);
845 else if (!strcmp(key, "rtptime"))
846 rtptime = strtoul(value, NULL, 10);
848 handle_rtp_info(rt, url, seq, rtptime);
857 handle_rtp_info(rt, url, seq, rtptime);
860 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
861 RTSPState *rt, const char *method)
865 /* NOTE: we do case independent match for broken servers */
867 if (av_stristart(p, "Session:", &p)) {
869 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
870 if (av_stristart(p, ";timeout=", &p) &&
871 (t = strtol(p, NULL, 10)) > 0) {
874 } else if (av_stristart(p, "Content-Length:", &p)) {
875 reply->content_length = strtol(p, NULL, 10);
876 } else if (av_stristart(p, "Transport:", &p)) {
877 rtsp_parse_transport(reply, p);
878 } else if (av_stristart(p, "CSeq:", &p)) {
879 reply->seq = strtol(p, NULL, 10);
880 } else if (av_stristart(p, "Range:", &p)) {
881 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
882 } else if (av_stristart(p, "RealChallenge1:", &p)) {
883 p += strspn(p, SPACE_CHARS);
884 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
885 } else if (av_stristart(p, "Server:", &p)) {
886 p += strspn(p, SPACE_CHARS);
887 av_strlcpy(reply->server, p, sizeof(reply->server));
888 } else if (av_stristart(p, "Notice:", &p) ||
889 av_stristart(p, "X-Notice:", &p)) {
890 reply->notice = strtol(p, NULL, 10);
891 } else if (av_stristart(p, "Location:", &p)) {
892 p += strspn(p, SPACE_CHARS);
893 av_strlcpy(reply->location, p , sizeof(reply->location));
894 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
895 p += strspn(p, SPACE_CHARS);
896 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
897 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
898 p += strspn(p, SPACE_CHARS);
899 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
900 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
901 p += strspn(p, SPACE_CHARS);
902 if (method && !strcmp(method, "DESCRIBE"))
903 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
904 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
905 p += strspn(p, SPACE_CHARS);
906 if (method && !strcmp(method, "PLAY"))
907 rtsp_parse_rtp_info(rt, p);
908 } else if (av_stristart(p, "Public:", &p) && rt) {
909 if (strstr(p, "GET_PARAMETER") &&
910 method && !strcmp(method, "OPTIONS"))
911 rt->get_parameter_supported = 1;
912 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
913 p += strspn(p, SPACE_CHARS);
914 rt->accept_dynamic_rate = atoi(p);
915 } else if (av_stristart(p, "Content-Type:", &p)) {
916 p += strspn(p, SPACE_CHARS);
917 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
921 /* skip a RTP/TCP interleaved packet */
922 void ff_rtsp_skip_packet(AVFormatContext *s)
924 RTSPState *rt = s->priv_data;
928 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
931 len = AV_RB16(buf + 1);
933 av_dlog(s, "skipping RTP packet len=%d\n", len);
938 if (len1 > sizeof(buf))
940 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
947 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
948 unsigned char **content_ptr,
949 int return_on_interleaved_data, const char *method)
951 RTSPState *rt = s->priv_data;
952 char buf[4096], buf1[1024], *q;
955 int ret, content_length, line_count = 0, request = 0;
956 unsigned char *content = NULL;
962 memset(reply, 0, sizeof(*reply));
964 /* parse reply (XXX: use buffers) */
965 rt->last_reply[0] = '\0';
969 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
970 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
976 /* XXX: only parse it if first char on line ? */
977 if (return_on_interleaved_data) {
980 ff_rtsp_skip_packet(s);
981 } else if (ch != '\r') {
982 if ((q - buf) < sizeof(buf) - 1)
988 av_dlog(s, "line='%s'\n", buf);
990 /* test if last line */
994 if (line_count == 0) {
996 get_word(buf1, sizeof(buf1), &p);
997 if (!strncmp(buf1, "RTSP/", 5)) {
998 get_word(buf1, sizeof(buf1), &p);
999 reply->status_code = atoi(buf1);
1000 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1002 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1003 get_word(buf1, sizeof(buf1), &p); // object
1007 ff_rtsp_parse_line(reply, p, rt, method);
1008 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1009 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1014 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1015 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1017 content_length = reply->content_length;
1018 if (content_length > 0) {
1019 /* leave some room for a trailing '\0' (useful for simple parsing) */
1020 content = av_malloc(content_length + 1);
1021 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1022 content[content_length] = '\0';
1025 *content_ptr = content;
1031 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1032 const char* ptr = buf;
1034 if (!strcmp(reply->reason, "OPTIONS")) {
1035 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1037 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1038 if (reply->session_id[0])
1039 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1042 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1044 av_strlcat(buf, "\r\n", sizeof(buf));
1046 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1047 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1050 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1052 rt->last_cmd_time = av_gettime();
1053 /* Even if the request from the server had data, it is not the data
1054 * that the caller wants or expects. The memory could also be leaked
1055 * if the actual following reply has content data. */
1057 av_freep(content_ptr);
1058 /* If method is set, this is called from ff_rtsp_send_cmd,
1059 * where a reply to exactly this request is awaited. For
1060 * callers from within packet receiving, we just want to
1061 * return to the caller and go back to receiving packets. */
1067 if (rt->seq != reply->seq) {
1068 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1069 rt->seq, reply->seq);
1073 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1074 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1075 reply->notice == 2306 /* Continuous Feed Terminated */) {
1076 rt->state = RTSP_STATE_IDLE;
1077 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1078 return AVERROR(EIO); /* data or server error */
1079 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1080 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1081 return AVERROR(EPERM);
1087 * Send a command to the RTSP server without waiting for the reply.
1089 * @param s RTSP (de)muxer context
1090 * @param method the method for the request
1091 * @param url the target url for the request
1092 * @param headers extra header lines to include in the request
1093 * @param send_content if non-null, the data to send as request body content
1094 * @param send_content_length the length of the send_content data, or 0 if
1095 * send_content is null
1097 * @return zero if success, nonzero otherwise
1099 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1100 const char *method, const char *url,
1101 const char *headers,
1102 const unsigned char *send_content,
1103 int send_content_length)
1105 RTSPState *rt = s->priv_data;
1106 char buf[4096], *out_buf;
1107 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1109 /* Add in RTSP headers */
1112 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1114 av_strlcat(buf, headers, sizeof(buf));
1115 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1116 if (rt->session_id[0] != '\0' && (!headers ||
1117 !strstr(headers, "\nIf-Match:"))) {
1118 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1121 char *str = ff_http_auth_create_response(&rt->auth_state,
1122 rt->auth, url, method);
1124 av_strlcat(buf, str, sizeof(buf));
1127 if (send_content_length > 0 && send_content)
1128 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1129 av_strlcat(buf, "\r\n", sizeof(buf));
1131 /* base64 encode rtsp if tunneling */
1132 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1133 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1134 out_buf = base64buf;
1137 av_dlog(s, "Sending:\n%s--\n", buf);
1139 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1140 if (send_content_length > 0 && send_content) {
1141 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1142 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1143 "with content data not supported\n");
1144 return AVERROR_PATCHWELCOME;
1146 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1148 rt->last_cmd_time = av_gettime();
1153 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1154 const char *url, const char *headers)
1156 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1159 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1160 const char *headers, RTSPMessageHeader *reply,
1161 unsigned char **content_ptr)
1163 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1164 content_ptr, NULL, 0);
1167 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1168 const char *method, const char *url,
1170 RTSPMessageHeader *reply,
1171 unsigned char **content_ptr,
1172 const unsigned char *send_content,
1173 int send_content_length)
1175 RTSPState *rt = s->priv_data;
1176 HTTPAuthType cur_auth_type;
1177 int ret, attempts = 0;
1180 cur_auth_type = rt->auth_state.auth_type;
1181 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1183 send_content_length)))
1186 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1190 if (reply->status_code == 401 &&
1191 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1192 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1195 if (reply->status_code > 400){
1196 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1200 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1206 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1207 int lower_transport, const char *real_challenge)
1209 RTSPState *rt = s->priv_data;
1210 int rtx = 0, j, i, err, interleave = 0, port_off;
1211 RTSPStream *rtsp_st;
1212 RTSPMessageHeader reply1, *reply = &reply1;
1214 const char *trans_pref;
1216 if (rt->transport == RTSP_TRANSPORT_RDT)
1217 trans_pref = "x-pn-tng";
1218 else if (rt->transport == RTSP_TRANSPORT_RAW)
1219 trans_pref = "RAW/RAW";
1221 trans_pref = "RTP/AVP";
1223 /* default timeout: 1 minute */
1226 /* for each stream, make the setup request */
1227 /* XXX: we assume the same server is used for the control of each
1230 /* Choose a random starting offset within the first half of the
1231 * port range, to allow for a number of ports to try even if the offset
1232 * happens to be at the end of the random range. */
1233 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1234 /* even random offset */
1235 port_off -= port_off & 0x01;
1237 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1238 char transport[2048];
1241 * WMS serves all UDP data over a single connection, the RTX, which
1242 * isn't necessarily the first in the SDP but has to be the first
1243 * to be set up, else the second/third SETUP will fail with a 461.
1245 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1246 rt->server_type == RTSP_SERVER_WMS) {
1249 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1250 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1252 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1256 if (rtx == rt->nb_rtsp_streams)
1257 return -1; /* no RTX found */
1258 rtsp_st = rt->rtsp_streams[rtx];
1260 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1262 rtsp_st = rt->rtsp_streams[i];
1265 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1268 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1269 port = reply->transports[0].client_port_min;
1273 /* first try in specified port range */
1274 while (j <= rt->rtp_port_max) {
1275 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1276 "?localport=%d", j);
1277 /* we will use two ports per rtp stream (rtp and rtcp) */
1279 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1280 &s->interrupt_callback, NULL))
1284 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1289 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1291 snprintf(transport, sizeof(transport) - 1,
1292 "%s/UDP;", trans_pref);
1293 if (rt->server_type != RTSP_SERVER_REAL)
1294 av_strlcat(transport, "unicast;", sizeof(transport));
1295 av_strlcatf(transport, sizeof(transport),
1296 "client_port=%d", port);
1297 if (rt->transport == RTSP_TRANSPORT_RTP &&
1298 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1299 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1303 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1304 /* For WMS streams, the application streams are only used for
1305 * UDP. When trying to set it up for TCP streams, the server
1306 * will return an error. Therefore, we skip those streams. */
1307 if (rt->server_type == RTSP_SERVER_WMS &&
1308 (rtsp_st->stream_index < 0 ||
1309 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1312 snprintf(transport, sizeof(transport) - 1,
1313 "%s/TCP;", trans_pref);
1314 if (rt->transport != RTSP_TRANSPORT_RDT)
1315 av_strlcat(transport, "unicast;", sizeof(transport));
1316 av_strlcatf(transport, sizeof(transport),
1317 "interleaved=%d-%d",
1318 interleave, interleave + 1);
1322 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1323 snprintf(transport, sizeof(transport) - 1,
1324 "%s/UDP;multicast", trans_pref);
1327 av_strlcat(transport, ";mode=record", sizeof(transport));
1328 } else if (rt->server_type == RTSP_SERVER_REAL ||
1329 rt->server_type == RTSP_SERVER_WMS)
1330 av_strlcat(transport, ";mode=play", sizeof(transport));
1331 snprintf(cmd, sizeof(cmd),
1332 "Transport: %s\r\n",
1334 if (rt->accept_dynamic_rate)
1335 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1336 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1337 char real_res[41], real_csum[9];
1338 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1340 av_strlcatf(cmd, sizeof(cmd),
1342 "RealChallenge2: %s, sd=%s\r\n",
1343 rt->session_id, real_res, real_csum);
1345 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1346 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1349 } else if (reply->status_code != RTSP_STATUS_OK ||
1350 reply->nb_transports != 1) {
1351 err = AVERROR_INVALIDDATA;
1355 /* XXX: same protocol for all streams is required */
1357 if (reply->transports[0].lower_transport != rt->lower_transport ||
1358 reply->transports[0].transport != rt->transport) {
1359 err = AVERROR_INVALIDDATA;
1363 rt->lower_transport = reply->transports[0].lower_transport;
1364 rt->transport = reply->transports[0].transport;
1367 /* Fail if the server responded with another lower transport mode
1368 * than what we requested. */
1369 if (reply->transports[0].lower_transport != lower_transport) {
1370 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1371 err = AVERROR_INVALIDDATA;
1375 switch(reply->transports[0].lower_transport) {
1376 case RTSP_LOWER_TRANSPORT_TCP:
1377 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1378 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1381 case RTSP_LOWER_TRANSPORT_UDP: {
1382 char url[1024], options[30] = "";
1384 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1385 av_strlcpy(options, "?connect=1", sizeof(options));
1386 /* Use source address if specified */
1387 if (reply->transports[0].source[0]) {
1388 ff_url_join(url, sizeof(url), "rtp", NULL,
1389 reply->transports[0].source,
1390 reply->transports[0].server_port_min, "%s", options);
1392 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1393 reply->transports[0].server_port_min, "%s", options);
1395 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1396 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1397 err = AVERROR_INVALIDDATA;
1400 /* Try to initialize the connection state in a
1401 * potential NAT router by sending dummy packets.
1402 * RTP/RTCP dummy packets are used for RDT, too.
1404 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1406 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1409 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1410 char url[1024], namebuf[50], optbuf[20] = "";
1411 struct sockaddr_storage addr;
1414 if (reply->transports[0].destination.ss_family) {
1415 addr = reply->transports[0].destination;
1416 port = reply->transports[0].port_min;
1417 ttl = reply->transports[0].ttl;
1419 addr = rtsp_st->sdp_ip;
1420 port = rtsp_st->sdp_port;
1421 ttl = rtsp_st->sdp_ttl;
1424 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1425 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1426 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1427 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1428 port, "%s", optbuf);
1429 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1430 &s->interrupt_callback, NULL) < 0) {
1431 err = AVERROR_INVALIDDATA;
1438 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1442 if (rt->nb_rtsp_streams && reply->timeout > 0)
1443 rt->timeout = reply->timeout;
1445 if (rt->server_type == RTSP_SERVER_REAL)
1446 rt->need_subscription = 1;
1451 ff_rtsp_undo_setup(s);
1455 void ff_rtsp_close_connections(AVFormatContext *s)
1457 RTSPState *rt = s->priv_data;
1458 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1459 ffurl_close(rt->rtsp_hd);
1460 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1463 int ff_rtsp_connect(AVFormatContext *s)
1465 RTSPState *rt = s->priv_data;
1466 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1467 int port, err, tcp_fd;
1468 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1469 int lower_transport_mask = 0;
1470 char real_challenge[64] = "";
1471 struct sockaddr_storage peer;
1472 socklen_t peer_len = sizeof(peer);
1474 if (rt->rtp_port_max < rt->rtp_port_min) {
1475 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1476 "than min port %d\n", rt->rtp_port_max,
1478 return AVERROR(EINVAL);
1481 if (!ff_network_init())
1482 return AVERROR(EIO);
1484 if (s->max_delay < 0) /* Not set by the caller */
1485 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1487 rt->control_transport = RTSP_MODE_PLAIN;
1488 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1489 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1490 rt->control_transport = RTSP_MODE_TUNNEL;
1492 /* Only pass through valid flags from here */
1493 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1496 lower_transport_mask = rt->lower_transport_mask;
1497 /* extract hostname and port */
1498 av_url_split(NULL, 0, auth, sizeof(auth),
1499 host, sizeof(host), &port, path, sizeof(path), s->filename);
1501 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1504 port = RTSP_DEFAULT_PORT;
1506 if (!lower_transport_mask)
1507 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1510 /* Only UDP or TCP - UDP multicast isn't supported. */
1511 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1512 (1 << RTSP_LOWER_TRANSPORT_TCP);
1513 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1514 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1515 "only UDP and TCP are supported for output.\n");
1516 err = AVERROR(EINVAL);
1521 /* Construct the URI used in request; this is similar to s->filename,
1522 * but with authentication credentials removed and RTSP specific options
1524 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1525 host, port, "%s", path);
1527 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1528 /* set up initial handshake for tunneling */
1529 char httpname[1024];
1530 char sessioncookie[17];
1533 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1534 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1535 av_get_random_seed(), av_get_random_seed());
1538 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1539 &s->interrupt_callback) < 0) {
1544 /* generate GET headers */
1545 snprintf(headers, sizeof(headers),
1546 "x-sessioncookie: %s\r\n"
1547 "Accept: application/x-rtsp-tunnelled\r\n"
1548 "Pragma: no-cache\r\n"
1549 "Cache-Control: no-cache\r\n",
1551 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1553 /* complete the connection */
1554 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1560 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1561 &s->interrupt_callback) < 0 ) {
1566 /* generate POST headers */
1567 snprintf(headers, sizeof(headers),
1568 "x-sessioncookie: %s\r\n"
1569 "Content-Type: application/x-rtsp-tunnelled\r\n"
1570 "Pragma: no-cache\r\n"
1571 "Cache-Control: no-cache\r\n"
1572 "Content-Length: 32767\r\n"
1573 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1575 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1576 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1578 /* Initialize the authentication state for the POST session. The HTTP
1579 * protocol implementation doesn't properly handle multi-pass
1580 * authentication for POST requests, since it would require one of
1582 * - implementing Expect: 100-continue, which many HTTP servers
1583 * don't support anyway, even less the RTSP servers that do HTTP
1585 * - sending the whole POST data until getting a 401 reply specifying
1586 * what authentication method to use, then resending all that data
1587 * - waiting for potential 401 replies directly after sending the
1588 * POST header (waiting for some unspecified time)
1589 * Therefore, we copy the full auth state, which works for both basic
1590 * and digest. (For digest, we would have to synchronize the nonce
1591 * count variable between the two sessions, if we'd do more requests
1592 * with the original session, though.)
1594 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1596 /* complete the connection */
1597 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1602 /* open the tcp connection */
1603 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1604 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1605 &s->interrupt_callback, NULL) < 0) {
1609 rt->rtsp_hd_out = rt->rtsp_hd;
1613 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1614 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1615 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1616 NULL, 0, NI_NUMERICHOST);
1619 /* request options supported by the server; this also detects server
1621 for (rt->server_type = RTSP_SERVER_RTP;;) {
1623 if (rt->server_type == RTSP_SERVER_REAL)
1626 * The following entries are required for proper
1627 * streaming from a Realmedia server. They are
1628 * interdependent in some way although we currently
1629 * don't quite understand how. Values were copied
1630 * from mplayer SVN r23589.
1631 * ClientChallenge is a 16-byte ID in hex
1632 * CompanyID is a 16-byte ID in base64
1634 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1635 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1636 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1637 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1639 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1640 if (reply->status_code != RTSP_STATUS_OK) {
1641 err = AVERROR_INVALIDDATA;
1645 /* detect server type if not standard-compliant RTP */
1646 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1647 rt->server_type = RTSP_SERVER_REAL;
1649 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1650 rt->server_type = RTSP_SERVER_WMS;
1651 } else if (rt->server_type == RTSP_SERVER_REAL)
1652 strcpy(real_challenge, reply->real_challenge);
1656 if (s->iformat && CONFIG_RTSP_DEMUXER)
1657 err = ff_rtsp_setup_input_streams(s, reply);
1658 else if (CONFIG_RTSP_MUXER)
1659 err = ff_rtsp_setup_output_streams(s, host);
1664 int lower_transport = ff_log2_tab[lower_transport_mask &
1665 ~(lower_transport_mask - 1)];
1667 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1668 rt->server_type == RTSP_SERVER_REAL ?
1669 real_challenge : NULL);
1672 lower_transport_mask &= ~(1 << lower_transport);
1673 if (lower_transport_mask == 0 && err == 1) {
1674 err = AVERROR(EPROTONOSUPPORT);
1679 rt->lower_transport_mask = lower_transport_mask;
1680 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1681 rt->state = RTSP_STATE_IDLE;
1682 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1685 ff_rtsp_close_streams(s);
1686 ff_rtsp_close_connections(s);
1687 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1688 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1689 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1697 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1700 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1701 uint8_t *buf, int buf_size, int64_t wait_end)
1703 RTSPState *rt = s->priv_data;
1704 RTSPStream *rtsp_st;
1705 int n, i, ret, tcp_fd, timeout_cnt = 0;
1707 struct pollfd *p = rt->p;
1708 int *fds = NULL, fdsnum, fdsidx;
1711 if (ff_check_interrupt(&s->interrupt_callback))
1712 return AVERROR_EXIT;
1713 if (wait_end && wait_end - av_gettime() < 0)
1714 return AVERROR(EAGAIN);
1717 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1718 p[max_p].fd = tcp_fd;
1719 p[max_p++].events = POLLIN;
1723 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1724 rtsp_st = rt->rtsp_streams[i];
1725 if (rtsp_st->rtp_handle) {
1726 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1728 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1732 av_log(s, AV_LOG_ERROR,
1733 "Number of fds %d not supported\n", fdsnum);
1734 return AVERROR_INVALIDDATA;
1736 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1737 p[max_p].fd = fds[fdsidx];
1738 p[max_p++].events = POLLIN;
1743 n = poll(p, max_p, POLL_TIMEOUT_MS);
1745 int j = 1 - (tcp_fd == -1);
1747 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1748 rtsp_st = rt->rtsp_streams[i];
1749 if (rtsp_st->rtp_handle) {
1750 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1751 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1753 *prtsp_st = rtsp_st;
1760 #if CONFIG_RTSP_DEMUXER
1761 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1762 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1763 if (rt->state == RTSP_STATE_STREAMING) {
1764 if (!ff_rtsp_parse_streaming_commands(s))
1767 av_log(s, AV_LOG_WARNING,
1768 "Unable to answer to TEARDOWN\n");
1772 RTSPMessageHeader reply;
1773 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1776 /* XXX: parse message */
1777 if (rt->state != RTSP_STATE_STREAMING)
1782 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1783 return AVERROR(ETIMEDOUT);
1784 } else if (n < 0 && errno != EINTR)
1785 return AVERROR(errno);
1789 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1791 RTSPState *rt = s->priv_data;
1793 RTSPStream *rtsp_st, *first_queue_st = NULL;
1794 int64_t wait_end = 0;
1796 if (rt->nb_byes == rt->nb_rtsp_streams)
1799 /* get next frames from the same RTP packet */
1800 if (rt->cur_transport_priv) {
1801 if (rt->transport == RTSP_TRANSPORT_RDT) {
1802 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1803 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1804 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1805 } else if (rt->ts && CONFIG_RTPDEC) {
1806 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1808 rt->recvbuf_pos += ret;
1809 ret = rt->recvbuf_pos < rt->recvbuf_len;
1814 rt->cur_transport_priv = NULL;
1816 } else if (ret == 1) {
1819 rt->cur_transport_priv = NULL;
1822 if (rt->transport == RTSP_TRANSPORT_RTP) {
1824 int64_t first_queue_time = 0;
1825 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1826 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1830 queue_time = ff_rtp_queued_packet_time(rtpctx);
1831 if (queue_time && (queue_time - first_queue_time < 0 ||
1832 !first_queue_time)) {
1833 first_queue_time = queue_time;
1834 first_queue_st = rt->rtsp_streams[i];
1837 if (first_queue_time)
1838 wait_end = first_queue_time + s->max_delay;
1841 /* read next RTP packet */
1844 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1846 return AVERROR(ENOMEM);
1849 switch(rt->lower_transport) {
1851 #if CONFIG_RTSP_DEMUXER
1852 case RTSP_LOWER_TRANSPORT_TCP:
1853 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1856 case RTSP_LOWER_TRANSPORT_UDP:
1857 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1858 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1859 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1860 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1863 if (len == AVERROR(EAGAIN) && first_queue_st &&
1864 rt->transport == RTSP_TRANSPORT_RTP) {
1865 rtsp_st = first_queue_st;
1866 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1873 if (rt->transport == RTSP_TRANSPORT_RDT) {
1874 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1875 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1876 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1878 /* Either bad packet, or a RTCP packet. Check if the
1879 * first_rtcp_ntp_time field was initialized. */
1880 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1881 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1882 /* first_rtcp_ntp_time has been initialized for this stream,
1883 * copy the same value to all other uninitialized streams,
1884 * in order to map their timestamp origin to the same ntp time
1887 AVStream *st = NULL;
1888 if (rtsp_st->stream_index >= 0)
1889 st = s->streams[rtsp_st->stream_index];
1890 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1891 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1892 AVStream *st2 = NULL;
1893 if (rt->rtsp_streams[i]->stream_index >= 0)
1894 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1895 if (rtpctx2 && st && st2 &&
1896 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1897 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1898 rtpctx2->rtcp_ts_offset = av_rescale_q(
1899 rtpctx->rtcp_ts_offset, st->time_base,
1904 if (ret == -RTCP_BYE) {
1907 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1908 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1910 if (rt->nb_byes == rt->nb_rtsp_streams)
1914 } else if (rt->ts && CONFIG_RTPDEC) {
1915 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1918 rt->recvbuf_len = len;
1919 rt->recvbuf_pos = ret;
1920 rt->cur_transport_priv = rt->ts;
1927 return AVERROR_INVALIDDATA;
1933 /* more packets may follow, so we save the RTP context */
1934 rt->cur_transport_priv = rtsp_st->transport_priv;
1938 #endif /* CONFIG_RTPDEC */
1940 #if CONFIG_SDP_DEMUXER
1941 static int sdp_probe(AVProbeData *p1)
1943 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1945 /* we look for a line beginning "c=IN IP" */
1946 while (p < p_end && *p != '\0') {
1947 if (p + sizeof("c=IN IP") - 1 < p_end &&
1948 av_strstart(p, "c=IN IP", NULL))
1949 return AVPROBE_SCORE_MAX / 2;
1951 while (p < p_end - 1 && *p != '\n') p++;
1960 static int sdp_read_header(AVFormatContext *s)
1962 RTSPState *rt = s->priv_data;
1963 RTSPStream *rtsp_st;
1968 if (!ff_network_init())
1969 return AVERROR(EIO);
1971 if (s->max_delay < 0) /* Not set by the caller */
1972 s->max_delay = DEFAULT_REORDERING_DELAY;
1974 /* read the whole sdp file */
1975 /* XXX: better loading */
1976 content = av_malloc(SDP_MAX_SIZE);
1977 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1980 return AVERROR_INVALIDDATA;
1982 content[size] ='\0';
1984 err = ff_sdp_parse(s, content);
1988 /* open each RTP stream */
1989 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1991 rtsp_st = rt->rtsp_streams[i];
1993 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1994 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1995 ff_url_join(url, sizeof(url), "rtp", NULL,
1996 namebuf, rtsp_st->sdp_port,
1997 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1999 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2000 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2001 &s->interrupt_callback, NULL) < 0) {
2002 err = AVERROR_INVALIDDATA;
2005 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2010 ff_rtsp_close_streams(s);
2015 static int sdp_read_close(AVFormatContext *s)
2017 ff_rtsp_close_streams(s);
2022 static const AVClass sdp_demuxer_class = {
2023 .class_name = "SDP demuxer",
2024 .item_name = av_default_item_name,
2025 .option = sdp_options,
2026 .version = LIBAVUTIL_VERSION_INT,
2029 AVInputFormat ff_sdp_demuxer = {
2031 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2032 .priv_data_size = sizeof(RTSPState),
2033 .read_probe = sdp_probe,
2034 .read_header = sdp_read_header,
2035 .read_packet = ff_rtsp_fetch_packet,
2036 .read_close = sdp_read_close,
2037 .priv_class = &sdp_demuxer_class,
2039 #endif /* CONFIG_SDP_DEMUXER */
2041 #if CONFIG_RTP_DEMUXER
2042 static int rtp_probe(AVProbeData *p)
2044 if (av_strstart(p->filename, "rtp:", NULL))
2045 return AVPROBE_SCORE_MAX;
2049 static int rtp_read_header(AVFormatContext *s)
2051 uint8_t recvbuf[1500];
2052 char host[500], sdp[500];
2054 URLContext* in = NULL;
2056 AVCodecContext codec = { 0 };
2057 struct sockaddr_storage addr;
2059 socklen_t addrlen = sizeof(addr);
2060 RTSPState *rt = s->priv_data;
2062 if (!ff_network_init())
2063 return AVERROR(EIO);
2065 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2066 &s->interrupt_callback, NULL);
2071 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2072 if (ret == AVERROR(EAGAIN))
2077 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2081 if ((recvbuf[0] & 0xc0) != 0x80) {
2082 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2087 if (RTP_PT_IS_RTCP(recvbuf[1]))
2090 payload_type = recvbuf[1] & 0x7f;
2093 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2097 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2098 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2099 "without an SDP file describing it\n",
2103 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2104 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2105 "properly you need an SDP file "
2109 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2110 NULL, 0, s->filename);
2112 snprintf(sdp, sizeof(sdp),
2113 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2114 addr.ss_family == AF_INET ? 4 : 6, host,
2115 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2116 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2117 port, payload_type);
2118 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2120 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2123 /* sdp_read_header initializes this again */
2126 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2128 ret = sdp_read_header(s);
2139 static const AVClass rtp_demuxer_class = {
2140 .class_name = "RTP demuxer",
2141 .item_name = av_default_item_name,
2142 .option = rtp_options,
2143 .version = LIBAVUTIL_VERSION_INT,
2146 AVInputFormat ff_rtp_demuxer = {
2148 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2149 .priv_data_size = sizeof(RTSPState),
2150 .read_probe = rtp_probe,
2151 .read_header = rtp_read_header,
2152 .read_packet = ff_rtsp_fetch_packet,
2153 .read_close = sdp_read_close,
2154 .flags = AVFMT_NOFILE,
2155 .priv_class = &rtp_demuxer_class,
2157 #endif /* CONFIG_RTP_DEMUXER */