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/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
29 #include <sys/select.h>
34 #include "os_support.h"
39 #include "rtpdec_asf.h"
40 #include "rtpdec_vorbis.h"
43 //#define DEBUG_RTP_TCP
45 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
46 int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
49 #define SPACE_CHARS " \t\r\n"
50 /* we use memchr() instead of strchr() here because strchr() will return
51 * the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
52 #define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
53 static void skip_spaces(const char **pp)
57 while (redir_isspace(*p))
62 static void get_word_until_chars(char *buf, int buf_size,
63 const char *sep, const char **pp)
71 while (!strchr(sep, *p) && *p != '\0') {
72 if ((q - buf) < buf_size - 1)
81 static void get_word_sep(char *buf, int buf_size, const char *sep,
84 if (**pp == '/') (*pp)++;
85 get_word_until_chars(buf, buf_size, sep, pp);
88 static void get_word(char *buf, int buf_size, const char **pp)
90 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
93 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
94 static int sdp_parse_rtpmap(AVFormatContext *s,
95 AVCodecContext *codec, RTSPStream *rtsp_st,
96 int payload_type, const char *p)
103 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
104 * see if we can handle this kind of payload.
105 * The space should normally not be there but some Real streams or
106 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
107 * have a trailing space. */
108 get_word_sep(buf, sizeof(buf), "/ ", &p);
109 if (payload_type >= RTP_PT_PRIVATE) {
110 RTPDynamicProtocolHandler *handler;
111 for (handler = RTPFirstDynamicPayloadHandler;
112 handler; handler = handler->next) {
113 if (!strcasecmp(buf, handler->enc_name) &&
114 codec->codec_type == handler->codec_type) {
115 codec->codec_id = handler->codec_id;
116 rtsp_st->dynamic_handler = handler;
118 rtsp_st->dynamic_protocol_context = handler->open();
123 /* We are in a standard case
124 * (from http://www.iana.org/assignments/rtp-parameters). */
125 /* search into AVRtpPayloadTypes[] */
126 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
129 c = avcodec_find_decoder(codec->codec_id);
135 get_word_sep(buf, sizeof(buf), "/", &p);
137 switch (codec->codec_type) {
138 case CODEC_TYPE_AUDIO:
139 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
140 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
141 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
143 codec->sample_rate = i;
144 get_word_sep(buf, sizeof(buf), "/", &p);
148 // TODO: there is a bug here; if it is a mono stream, and
149 // less than 22000Hz, faad upconverts to stereo and twice
150 // the frequency. No problem, but the sample rate is being
151 // set here by the sdp line. Patch on its way. (rdm)
153 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
155 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
158 case CODEC_TYPE_VIDEO:
159 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
167 /* return the length and optionally the data */
168 static int hex_to_data(uint8_t *data, const char *p)
178 c = toupper((unsigned char) *p++);
179 if (c >= '0' && c <= '9')
181 else if (c >= 'A' && c <= 'F')
196 static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
197 char *attr, char *value)
199 switch (codec->codec_id) {
202 if (!strcmp(attr, "config")) {
203 /* decode the hexa encoded parameter */
204 int len = hex_to_data(NULL, value);
205 if (codec->extradata)
206 av_free(codec->extradata);
207 codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
208 if (!codec->extradata)
210 codec->extradata_size = len;
211 hex_to_data(codec->extradata, value);
214 case CODEC_ID_VORBIS:
215 ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
229 /* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
230 #define ATTR_NAME_TYPE_INT 0
231 #define ATTR_NAME_TYPE_STR 1
232 static const AttrNameMap attr_names[]=
234 { "SizeLength", ATTR_NAME_TYPE_INT,
235 offsetof(RTPPayloadData, sizelength) },
236 { "IndexLength", ATTR_NAME_TYPE_INT,
237 offsetof(RTPPayloadData, indexlength) },
238 { "IndexDeltaLength", ATTR_NAME_TYPE_INT,
239 offsetof(RTPPayloadData, indexdeltalength) },
240 { "profile-level-id", ATTR_NAME_TYPE_INT,
241 offsetof(RTPPayloadData, profile_level_id) },
242 { "StreamType", ATTR_NAME_TYPE_INT,
243 offsetof(RTPPayloadData, streamtype) },
244 { "mode", ATTR_NAME_TYPE_STR,
245 offsetof(RTPPayloadData, mode) },
249 /* parse the attribute line from the fmtp a line of an sdp resonse. This
250 * is broken out as a function because it is used in rtp_h264.c, which is
252 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
253 char *value, int value_size)
257 get_word_sep(attr, attr_size, "=", p);
260 get_word_sep(value, value_size, ";", p);
268 /* parse a SDP line and save stream attributes */
269 static void sdp_parse_fmtp(AVStream *st, const char *p)
272 /* Vorbis setup headers can be up to 12KB and are sent base64
273 * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
276 RTSPStream *rtsp_st = st->priv_data;
277 AVCodecContext *codec = st->codec;
278 RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
280 /* loop on each attribute */
281 while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
282 value, sizeof(value))) {
283 /* grab the codec extra_data from the config parameter of the fmtp
285 sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
287 /* Looking for a known attribute */
288 for (i = 0; attr_names[i].str; ++i) {
289 if (!strcasecmp(attr, attr_names[i].str)) {
290 if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
291 *(int *)((char *)rtp_payload_data +
292 attr_names[i].offset) = atoi(value);
293 } else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
294 *(char **)((char *)rtp_payload_data +
295 attr_names[i].offset) = av_strdup(value);
301 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
303 * Used for seeking in the rtp stream.
305 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
310 if (!av_stristart(p, "npt=", &p))
313 *start = AV_NOPTS_VALUE;
314 *end = AV_NOPTS_VALUE;
316 get_word_sep(buf, sizeof(buf), "-", &p);
317 *start = parse_date(buf, 1);
320 get_word_sep(buf, sizeof(buf), "-", &p);
321 *end = parse_date(buf, 1);
323 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
324 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
327 typedef struct SDPParseState {
329 struct in_addr default_ip;
331 int skip_media; ///< set if an unknown m= line occurs
334 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
335 int letter, const char *buf)
337 RTSPState *rt = s->priv_data;
338 char buf1[64], st_type[64];
340 enum CodecType codec_type;
344 struct in_addr sdp_ip;
347 dprintf(s, "sdp: %c='%s'\n", letter, buf);
350 if (s1->skip_media && letter != 'm')
354 get_word(buf1, sizeof(buf1), &p);
355 if (strcmp(buf1, "IN") != 0)
357 get_word(buf1, sizeof(buf1), &p);
358 if (strcmp(buf1, "IP4") != 0)
360 get_word_sep(buf1, sizeof(buf1), "/", &p);
361 if (ff_inet_aton(buf1, &sdp_ip) == 0)
366 get_word_sep(buf1, sizeof(buf1), "/", &p);
369 if (s->nb_streams == 0) {
370 s1->default_ip = sdp_ip;
371 s1->default_ttl = ttl;
373 st = s->streams[s->nb_streams - 1];
374 rtsp_st = st->priv_data;
375 rtsp_st->sdp_ip = sdp_ip;
376 rtsp_st->sdp_ttl = ttl;
380 av_metadata_set(&s->metadata, "title", p);
383 if (s->nb_streams == 0) {
384 av_metadata_set(&s->metadata, "comment", p);
391 get_word(st_type, sizeof(st_type), &p);
392 if (!strcmp(st_type, "audio")) {
393 codec_type = CODEC_TYPE_AUDIO;
394 } else if (!strcmp(st_type, "video")) {
395 codec_type = CODEC_TYPE_VIDEO;
396 } else if (!strcmp(st_type, "application")) {
397 codec_type = CODEC_TYPE_DATA;
402 rtsp_st = av_mallocz(sizeof(RTSPStream));
405 rtsp_st->stream_index = -1;
406 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
408 rtsp_st->sdp_ip = s1->default_ip;
409 rtsp_st->sdp_ttl = s1->default_ttl;
411 get_word(buf1, sizeof(buf1), &p); /* port */
412 rtsp_st->sdp_port = atoi(buf1);
414 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
416 /* XXX: handle list of formats */
417 get_word(buf1, sizeof(buf1), &p); /* format list */
418 rtsp_st->sdp_payload_type = atoi(buf1);
420 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
421 /* no corresponding stream */
423 st = av_new_stream(s, 0);
426 st->priv_data = rtsp_st;
427 rtsp_st->stream_index = st->index;
428 st->codec->codec_type = codec_type;
429 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
430 /* if standard payload type, we can find the codec right now */
431 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
434 /* put a default control url */
435 av_strlcpy(rtsp_st->control_url, rt->control_uri,
436 sizeof(rtsp_st->control_url));
439 if (av_strstart(p, "control:", &p)) {
440 if (s->nb_streams == 0) {
441 if (!strncmp(p, "rtsp://", 7))
442 av_strlcpy(rt->control_uri, p,
443 sizeof(rt->control_uri));
446 /* get the control url */
447 st = s->streams[s->nb_streams - 1];
448 rtsp_st = st->priv_data;
450 /* XXX: may need to add full url resolution */
451 ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
453 if (proto[0] == '\0') {
454 /* relative control URL */
455 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
456 av_strlcat(rtsp_st->control_url, "/",
457 sizeof(rtsp_st->control_url));
458 av_strlcat(rtsp_st->control_url, p,
459 sizeof(rtsp_st->control_url));
461 av_strlcpy(rtsp_st->control_url, p,
462 sizeof(rtsp_st->control_url));
464 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
465 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
466 get_word(buf1, sizeof(buf1), &p);
467 payload_type = atoi(buf1);
468 st = s->streams[s->nb_streams - 1];
469 rtsp_st = st->priv_data;
470 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
471 } else if (av_strstart(p, "fmtp:", &p)) {
472 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
473 get_word(buf1, sizeof(buf1), &p);
474 payload_type = atoi(buf1);
475 for (i = 0; i < s->nb_streams; i++) {
477 rtsp_st = st->priv_data;
478 if (rtsp_st->sdp_payload_type == payload_type) {
479 if (!(rtsp_st->dynamic_handler &&
480 rtsp_st->dynamic_handler->parse_sdp_a_line &&
481 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
482 i, rtsp_st->dynamic_protocol_context, buf)))
483 sdp_parse_fmtp(st, p);
486 } else if (av_strstart(p, "framesize:", &p)) {
487 // let dynamic protocol handlers have a stab at the line.
488 get_word(buf1, sizeof(buf1), &p);
489 payload_type = atoi(buf1);
490 for (i = 0; i < s->nb_streams; i++) {
492 rtsp_st = st->priv_data;
493 if (rtsp_st->sdp_payload_type == payload_type &&
494 rtsp_st->dynamic_handler &&
495 rtsp_st->dynamic_handler->parse_sdp_a_line)
496 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
497 rtsp_st->dynamic_protocol_context, buf);
499 } else if (av_strstart(p, "range:", &p)) {
502 // this is so that seeking on a streamed file can work.
503 rtsp_parse_range_npt(p, &start, &end);
504 s->start_time = start;
505 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
506 s->duration = (end == AV_NOPTS_VALUE) ?
507 AV_NOPTS_VALUE : end - start;
508 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
510 rt->transport = RTSP_TRANSPORT_RDT;
512 if (rt->server_type == RTSP_SERVER_WMS)
513 ff_wms_parse_sdp_a_line(s, p);
514 if (s->nb_streams > 0) {
515 if (rt->server_type == RTSP_SERVER_REAL)
516 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
518 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
519 if (rtsp_st->dynamic_handler &&
520 rtsp_st->dynamic_handler->parse_sdp_a_line)
521 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
523 rtsp_st->dynamic_protocol_context, buf);
530 static int sdp_parse(AVFormatContext *s, const char *content)
534 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
535 * contain long SDP lines containing complete ASF Headers (several
536 * kB) or arrays of MDPR (RM stream descriptor) headers plus
537 * "rulebooks" describing their properties. Therefore, the SDP line
540 * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
542 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
544 memset(s1, 0, sizeof(SDPParseState));
555 /* get the content */
557 while (*p != '\n' && *p != '\r' && *p != '\0') {
558 if ((q - buf) < sizeof(buf) - 1)
563 sdp_parse_line(s, s1, letter, buf);
565 while (*p != '\n' && *p != '\0')
573 /* close and free RTSP streams */
574 void ff_rtsp_close_streams(AVFormatContext *s)
576 RTSPState *rt = s->priv_data;
580 for (i = 0; i < rt->nb_rtsp_streams; i++) {
581 rtsp_st = rt->rtsp_streams[i];
583 if (rtsp_st->transport_priv) {
585 AVFormatContext *rtpctx = rtsp_st->transport_priv;
586 av_write_trailer(rtpctx);
587 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
589 url_close_dyn_buf(rtpctx->pb, &ptr);
592 url_fclose(rtpctx->pb);
594 av_metadata_free(&rtpctx->streams[0]->metadata);
595 av_metadata_free(&rtpctx->metadata);
596 av_free(rtpctx->streams[0]);
598 } else if (rt->transport == RTSP_TRANSPORT_RDT)
599 ff_rdt_parse_close(rtsp_st->transport_priv);
601 rtp_parse_close(rtsp_st->transport_priv);
603 if (rtsp_st->rtp_handle)
604 url_close(rtsp_st->rtp_handle);
605 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
606 rtsp_st->dynamic_handler->close(
607 rtsp_st->dynamic_protocol_context);
610 av_free(rt->rtsp_streams);
612 av_close_input_stream (rt->asf_ctx);
617 static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
620 RTSPState *rt = s->priv_data;
621 AVFormatContext *rtpctx;
623 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
628 /* Allocate an AVFormatContext for each output stream */
629 rtpctx = avformat_alloc_context();
633 rtpctx->oformat = rtp_format;
634 if (!av_new_stream(rtpctx, 0)) {
638 /* Copy the max delay setting; the rtp muxer reads this. */
639 rtpctx->max_delay = s->max_delay;
640 /* Copy other stream parameters. */
641 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
643 /* Set the synchronized start time. */
644 rtpctx->start_time_realtime = rt->start_time;
646 /* Remove the local codec, link to the original codec
647 * context instead, to give the rtp muxer access to
648 * codec parameters. */
649 av_free(rtpctx->streams[0]->codec);
650 rtpctx->streams[0]->codec = st->codec;
653 url_fdopen(&rtpctx->pb, handle);
655 url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
656 ret = av_write_header(rtpctx);
660 url_fclose(rtpctx->pb);
663 url_close_dyn_buf(rtpctx->pb, &ptr);
666 av_free(rtpctx->streams[0]);
671 /* Copy the RTP AVStream timebase back to the original AVStream */
672 st->time_base = rtpctx->streams[0]->time_base;
676 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
678 RTSPState *rt = s->priv_data;
681 /* open the RTP context */
682 if (rtsp_st->stream_index >= 0)
683 st = s->streams[rtsp_st->stream_index];
685 s->ctx_flags |= AVFMTCTX_NOHEADER;
688 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
689 /* Ownage of rtp_handle is passed to the rtp mux context */
690 rtsp_st->rtp_handle = NULL;
691 } else if (rt->transport == RTSP_TRANSPORT_RDT)
692 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
693 rtsp_st->dynamic_protocol_context,
694 rtsp_st->dynamic_handler);
696 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
697 rtsp_st->sdp_payload_type,
698 &rtsp_st->rtp_payload_data);
700 if (!rtsp_st->transport_priv) {
701 return AVERROR(ENOMEM);
702 } else if (rt->transport != RTSP_TRANSPORT_RDT) {
703 if (rtsp_st->dynamic_handler) {
704 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
705 rtsp_st->dynamic_protocol_context,
706 rtsp_st->dynamic_handler);
713 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
714 static int rtsp_probe(AVProbeData *p)
716 if (av_strstart(p->filename, "rtsp:", NULL))
717 return AVPROBE_SCORE_MAX;
721 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
728 v = strtol(p, (char **)&p, 10);
732 v = strtol(p, (char **)&p, 10);
741 /* XXX: only one transport specification is parsed */
742 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
744 char transport_protocol[16];
746 char lower_transport[16];
748 RTSPTransportField *th;
751 reply->nb_transports = 0;
758 th = &reply->transports[reply->nb_transports];
760 get_word_sep(transport_protocol, sizeof(transport_protocol),
762 if (!strcasecmp (transport_protocol, "rtp")) {
763 get_word_sep(profile, sizeof(profile), "/;,", &p);
764 lower_transport[0] = '\0';
765 /* rtp/avp/<protocol> */
767 get_word_sep(lower_transport, sizeof(lower_transport),
770 th->transport = RTSP_TRANSPORT_RTP;
771 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
772 !strcasecmp (transport_protocol, "x-real-rdt")) {
773 /* x-pn-tng/<protocol> */
774 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
776 th->transport = RTSP_TRANSPORT_RDT;
778 if (!strcasecmp(lower_transport, "TCP"))
779 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
781 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
785 /* get each parameter */
786 while (*p != '\0' && *p != ',') {
787 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
788 if (!strcmp(parameter, "port")) {
791 rtsp_parse_range(&th->port_min, &th->port_max, &p);
793 } else if (!strcmp(parameter, "client_port")) {
796 rtsp_parse_range(&th->client_port_min,
797 &th->client_port_max, &p);
799 } else if (!strcmp(parameter, "server_port")) {
802 rtsp_parse_range(&th->server_port_min,
803 &th->server_port_max, &p);
805 } else if (!strcmp(parameter, "interleaved")) {
808 rtsp_parse_range(&th->interleaved_min,
809 &th->interleaved_max, &p);
811 } else if (!strcmp(parameter, "multicast")) {
812 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
813 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
814 } else if (!strcmp(parameter, "ttl")) {
817 th->ttl = strtol(p, (char **)&p, 10);
819 } else if (!strcmp(parameter, "destination")) {
820 struct in_addr ipaddr;
824 get_word_sep(buf, sizeof(buf), ";,", &p);
825 if (ff_inet_aton(buf, &ipaddr))
826 th->destination = ntohl(ipaddr.s_addr);
829 while (*p != ';' && *p != '\0' && *p != ',')
837 reply->nb_transports++;
841 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
842 HTTPAuthState *auth_state)
846 /* NOTE: we do case independent match for broken servers */
848 if (av_stristart(p, "Session:", &p)) {
850 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
851 if (av_stristart(p, ";timeout=", &p) &&
852 (t = strtol(p, NULL, 10)) > 0) {
855 } else if (av_stristart(p, "Content-Length:", &p)) {
856 reply->content_length = strtol(p, NULL, 10);
857 } else if (av_stristart(p, "Transport:", &p)) {
858 rtsp_parse_transport(reply, p);
859 } else if (av_stristart(p, "CSeq:", &p)) {
860 reply->seq = strtol(p, NULL, 10);
861 } else if (av_stristart(p, "Range:", &p)) {
862 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
863 } else if (av_stristart(p, "RealChallenge1:", &p)) {
865 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
866 } else if (av_stristart(p, "Server:", &p)) {
868 av_strlcpy(reply->server, p, sizeof(reply->server));
869 } else if (av_stristart(p, "Notice:", &p) ||
870 av_stristart(p, "X-Notice:", &p)) {
871 reply->notice = strtol(p, NULL, 10);
872 } else if (av_stristart(p, "Location:", &p)) {
874 av_strlcpy(reply->location, p , sizeof(reply->location));
875 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
877 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
878 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
880 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
884 /* skip a RTP/TCP interleaved packet */
885 void ff_rtsp_skip_packet(AVFormatContext *s)
887 RTSPState *rt = s->priv_data;
891 ret = url_read_complete(rt->rtsp_hd, buf, 3);
894 len = AV_RB16(buf + 1);
896 dprintf(s, "skipping RTP packet len=%d\n", len);
901 if (len1 > sizeof(buf))
903 ret = url_read_complete(rt->rtsp_hd, buf, len1);
910 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
911 unsigned char **content_ptr,
912 int return_on_interleaved_data)
914 RTSPState *rt = s->priv_data;
915 char buf[4096], buf1[1024], *q;
918 int ret, content_length, line_count = 0;
919 unsigned char *content = NULL;
921 memset(reply, 0, sizeof(*reply));
923 /* parse reply (XXX: use buffers) */
924 rt->last_reply[0] = '\0';
928 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
930 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
937 /* XXX: only parse it if first char on line ? */
938 if (return_on_interleaved_data) {
941 ff_rtsp_skip_packet(s);
942 } else if (ch != '\r') {
943 if ((q - buf) < sizeof(buf) - 1)
949 dprintf(s, "line='%s'\n", buf);
951 /* test if last line */
955 if (line_count == 0) {
957 get_word(buf1, sizeof(buf1), &p);
958 get_word(buf1, sizeof(buf1), &p);
959 reply->status_code = atoi(buf1);
961 ff_rtsp_parse_line(reply, p, &rt->auth_state);
962 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
963 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
968 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
969 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
971 content_length = reply->content_length;
972 if (content_length > 0) {
973 /* leave some room for a trailing '\0' (useful for simple parsing) */
974 content = av_malloc(content_length + 1);
975 (void)url_read_complete(rt->rtsp_hd, content, content_length);
976 content[content_length] = '\0';
979 *content_ptr = content;
983 if (rt->seq != reply->seq) {
984 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
985 rt->seq, reply->seq);
989 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
990 reply->notice == 2104 /* Start-of-Stream Reached */ ||
991 reply->notice == 2306 /* Continuous Feed Terminated */) {
992 rt->state = RTSP_STATE_IDLE;
993 } else if (reply->notice >= 4400 && reply->notice < 5500) {
994 return AVERROR(EIO); /* data or server error */
995 } else if (reply->notice == 2401 /* Ticket Expired */ ||
996 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
997 return AVERROR(EPERM);
1002 void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1003 const char *method, const char *url,
1004 const char *headers,
1005 const unsigned char *send_content,
1006 int send_content_length)
1008 RTSPState *rt = s->priv_data;
1009 char buf[4096], buf1[1024];
1012 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1014 av_strlcat(buf, headers, sizeof(buf));
1015 snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
1016 av_strlcat(buf, buf1, sizeof(buf));
1017 if (rt->session_id[0] != '\0' && (!headers ||
1018 !strstr(headers, "\nIf-Match:"))) {
1019 snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
1020 av_strlcat(buf, buf1, sizeof(buf));
1023 char *str = ff_http_auth_create_response(&rt->auth_state,
1024 rt->auth, url, method);
1026 av_strlcat(buf, str, sizeof(buf));
1029 if (send_content_length > 0 && send_content)
1030 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1031 av_strlcat(buf, "\r\n", sizeof(buf));
1033 dprintf(s, "Sending:\n%s--\n", buf);
1035 url_write(rt->rtsp_hd, buf, strlen(buf));
1036 if (send_content_length > 0 && send_content)
1037 url_write(rt->rtsp_hd, send_content, send_content_length);
1038 rt->last_cmd_time = av_gettime();
1041 void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1042 const char *url, const char *headers)
1044 ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1047 void ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1048 const char *headers, RTSPMessageHeader *reply,
1049 unsigned char **content_ptr)
1051 ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1052 content_ptr, NULL, 0);
1055 void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1056 const char *method, const char *url,
1058 RTSPMessageHeader *reply,
1059 unsigned char **content_ptr,
1060 const unsigned char *send_content,
1061 int send_content_length)
1063 ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1064 send_content, send_content_length);
1066 ff_rtsp_read_reply(s, reply, content_ptr, 0);
1070 * @returns 0 on success, <0 on error, 1 if protocol is unavailable.
1072 static int make_setup_request(AVFormatContext *s, const char *host, int port,
1073 int lower_transport, const char *real_challenge)
1075 RTSPState *rt = s->priv_data;
1076 int rtx, j, i, err, interleave = 0;
1077 RTSPStream *rtsp_st;
1078 RTSPMessageHeader reply1, *reply = &reply1;
1080 const char *trans_pref;
1082 if (rt->transport == RTSP_TRANSPORT_RDT)
1083 trans_pref = "x-pn-tng";
1085 trans_pref = "RTP/AVP";
1087 /* default timeout: 1 minute */
1090 /* for each stream, make the setup request */
1091 /* XXX: we assume the same server is used for the control of each
1094 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1095 char transport[2048];
1098 * WMS serves all UDP data over a single connection, the RTX, which
1099 * isn't necessarily the first in the SDP but has to be the first
1100 * to be set up, else the second/third SETUP will fail with a 461.
1102 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1103 rt->server_type == RTSP_SERVER_WMS) {
1106 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1107 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1109 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1113 if (rtx == rt->nb_rtsp_streams)
1114 return -1; /* no RTX found */
1115 rtsp_st = rt->rtsp_streams[rtx];
1117 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1119 rtsp_st = rt->rtsp_streams[i];
1122 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1125 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1126 port = reply->transports[0].client_port_min;
1130 /* first try in specified port range */
1131 if (RTSP_RTP_PORT_MIN != 0) {
1132 while (j <= RTSP_RTP_PORT_MAX) {
1133 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1134 "?localport=%d", j);
1135 /* we will use two ports per rtp stream (rtp and rtcp) */
1137 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1143 /* then try on any port */
1144 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1145 err = AVERROR_INVALIDDATA;
1151 port = rtp_get_local_port(rtsp_st->rtp_handle);
1153 snprintf(transport, sizeof(transport) - 1,
1154 "%s/UDP;", trans_pref);
1155 if (rt->server_type != RTSP_SERVER_REAL)
1156 av_strlcat(transport, "unicast;", sizeof(transport));
1157 av_strlcatf(transport, sizeof(transport),
1158 "client_port=%d", port);
1159 if (rt->transport == RTSP_TRANSPORT_RTP &&
1160 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1161 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1165 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1166 /** For WMS streams, the application streams are only used for
1167 * UDP. When trying to set it up for TCP streams, the server
1168 * will return an error. Therefore, we skip those streams. */
1169 if (rt->server_type == RTSP_SERVER_WMS &&
1170 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1173 snprintf(transport, sizeof(transport) - 1,
1174 "%s/TCP;", trans_pref);
1175 if (rt->server_type == RTSP_SERVER_WMS)
1176 av_strlcat(transport, "unicast;", sizeof(transport));
1177 av_strlcatf(transport, sizeof(transport),
1178 "interleaved=%d-%d",
1179 interleave, interleave + 1);
1183 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1184 snprintf(transport, sizeof(transport) - 1,
1185 "%s/UDP;multicast", trans_pref);
1188 av_strlcat(transport, ";mode=receive", sizeof(transport));
1189 } else if (rt->server_type == RTSP_SERVER_REAL ||
1190 rt->server_type == RTSP_SERVER_WMS)
1191 av_strlcat(transport, ";mode=play", sizeof(transport));
1192 snprintf(cmd, sizeof(cmd),
1193 "Transport: %s\r\n",
1195 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1196 char real_res[41], real_csum[9];
1197 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1199 av_strlcatf(cmd, sizeof(cmd),
1201 "RealChallenge2: %s, sd=%s\r\n",
1202 rt->session_id, real_res, real_csum);
1204 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1205 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1208 } else if (reply->status_code != RTSP_STATUS_OK ||
1209 reply->nb_transports != 1) {
1210 err = AVERROR_INVALIDDATA;
1214 /* XXX: same protocol for all streams is required */
1216 if (reply->transports[0].lower_transport != rt->lower_transport ||
1217 reply->transports[0].transport != rt->transport) {
1218 err = AVERROR_INVALIDDATA;
1222 rt->lower_transport = reply->transports[0].lower_transport;
1223 rt->transport = reply->transports[0].transport;
1226 /* close RTP connection if not choosen */
1227 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1228 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1229 url_close(rtsp_st->rtp_handle);
1230 rtsp_st->rtp_handle = NULL;
1233 switch(reply->transports[0].lower_transport) {
1234 case RTSP_LOWER_TRANSPORT_TCP:
1235 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1236 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1239 case RTSP_LOWER_TRANSPORT_UDP: {
1242 /* XXX: also use address if specified */
1243 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1244 reply->transports[0].server_port_min, NULL);
1245 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1246 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1247 err = AVERROR_INVALIDDATA;
1250 /* Try to initialize the connection state in a
1251 * potential NAT router by sending dummy packets.
1252 * RTP/RTCP dummy packets are used for RDT, too.
1254 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1255 rtp_send_punch_packets(rtsp_st->rtp_handle);
1258 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1263 if (reply->transports[0].destination) {
1264 in.s_addr = htonl(reply->transports[0].destination);
1265 port = reply->transports[0].port_min;
1266 ttl = reply->transports[0].ttl;
1268 in = rtsp_st->sdp_ip;
1269 port = rtsp_st->sdp_port;
1270 ttl = rtsp_st->sdp_ttl;
1272 ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
1273 port, "?ttl=%d", ttl);
1274 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1275 err = AVERROR_INVALIDDATA;
1282 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1286 if (reply->timeout > 0)
1287 rt->timeout = reply->timeout;
1289 if (rt->server_type == RTSP_SERVER_REAL)
1290 rt->need_subscription = 1;
1295 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1296 if (rt->rtsp_streams[i]->rtp_handle) {
1297 url_close(rt->rtsp_streams[i]->rtp_handle);
1298 rt->rtsp_streams[i]->rtp_handle = NULL;
1304 static int rtsp_read_play(AVFormatContext *s)
1306 RTSPState *rt = s->priv_data;
1307 RTSPMessageHeader reply1, *reply = &reply1;
1310 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1312 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1313 if (rt->state == RTSP_STATE_PAUSED) {
1316 snprintf(cmd, sizeof(cmd),
1317 "Range: npt=%0.3f-\r\n",
1318 (double)rt->seek_timestamp / AV_TIME_BASE);
1320 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1321 if (reply->status_code != RTSP_STATUS_OK) {
1325 rt->state = RTSP_STATE_STREAMING;
1329 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1331 RTSPState *rt = s->priv_data;
1333 unsigned char *content = NULL;
1336 /* describe the stream */
1337 snprintf(cmd, sizeof(cmd),
1338 "Accept: application/sdp\r\n");
1339 if (rt->server_type == RTSP_SERVER_REAL) {
1341 * The Require: attribute is needed for proper streaming from
1342 * Realmedia servers.
1345 "Require: com.real.retain-entity-for-setup\r\n",
1348 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1350 return AVERROR_INVALIDDATA;
1351 if (reply->status_code != RTSP_STATUS_OK) {
1353 return AVERROR_INVALIDDATA;
1356 /* now we got the SDP description, we parse it */
1357 ret = sdp_parse(s, (const char *)content);
1360 return AVERROR_INVALIDDATA;
1365 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1367 RTSPState *rt = s->priv_data;
1368 RTSPMessageHeader reply1, *reply = &reply1;
1371 AVFormatContext sdp_ctx, *ctx_array[1];
1373 rt->start_time = av_gettime();
1375 /* Announce the stream */
1376 sdp = av_mallocz(8192);
1378 return AVERROR(ENOMEM);
1379 /* We create the SDP based on the RTSP AVFormatContext where we
1380 * aren't allowed to change the filename field. (We create the SDP
1381 * based on the RTSP context since the contexts for the RTP streams
1382 * don't exist yet.) In order to specify a custom URL with the actual
1383 * peer IP instead of the originally specified hostname, we create
1384 * a temporary copy of the AVFormatContext, where the custom URL is set.
1386 * FIXME: Create the SDP without copying the AVFormatContext.
1387 * This either requires setting up the RTP stream AVFormatContexts
1388 * already here (complicating things immensely) or getting a more
1389 * flexible SDP creation interface.
1392 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1393 "rtsp", NULL, addr, -1, NULL);
1394 ctx_array[0] = &sdp_ctx;
1395 if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
1397 return AVERROR_INVALIDDATA;
1399 av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1400 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1401 "Content-Type: application/sdp\r\n",
1402 reply, NULL, sdp, strlen(sdp));
1404 if (reply->status_code != RTSP_STATUS_OK)
1405 return AVERROR_INVALIDDATA;
1407 /* Set up the RTSPStreams for each AVStream */
1408 for (i = 0; i < s->nb_streams; i++) {
1409 RTSPStream *rtsp_st;
1410 AVStream *st = s->streams[i];
1412 rtsp_st = av_mallocz(sizeof(RTSPStream));
1414 return AVERROR(ENOMEM);
1415 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1417 st->priv_data = rtsp_st;
1418 rtsp_st->stream_index = i;
1420 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1421 /* Note, this must match the relative uri set in the sdp content */
1422 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1429 int ff_rtsp_connect(AVFormatContext *s)
1431 RTSPState *rt = s->priv_data;
1432 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1433 char *option_list, *option, *filename;
1434 URLContext *rtsp_hd;
1435 int port, err, tcp_fd;
1436 RTSPMessageHeader reply1, *reply = &reply1;
1437 int lower_transport_mask = 0;
1438 char real_challenge[64];
1439 struct sockaddr_storage peer;
1440 socklen_t peer_len = sizeof(peer);
1442 if (!ff_network_init())
1443 return AVERROR(EIO);
1445 /* extract hostname and port */
1446 ff_url_split(NULL, 0, auth, sizeof(auth),
1447 host, sizeof(host), &port, path, sizeof(path), s->filename);
1449 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1450 rt->auth_state.auth_type = HTTP_AUTH_BASIC;
1453 port = RTSP_DEFAULT_PORT;
1455 /* search for options */
1456 option_list = strrchr(path, '?');
1458 /* Strip out the RTSP specific options, write out the rest of
1459 * the options back into the same string. */
1460 filename = option_list;
1461 while (option_list) {
1462 /* move the option pointer */
1463 option = ++option_list;
1464 option_list = strchr(option_list, '&');
1468 /* handle the options */
1469 if (!strcmp(option, "udp")) {
1470 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1471 } else if (!strcmp(option, "multicast")) {
1472 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1473 } else if (!strcmp(option, "tcp")) {
1474 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1476 /* Write options back into the buffer, using memmove instead
1477 * of strcpy since the strings may overlap. */
1478 int len = strlen(option);
1479 memmove(++filename, option, len);
1481 if (option_list) *filename = '&';
1487 if (!lower_transport_mask)
1488 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1491 /* Only UDP or TCP - UDP multicast isn't supported. */
1492 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1493 (1 << RTSP_LOWER_TRANSPORT_TCP);
1494 if (!lower_transport_mask) {
1495 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1496 "only UDP and TCP are supported for output.\n");
1497 err = AVERROR(EINVAL);
1502 /* open the tcp connexion */
1503 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1504 if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
1508 rt->rtsp_hd = rtsp_hd;
1511 tcp_fd = url_get_file_handle(rtsp_hd);
1512 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1513 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1514 NULL, 0, NI_NUMERICHOST);
1517 /* Construct the URI used in request; this is similar to s->filename,
1518 * but with authentication credentials removed and RTSP specific options
1520 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1521 host, port, "%s", path);
1522 /* request options supported by the server; this also detects server
1524 for (rt->server_type = RTSP_SERVER_RTP;;) {
1526 if (rt->server_type == RTSP_SERVER_REAL)
1529 * The following entries are required for proper
1530 * streaming from a Realmedia server. They are
1531 * interdependent in some way although we currently
1532 * don't quite understand how. Values were copied
1533 * from mplayer SVN r23589.
1534 * @param CompanyID is a 16-byte ID in base64
1535 * @param ClientChallenge is a 16-byte ID in hex
1537 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1538 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1539 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1540 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1542 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1543 if (reply->status_code != RTSP_STATUS_OK) {
1544 err = AVERROR_INVALIDDATA;
1548 /* detect server type if not standard-compliant RTP */
1549 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1550 rt->server_type = RTSP_SERVER_REAL;
1552 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1553 rt->server_type = RTSP_SERVER_WMS;
1554 } else if (rt->server_type == RTSP_SERVER_REAL)
1555 strcpy(real_challenge, reply->real_challenge);
1560 err = rtsp_setup_input_streams(s, reply);
1562 err = rtsp_setup_output_streams(s, host);
1567 int lower_transport = ff_log2_tab[lower_transport_mask &
1568 ~(lower_transport_mask - 1)];
1570 err = make_setup_request(s, host, port, lower_transport,
1571 rt->server_type == RTSP_SERVER_REAL ?
1572 real_challenge : NULL);
1575 lower_transport_mask &= ~(1 << lower_transport);
1576 if (lower_transport_mask == 0 && err == 1) {
1577 err = AVERROR(FF_NETERROR(EPROTONOSUPPORT));
1582 rt->state = RTSP_STATE_IDLE;
1583 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1586 ff_rtsp_close_streams(s);
1587 url_close(rt->rtsp_hd);
1588 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1589 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1590 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1600 #if CONFIG_RTSP_DEMUXER
1601 static int rtsp_read_header(AVFormatContext *s,
1602 AVFormatParameters *ap)
1604 RTSPState *rt = s->priv_data;
1607 ret = ff_rtsp_connect(s);
1611 if (ap->initial_pause) {
1612 /* do not start immediately */
1614 if (rtsp_read_play(s) < 0) {
1615 ff_rtsp_close_streams(s);
1616 url_close(rt->rtsp_hd);
1617 return AVERROR_INVALIDDATA;
1624 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1625 uint8_t *buf, int buf_size)
1627 RTSPState *rt = s->priv_data;
1628 RTSPStream *rtsp_st;
1630 int fd, fd_max, n, i, ret, tcp_fd;
1634 if (url_interrupt_cb())
1635 return AVERROR(EINTR);
1638 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1639 FD_SET(tcp_fd, &rfds);
1644 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1645 rtsp_st = rt->rtsp_streams[i];
1646 if (rtsp_st->rtp_handle) {
1647 /* currently, we cannot probe RTCP handle because of
1648 * blocking restrictions */
1649 fd = url_get_file_handle(rtsp_st->rtp_handle);
1656 tv.tv_usec = 100 * 1000;
1657 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1659 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1660 rtsp_st = rt->rtsp_streams[i];
1661 if (rtsp_st->rtp_handle) {
1662 fd = url_get_file_handle(rtsp_st->rtp_handle);
1663 if (FD_ISSET(fd, &rfds)) {
1664 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1666 *prtsp_st = rtsp_st;
1672 #if CONFIG_RTSP_DEMUXER
1673 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1674 RTSPMessageHeader reply;
1676 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1679 /* XXX: parse message */
1680 if (rt->state != RTSP_STATE_STREAMING)
1688 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1689 uint8_t *buf, int buf_size)
1691 RTSPState *rt = s->priv_data;
1692 int id, len, i, ret;
1693 RTSPStream *rtsp_st;
1695 #ifdef DEBUG_RTP_TCP
1696 dprintf(s, "tcp_read_packet:\n");
1700 RTSPMessageHeader reply;
1702 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1705 if (ret == 1) /* received '$' */
1707 /* XXX: parse message */
1708 if (rt->state != RTSP_STATE_STREAMING)
1711 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1715 len = AV_RB16(buf + 1);
1716 #ifdef DEBUG_RTP_TCP
1717 dprintf(s, "id=%d len=%d\n", id, len);
1719 if (len > buf_size || len < 12)
1722 ret = url_read_complete(rt->rtsp_hd, buf, len);
1725 if (rt->transport == RTSP_TRANSPORT_RDT &&
1726 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1729 /* find the matching stream */
1730 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1731 rtsp_st = rt->rtsp_streams[i];
1732 if (id >= rtsp_st->interleaved_min &&
1733 id <= rtsp_st->interleaved_max)
1738 *prtsp_st = rtsp_st;
1742 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1744 RTSPState *rt = s->priv_data;
1746 uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
1747 RTSPStream *rtsp_st;
1749 /* get next frames from the same RTP packet */
1750 if (rt->cur_transport_priv) {
1751 if (rt->transport == RTSP_TRANSPORT_RDT) {
1752 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1754 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1756 rt->cur_transport_priv = NULL;
1758 } else if (ret == 1) {
1761 rt->cur_transport_priv = NULL;
1764 /* read next RTP packet */
1766 switch(rt->lower_transport) {
1768 #if CONFIG_RTSP_DEMUXER
1769 case RTSP_LOWER_TRANSPORT_TCP:
1770 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1773 case RTSP_LOWER_TRANSPORT_UDP:
1774 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1775 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1776 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1777 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1784 if (rt->transport == RTSP_TRANSPORT_RDT) {
1785 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1787 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1791 /* more packets may follow, so we save the RTP context */
1792 rt->cur_transport_priv = rtsp_st->transport_priv;
1797 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1799 RTSPState *rt = s->priv_data;
1801 RTSPMessageHeader reply1, *reply = &reply1;
1804 if (rt->server_type == RTSP_SERVER_REAL) {
1806 enum AVDiscard cache[MAX_STREAMS];
1808 for (i = 0; i < s->nb_streams; i++)
1809 cache[i] = s->streams[i]->discard;
1811 if (!rt->need_subscription) {
1812 if (memcmp (cache, rt->real_setup_cache,
1813 sizeof(enum AVDiscard) * s->nb_streams)) {
1814 snprintf(cmd, sizeof(cmd),
1815 "Unsubscribe: %s\r\n",
1816 rt->last_subscription);
1817 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1819 if (reply->status_code != RTSP_STATUS_OK)
1820 return AVERROR_INVALIDDATA;
1821 rt->need_subscription = 1;
1825 if (rt->need_subscription) {
1826 int r, rule_nr, first = 1;
1828 memcpy(rt->real_setup_cache, cache,
1829 sizeof(enum AVDiscard) * s->nb_streams);
1830 rt->last_subscription[0] = 0;
1832 snprintf(cmd, sizeof(cmd),
1834 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1836 for (r = 0; r < s->nb_streams; r++) {
1837 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1838 if (s->streams[r]->discard != AVDISCARD_ALL) {
1840 av_strlcat(rt->last_subscription, ",",
1841 sizeof(rt->last_subscription));
1842 ff_rdt_subscribe_rule(
1843 rt->last_subscription,
1844 sizeof(rt->last_subscription), i, rule_nr);
1851 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1852 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1854 if (reply->status_code != RTSP_STATUS_OK)
1855 return AVERROR_INVALIDDATA;
1856 rt->need_subscription = 0;
1858 if (rt->state == RTSP_STATE_STREAMING)
1863 ret = rtsp_fetch_packet(s, pkt);
1867 /* send dummy request to keep TCP connection alive */
1868 if ((rt->server_type == RTSP_SERVER_WMS ||
1869 rt->server_type == RTSP_SERVER_REAL) &&
1870 (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1871 if (rt->server_type == RTSP_SERVER_WMS) {
1872 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1874 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1881 /* pause the stream */
1882 static int rtsp_read_pause(AVFormatContext *s)
1884 RTSPState *rt = s->priv_data;
1885 RTSPMessageHeader reply1, *reply = &reply1;
1889 if (rt->state != RTSP_STATE_STREAMING)
1891 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1892 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1893 if (reply->status_code != RTSP_STATUS_OK) {
1897 rt->state = RTSP_STATE_PAUSED;
1901 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1902 int64_t timestamp, int flags)
1904 RTSPState *rt = s->priv_data;
1906 rt->seek_timestamp = av_rescale_q(timestamp,
1907 s->streams[stream_index]->time_base,
1911 case RTSP_STATE_IDLE:
1913 case RTSP_STATE_STREAMING:
1914 if (rtsp_read_pause(s) != 0)
1916 rt->state = RTSP_STATE_SEEKING;
1917 if (rtsp_read_play(s) != 0)
1920 case RTSP_STATE_PAUSED:
1921 rt->state = RTSP_STATE_IDLE;
1927 static int rtsp_read_close(AVFormatContext *s)
1929 RTSPState *rt = s->priv_data;
1932 /* NOTE: it is valid to flush the buffer here */
1933 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1934 url_fclose(&rt->rtsp_gb);
1937 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1939 ff_rtsp_close_streams(s);
1940 url_close(rt->rtsp_hd);
1945 AVInputFormat rtsp_demuxer = {
1947 NULL_IF_CONFIG_SMALL("RTSP input format"),
1954 .flags = AVFMT_NOFILE,
1955 .read_play = rtsp_read_play,
1956 .read_pause = rtsp_read_pause,
1960 static int sdp_probe(AVProbeData *p1)
1962 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1964 /* we look for a line beginning "c=IN IP4" */
1965 while (p < p_end && *p != '\0') {
1966 if (p + sizeof("c=IN IP4") - 1 < p_end &&
1967 av_strstart(p, "c=IN IP4", NULL))
1968 return AVPROBE_SCORE_MAX / 2;
1970 while (p < p_end - 1 && *p != '\n') p++;
1979 #define SDP_MAX_SIZE 8192
1981 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1983 RTSPState *rt = s->priv_data;
1984 RTSPStream *rtsp_st;
1989 if (!ff_network_init())
1990 return AVERROR(EIO);
1992 /* read the whole sdp file */
1993 /* XXX: better loading */
1994 content = av_malloc(SDP_MAX_SIZE);
1995 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1998 return AVERROR_INVALIDDATA;
2000 content[size] ='\0';
2002 sdp_parse(s, content);
2005 /* open each RTP stream */
2006 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2007 rtsp_st = rt->rtsp_streams[i];
2009 ff_url_join(url, sizeof(url), "rtp", NULL,
2010 inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
2011 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2013 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2014 err = AVERROR_INVALIDDATA;
2017 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2022 ff_rtsp_close_streams(s);
2027 static int sdp_read_close(AVFormatContext *s)
2029 ff_rtsp_close_streams(s);
2034 AVInputFormat sdp_demuxer = {
2036 NULL_IF_CONFIG_SMALL("SDP"),