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"
42 //#define DEBUG_RTP_TCP
44 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
45 int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
48 /* Timeout values for socket select, in ms,
49 * and read_packet(), in seconds */
50 #define SELECT_TIMEOUT_MS 100
51 #define READ_PACKET_TIMEOUT_S 10
52 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
54 #define SPACE_CHARS " \t\r\n"
55 /* we use memchr() instead of strchr() here because strchr() will return
56 * the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
57 #define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
58 static void skip_spaces(const char **pp)
62 while (redir_isspace(*p))
67 static void get_word_until_chars(char *buf, int buf_size,
68 const char *sep, const char **pp)
76 while (!strchr(sep, *p) && *p != '\0') {
77 if ((q - buf) < buf_size - 1)
86 static void get_word_sep(char *buf, int buf_size, const char *sep,
89 if (**pp == '/') (*pp)++;
90 get_word_until_chars(buf, buf_size, sep, pp);
93 static void get_word(char *buf, int buf_size, const char **pp)
95 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
98 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
99 static int sdp_parse_rtpmap(AVFormatContext *s,
100 AVCodecContext *codec, RTSPStream *rtsp_st,
101 int payload_type, const char *p)
108 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
109 * see if we can handle this kind of payload.
110 * The space should normally not be there but some Real streams or
111 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
112 * have a trailing space. */
113 get_word_sep(buf, sizeof(buf), "/ ", &p);
114 if (payload_type >= RTP_PT_PRIVATE) {
115 RTPDynamicProtocolHandler *handler;
116 for (handler = RTPFirstDynamicPayloadHandler;
117 handler; handler = handler->next) {
118 if (!strcasecmp(buf, handler->enc_name) &&
119 codec->codec_type == handler->codec_type) {
120 codec->codec_id = handler->codec_id;
121 rtsp_st->dynamic_handler = handler;
123 rtsp_st->dynamic_protocol_context = handler->open();
128 /* We are in a standard case
129 * (from http://www.iana.org/assignments/rtp-parameters). */
130 /* search into AVRtpPayloadTypes[] */
131 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
134 c = avcodec_find_decoder(codec->codec_id);
140 get_word_sep(buf, sizeof(buf), "/", &p);
142 switch (codec->codec_type) {
143 case AVMEDIA_TYPE_AUDIO:
144 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
145 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
146 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
148 codec->sample_rate = i;
149 get_word_sep(buf, sizeof(buf), "/", &p);
153 // TODO: there is a bug here; if it is a mono stream, and
154 // less than 22000Hz, faad upconverts to stereo and twice
155 // the frequency. No problem, but the sample rate is being
156 // set here by the sdp line. Patch on its way. (rdm)
158 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
160 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
163 case AVMEDIA_TYPE_VIDEO:
164 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
172 /* return the length and optionally the data */
173 static int hex_to_data(uint8_t *data, const char *p)
183 c = toupper((unsigned char) *p++);
184 if (c >= '0' && c <= '9')
186 else if (c >= 'A' && c <= 'F')
201 static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
202 char *attr, char *value)
204 switch (codec->codec_id) {
207 if (!strcmp(attr, "config")) {
208 /* decode the hexa encoded parameter */
209 int len = hex_to_data(NULL, value);
210 if (codec->extradata)
211 av_free(codec->extradata);
212 codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
213 if (!codec->extradata)
215 codec->extradata_size = len;
216 hex_to_data(codec->extradata, value);
231 /* All known fmtp parameters and the corresponding RTPAttrTypeEnum */
232 #define ATTR_NAME_TYPE_INT 0
233 #define ATTR_NAME_TYPE_STR 1
234 static const AttrNameMap attr_names[]=
236 { "SizeLength", ATTR_NAME_TYPE_INT,
237 offsetof(RTPPayloadData, sizelength) },
238 { "IndexLength", ATTR_NAME_TYPE_INT,
239 offsetof(RTPPayloadData, indexlength) },
240 { "IndexDeltaLength", ATTR_NAME_TYPE_INT,
241 offsetof(RTPPayloadData, indexdeltalength) },
242 { "profile-level-id", ATTR_NAME_TYPE_INT,
243 offsetof(RTPPayloadData, profile_level_id) },
244 { "StreamType", ATTR_NAME_TYPE_INT,
245 offsetof(RTPPayloadData, streamtype) },
246 { "mode", ATTR_NAME_TYPE_STR,
247 offsetof(RTPPayloadData, mode) },
251 /* parse the attribute line from the fmtp a line of an sdp response. This
252 * is broken out as a function because it is used in rtp_h264.c, which is
254 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
255 char *value, int value_size)
259 get_word_sep(attr, attr_size, "=", p);
262 get_word_sep(value, value_size, ";", p);
270 /* parse a SDP line and save stream attributes */
271 static void sdp_parse_fmtp(AVStream *st, const char *p)
274 /* Vorbis setup headers can be up to 12KB and are sent base64
275 * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
278 RTSPStream *rtsp_st = st->priv_data;
279 AVCodecContext *codec = st->codec;
280 RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
282 /* loop on each attribute */
283 while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
284 value, sizeof(value))) {
285 /* grab the codec extra_data from the config parameter of the fmtp
287 sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
289 /* Looking for a known attribute */
290 for (i = 0; attr_names[i].str; ++i) {
291 if (!strcasecmp(attr, attr_names[i].str)) {
292 if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
293 *(int *)((char *)rtp_payload_data +
294 attr_names[i].offset) = atoi(value);
295 } else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
296 *(char **)((char *)rtp_payload_data +
297 attr_names[i].offset) = av_strdup(value);
303 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
305 * Used for seeking in the rtp stream.
307 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
312 if (!av_stristart(p, "npt=", &p))
315 *start = AV_NOPTS_VALUE;
316 *end = AV_NOPTS_VALUE;
318 get_word_sep(buf, sizeof(buf), "-", &p);
319 *start = parse_date(buf, 1);
322 get_word_sep(buf, sizeof(buf), "-", &p);
323 *end = parse_date(buf, 1);
325 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
326 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
329 typedef struct SDPParseState {
331 struct in_addr default_ip;
333 int skip_media; ///< set if an unknown m= line occurs
336 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
337 int letter, const char *buf)
339 RTSPState *rt = s->priv_data;
340 char buf1[64], st_type[64];
342 enum AVMediaType codec_type;
346 struct in_addr sdp_ip;
349 dprintf(s, "sdp: %c='%s'\n", letter, buf);
352 if (s1->skip_media && letter != 'm')
356 get_word(buf1, sizeof(buf1), &p);
357 if (strcmp(buf1, "IN") != 0)
359 get_word(buf1, sizeof(buf1), &p);
360 if (strcmp(buf1, "IP4") != 0)
362 get_word_sep(buf1, sizeof(buf1), "/", &p);
363 if (ff_inet_aton(buf1, &sdp_ip) == 0)
368 get_word_sep(buf1, sizeof(buf1), "/", &p);
371 if (s->nb_streams == 0) {
372 s1->default_ip = sdp_ip;
373 s1->default_ttl = ttl;
375 st = s->streams[s->nb_streams - 1];
376 rtsp_st = st->priv_data;
377 rtsp_st->sdp_ip = sdp_ip;
378 rtsp_st->sdp_ttl = ttl;
382 av_metadata_set2(&s->metadata, "title", p, 0);
385 if (s->nb_streams == 0) {
386 av_metadata_set2(&s->metadata, "comment", p, 0);
393 get_word(st_type, sizeof(st_type), &p);
394 if (!strcmp(st_type, "audio")) {
395 codec_type = AVMEDIA_TYPE_AUDIO;
396 } else if (!strcmp(st_type, "video")) {
397 codec_type = AVMEDIA_TYPE_VIDEO;
398 } else if (!strcmp(st_type, "application")) {
399 codec_type = AVMEDIA_TYPE_DATA;
404 rtsp_st = av_mallocz(sizeof(RTSPStream));
407 rtsp_st->stream_index = -1;
408 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
410 rtsp_st->sdp_ip = s1->default_ip;
411 rtsp_st->sdp_ttl = s1->default_ttl;
413 get_word(buf1, sizeof(buf1), &p); /* port */
414 rtsp_st->sdp_port = atoi(buf1);
416 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
418 /* XXX: handle list of formats */
419 get_word(buf1, sizeof(buf1), &p); /* format list */
420 rtsp_st->sdp_payload_type = atoi(buf1);
422 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
423 /* no corresponding stream */
425 st = av_new_stream(s, 0);
428 st->priv_data = rtsp_st;
429 rtsp_st->stream_index = st->index;
430 st->codec->codec_type = codec_type;
431 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
432 /* if standard payload type, we can find the codec right now */
433 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
436 /* put a default control url */
437 av_strlcpy(rtsp_st->control_url, rt->control_uri,
438 sizeof(rtsp_st->control_url));
441 if (av_strstart(p, "control:", &p)) {
442 if (s->nb_streams == 0) {
443 if (!strncmp(p, "rtsp://", 7))
444 av_strlcpy(rt->control_uri, p,
445 sizeof(rt->control_uri));
448 /* get the control url */
449 st = s->streams[s->nb_streams - 1];
450 rtsp_st = st->priv_data;
452 /* XXX: may need to add full url resolution */
453 ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
455 if (proto[0] == '\0') {
456 /* relative control URL */
457 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
458 av_strlcat(rtsp_st->control_url, "/",
459 sizeof(rtsp_st->control_url));
460 av_strlcat(rtsp_st->control_url, p,
461 sizeof(rtsp_st->control_url));
463 av_strlcpy(rtsp_st->control_url, p,
464 sizeof(rtsp_st->control_url));
466 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
467 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
468 get_word(buf1, sizeof(buf1), &p);
469 payload_type = atoi(buf1);
470 st = s->streams[s->nb_streams - 1];
471 rtsp_st = st->priv_data;
472 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
473 } else if (av_strstart(p, "fmtp:", &p)) {
474 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
475 get_word(buf1, sizeof(buf1), &p);
476 payload_type = atoi(buf1);
477 for (i = 0; i < s->nb_streams; i++) {
479 rtsp_st = st->priv_data;
480 if (rtsp_st->sdp_payload_type == payload_type) {
481 if (!(rtsp_st->dynamic_handler &&
482 rtsp_st->dynamic_handler->parse_sdp_a_line &&
483 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
484 i, rtsp_st->dynamic_protocol_context, buf)))
485 sdp_parse_fmtp(st, p);
488 } else if (av_strstart(p, "framesize:", &p)) {
489 // let dynamic protocol handlers have a stab at the line.
490 get_word(buf1, sizeof(buf1), &p);
491 payload_type = atoi(buf1);
492 for (i = 0; i < s->nb_streams; i++) {
494 rtsp_st = st->priv_data;
495 if (rtsp_st->sdp_payload_type == payload_type &&
496 rtsp_st->dynamic_handler &&
497 rtsp_st->dynamic_handler->parse_sdp_a_line)
498 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
499 rtsp_st->dynamic_protocol_context, buf);
501 } else if (av_strstart(p, "range:", &p)) {
504 // this is so that seeking on a streamed file can work.
505 rtsp_parse_range_npt(p, &start, &end);
506 s->start_time = start;
507 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
508 s->duration = (end == AV_NOPTS_VALUE) ?
509 AV_NOPTS_VALUE : end - start;
510 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
512 rt->transport = RTSP_TRANSPORT_RDT;
514 if (rt->server_type == RTSP_SERVER_WMS)
515 ff_wms_parse_sdp_a_line(s, p);
516 if (s->nb_streams > 0) {
517 if (rt->server_type == RTSP_SERVER_REAL)
518 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
520 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
521 if (rtsp_st->dynamic_handler &&
522 rtsp_st->dynamic_handler->parse_sdp_a_line)
523 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
525 rtsp_st->dynamic_protocol_context, buf);
532 static int sdp_parse(AVFormatContext *s, const char *content)
536 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
537 * contain long SDP lines containing complete ASF Headers (several
538 * kB) or arrays of MDPR (RM stream descriptor) headers plus
539 * "rulebooks" describing their properties. Therefore, the SDP line
542 * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
544 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
546 memset(s1, 0, sizeof(SDPParseState));
557 /* get the content */
559 while (*p != '\n' && *p != '\r' && *p != '\0') {
560 if ((q - buf) < sizeof(buf) - 1)
565 sdp_parse_line(s, s1, letter, buf);
567 while (*p != '\n' && *p != '\0')
575 /* close and free RTSP streams */
576 void ff_rtsp_close_streams(AVFormatContext *s)
578 RTSPState *rt = s->priv_data;
582 for (i = 0; i < rt->nb_rtsp_streams; i++) {
583 rtsp_st = rt->rtsp_streams[i];
585 if (rtsp_st->transport_priv) {
587 AVFormatContext *rtpctx = rtsp_st->transport_priv;
588 av_write_trailer(rtpctx);
589 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
591 url_close_dyn_buf(rtpctx->pb, &ptr);
594 url_fclose(rtpctx->pb);
596 av_metadata_free(&rtpctx->streams[0]->metadata);
597 av_metadata_free(&rtpctx->metadata);
598 av_free(rtpctx->streams[0]);
600 } else if (rt->transport == RTSP_TRANSPORT_RDT)
601 ff_rdt_parse_close(rtsp_st->transport_priv);
603 rtp_parse_close(rtsp_st->transport_priv);
605 if (rtsp_st->rtp_handle)
606 url_close(rtsp_st->rtp_handle);
607 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
608 rtsp_st->dynamic_handler->close(
609 rtsp_st->dynamic_protocol_context);
612 av_free(rt->rtsp_streams);
614 av_close_input_stream (rt->asf_ctx);
619 static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
622 RTSPState *rt = s->priv_data;
623 AVFormatContext *rtpctx;
625 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
630 /* Allocate an AVFormatContext for each output stream */
631 rtpctx = avformat_alloc_context();
635 rtpctx->oformat = rtp_format;
636 if (!av_new_stream(rtpctx, 0)) {
640 /* Copy the max delay setting; the rtp muxer reads this. */
641 rtpctx->max_delay = s->max_delay;
642 /* Copy other stream parameters. */
643 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
645 /* Set the synchronized start time. */
646 rtpctx->start_time_realtime = rt->start_time;
648 /* Remove the local codec, link to the original codec
649 * context instead, to give the rtp muxer access to
650 * codec parameters. */
651 av_free(rtpctx->streams[0]->codec);
652 rtpctx->streams[0]->codec = st->codec;
655 url_fdopen(&rtpctx->pb, handle);
657 url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
658 ret = av_write_header(rtpctx);
662 url_fclose(rtpctx->pb);
665 url_close_dyn_buf(rtpctx->pb, &ptr);
668 av_free(rtpctx->streams[0]);
673 /* Copy the RTP AVStream timebase back to the original AVStream */
674 st->time_base = rtpctx->streams[0]->time_base;
678 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
680 RTSPState *rt = s->priv_data;
683 /* open the RTP context */
684 if (rtsp_st->stream_index >= 0)
685 st = s->streams[rtsp_st->stream_index];
687 s->ctx_flags |= AVFMTCTX_NOHEADER;
690 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
691 /* Ownership of rtp_handle is passed to the rtp mux context */
692 rtsp_st->rtp_handle = NULL;
693 } else if (rt->transport == RTSP_TRANSPORT_RDT)
694 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
695 rtsp_st->dynamic_protocol_context,
696 rtsp_st->dynamic_handler);
698 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
699 rtsp_st->sdp_payload_type,
700 &rtsp_st->rtp_payload_data);
702 if (!rtsp_st->transport_priv) {
703 return AVERROR(ENOMEM);
704 } else if (rt->transport != RTSP_TRANSPORT_RDT) {
705 if (rtsp_st->dynamic_handler) {
706 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
707 rtsp_st->dynamic_protocol_context,
708 rtsp_st->dynamic_handler);
715 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
716 static int rtsp_probe(AVProbeData *p)
718 if (av_strstart(p->filename, "rtsp:", NULL))
719 return AVPROBE_SCORE_MAX;
723 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
730 v = strtol(p, (char **)&p, 10);
734 v = strtol(p, (char **)&p, 10);
743 /* XXX: only one transport specification is parsed */
744 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
746 char transport_protocol[16];
748 char lower_transport[16];
750 RTSPTransportField *th;
753 reply->nb_transports = 0;
760 th = &reply->transports[reply->nb_transports];
762 get_word_sep(transport_protocol, sizeof(transport_protocol),
764 if (!strcasecmp (transport_protocol, "rtp")) {
765 get_word_sep(profile, sizeof(profile), "/;,", &p);
766 lower_transport[0] = '\0';
767 /* rtp/avp/<protocol> */
769 get_word_sep(lower_transport, sizeof(lower_transport),
772 th->transport = RTSP_TRANSPORT_RTP;
773 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
774 !strcasecmp (transport_protocol, "x-real-rdt")) {
775 /* x-pn-tng/<protocol> */
776 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
778 th->transport = RTSP_TRANSPORT_RDT;
780 if (!strcasecmp(lower_transport, "TCP"))
781 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
783 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
787 /* get each parameter */
788 while (*p != '\0' && *p != ',') {
789 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
790 if (!strcmp(parameter, "port")) {
793 rtsp_parse_range(&th->port_min, &th->port_max, &p);
795 } else if (!strcmp(parameter, "client_port")) {
798 rtsp_parse_range(&th->client_port_min,
799 &th->client_port_max, &p);
801 } else if (!strcmp(parameter, "server_port")) {
804 rtsp_parse_range(&th->server_port_min,
805 &th->server_port_max, &p);
807 } else if (!strcmp(parameter, "interleaved")) {
810 rtsp_parse_range(&th->interleaved_min,
811 &th->interleaved_max, &p);
813 } else if (!strcmp(parameter, "multicast")) {
814 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
815 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
816 } else if (!strcmp(parameter, "ttl")) {
819 th->ttl = strtol(p, (char **)&p, 10);
821 } else if (!strcmp(parameter, "destination")) {
822 struct in_addr ipaddr;
826 get_word_sep(buf, sizeof(buf), ";,", &p);
827 if (ff_inet_aton(buf, &ipaddr))
828 th->destination = ntohl(ipaddr.s_addr);
831 while (*p != ';' && *p != '\0' && *p != ',')
839 reply->nb_transports++;
843 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
844 HTTPAuthState *auth_state)
848 /* NOTE: we do case independent match for broken servers */
850 if (av_stristart(p, "Session:", &p)) {
852 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
853 if (av_stristart(p, ";timeout=", &p) &&
854 (t = strtol(p, NULL, 10)) > 0) {
857 } else if (av_stristart(p, "Content-Length:", &p)) {
858 reply->content_length = strtol(p, NULL, 10);
859 } else if (av_stristart(p, "Transport:", &p)) {
860 rtsp_parse_transport(reply, p);
861 } else if (av_stristart(p, "CSeq:", &p)) {
862 reply->seq = strtol(p, NULL, 10);
863 } else if (av_stristart(p, "Range:", &p)) {
864 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
865 } else if (av_stristart(p, "RealChallenge1:", &p)) {
867 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
868 } else if (av_stristart(p, "Server:", &p)) {
870 av_strlcpy(reply->server, p, sizeof(reply->server));
871 } else if (av_stristart(p, "Notice:", &p) ||
872 av_stristart(p, "X-Notice:", &p)) {
873 reply->notice = strtol(p, NULL, 10);
874 } else if (av_stristart(p, "Location:", &p)) {
876 av_strlcpy(reply->location, p , sizeof(reply->location));
877 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
879 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
880 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
882 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
886 /* skip a RTP/TCP interleaved packet */
887 void ff_rtsp_skip_packet(AVFormatContext *s)
889 RTSPState *rt = s->priv_data;
893 ret = url_read_complete(rt->rtsp_hd, buf, 3);
896 len = AV_RB16(buf + 1);
898 dprintf(s, "skipping RTP packet len=%d\n", len);
903 if (len1 > sizeof(buf))
905 ret = url_read_complete(rt->rtsp_hd, buf, len1);
912 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
913 unsigned char **content_ptr,
914 int return_on_interleaved_data)
916 RTSPState *rt = s->priv_data;
917 char buf[4096], buf1[1024], *q;
920 int ret, content_length, line_count = 0;
921 unsigned char *content = NULL;
923 memset(reply, 0, sizeof(*reply));
925 /* parse reply (XXX: use buffers) */
926 rt->last_reply[0] = '\0';
930 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
932 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
939 /* XXX: only parse it if first char on line ? */
940 if (return_on_interleaved_data) {
943 ff_rtsp_skip_packet(s);
944 } else if (ch != '\r') {
945 if ((q - buf) < sizeof(buf) - 1)
951 dprintf(s, "line='%s'\n", buf);
953 /* test if last line */
957 if (line_count == 0) {
959 get_word(buf1, sizeof(buf1), &p);
960 get_word(buf1, sizeof(buf1), &p);
961 reply->status_code = atoi(buf1);
963 ff_rtsp_parse_line(reply, p, &rt->auth_state);
964 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
965 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
970 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
971 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
973 content_length = reply->content_length;
974 if (content_length > 0) {
975 /* leave some room for a trailing '\0' (useful for simple parsing) */
976 content = av_malloc(content_length + 1);
977 (void)url_read_complete(rt->rtsp_hd, content, content_length);
978 content[content_length] = '\0';
981 *content_ptr = content;
985 if (rt->seq != reply->seq) {
986 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
987 rt->seq, reply->seq);
991 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
992 reply->notice == 2104 /* Start-of-Stream Reached */ ||
993 reply->notice == 2306 /* Continuous Feed Terminated */) {
994 rt->state = RTSP_STATE_IDLE;
995 } else if (reply->notice >= 4400 && reply->notice < 5500) {
996 return AVERROR(EIO); /* data or server error */
997 } else if (reply->notice == 2401 /* Ticket Expired */ ||
998 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
999 return AVERROR(EPERM);
1004 void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1005 const char *method, const char *url,
1006 const char *headers,
1007 const unsigned char *send_content,
1008 int send_content_length)
1010 RTSPState *rt = s->priv_data;
1014 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1016 av_strlcat(buf, headers, sizeof(buf));
1017 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1018 if (rt->session_id[0] != '\0' && (!headers ||
1019 !strstr(headers, "\nIf-Match:"))) {
1020 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
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 RTSPState *rt = s->priv_data;
1064 HTTPAuthType cur_auth_type;
1067 cur_auth_type = rt->auth_state.auth_type;
1068 ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1069 send_content, send_content_length);
1071 ff_rtsp_read_reply(s, reply, content_ptr, 0);
1073 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1074 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1079 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1081 static int make_setup_request(AVFormatContext *s, const char *host, int port,
1082 int lower_transport, const char *real_challenge)
1084 RTSPState *rt = s->priv_data;
1085 int rtx, j, i, err, interleave = 0;
1086 RTSPStream *rtsp_st;
1087 RTSPMessageHeader reply1, *reply = &reply1;
1089 const char *trans_pref;
1091 if (rt->transport == RTSP_TRANSPORT_RDT)
1092 trans_pref = "x-pn-tng";
1094 trans_pref = "RTP/AVP";
1096 /* default timeout: 1 minute */
1099 /* for each stream, make the setup request */
1100 /* XXX: we assume the same server is used for the control of each
1103 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1104 char transport[2048];
1107 * WMS serves all UDP data over a single connection, the RTX, which
1108 * isn't necessarily the first in the SDP but has to be the first
1109 * to be set up, else the second/third SETUP will fail with a 461.
1111 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1112 rt->server_type == RTSP_SERVER_WMS) {
1115 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1116 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1118 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1122 if (rtx == rt->nb_rtsp_streams)
1123 return -1; /* no RTX found */
1124 rtsp_st = rt->rtsp_streams[rtx];
1126 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1128 rtsp_st = rt->rtsp_streams[i];
1131 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1134 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1135 port = reply->transports[0].client_port_min;
1139 /* first try in specified port range */
1140 if (RTSP_RTP_PORT_MIN != 0) {
1141 while (j <= RTSP_RTP_PORT_MAX) {
1142 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1143 "?localport=%d", j);
1144 /* we will use two ports per rtp stream (rtp and rtcp) */
1146 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1152 /* then try on any port */
1153 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1154 err = AVERROR_INVALIDDATA;
1160 port = rtp_get_local_port(rtsp_st->rtp_handle);
1162 snprintf(transport, sizeof(transport) - 1,
1163 "%s/UDP;", trans_pref);
1164 if (rt->server_type != RTSP_SERVER_REAL)
1165 av_strlcat(transport, "unicast;", sizeof(transport));
1166 av_strlcatf(transport, sizeof(transport),
1167 "client_port=%d", port);
1168 if (rt->transport == RTSP_TRANSPORT_RTP &&
1169 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1170 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1174 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1175 /** For WMS streams, the application streams are only used for
1176 * UDP. When trying to set it up for TCP streams, the server
1177 * will return an error. Therefore, we skip those streams. */
1178 if (rt->server_type == RTSP_SERVER_WMS &&
1179 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1182 snprintf(transport, sizeof(transport) - 1,
1183 "%s/TCP;", trans_pref);
1184 if (rt->server_type == RTSP_SERVER_WMS)
1185 av_strlcat(transport, "unicast;", sizeof(transport));
1186 av_strlcatf(transport, sizeof(transport),
1187 "interleaved=%d-%d",
1188 interleave, interleave + 1);
1192 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1193 snprintf(transport, sizeof(transport) - 1,
1194 "%s/UDP;multicast", trans_pref);
1197 av_strlcat(transport, ";mode=receive", sizeof(transport));
1198 } else if (rt->server_type == RTSP_SERVER_REAL ||
1199 rt->server_type == RTSP_SERVER_WMS)
1200 av_strlcat(transport, ";mode=play", sizeof(transport));
1201 snprintf(cmd, sizeof(cmd),
1202 "Transport: %s\r\n",
1204 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1205 char real_res[41], real_csum[9];
1206 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1208 av_strlcatf(cmd, sizeof(cmd),
1210 "RealChallenge2: %s, sd=%s\r\n",
1211 rt->session_id, real_res, real_csum);
1213 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1214 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1217 } else if (reply->status_code != RTSP_STATUS_OK ||
1218 reply->nb_transports != 1) {
1219 err = AVERROR_INVALIDDATA;
1223 /* XXX: same protocol for all streams is required */
1225 if (reply->transports[0].lower_transport != rt->lower_transport ||
1226 reply->transports[0].transport != rt->transport) {
1227 err = AVERROR_INVALIDDATA;
1231 rt->lower_transport = reply->transports[0].lower_transport;
1232 rt->transport = reply->transports[0].transport;
1235 /* close RTP connection if not choosen */
1236 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1237 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1238 url_close(rtsp_st->rtp_handle);
1239 rtsp_st->rtp_handle = NULL;
1242 switch(reply->transports[0].lower_transport) {
1243 case RTSP_LOWER_TRANSPORT_TCP:
1244 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1245 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1248 case RTSP_LOWER_TRANSPORT_UDP: {
1251 /* XXX: also use address if specified */
1252 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1253 reply->transports[0].server_port_min, NULL);
1254 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1255 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1256 err = AVERROR_INVALIDDATA;
1259 /* Try to initialize the connection state in a
1260 * potential NAT router by sending dummy packets.
1261 * RTP/RTCP dummy packets are used for RDT, too.
1263 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1264 rtp_send_punch_packets(rtsp_st->rtp_handle);
1267 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1272 if (reply->transports[0].destination) {
1273 in.s_addr = htonl(reply->transports[0].destination);
1274 port = reply->transports[0].port_min;
1275 ttl = reply->transports[0].ttl;
1277 in = rtsp_st->sdp_ip;
1278 port = rtsp_st->sdp_port;
1279 ttl = rtsp_st->sdp_ttl;
1281 ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
1282 port, "?ttl=%d", ttl);
1283 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1284 err = AVERROR_INVALIDDATA;
1291 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1295 if (reply->timeout > 0)
1296 rt->timeout = reply->timeout;
1298 if (rt->server_type == RTSP_SERVER_REAL)
1299 rt->need_subscription = 1;
1304 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1305 if (rt->rtsp_streams[i]->rtp_handle) {
1306 url_close(rt->rtsp_streams[i]->rtp_handle);
1307 rt->rtsp_streams[i]->rtp_handle = NULL;
1313 static int rtsp_read_play(AVFormatContext *s)
1315 RTSPState *rt = s->priv_data;
1316 RTSPMessageHeader reply1, *reply = &reply1;
1320 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1322 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1323 if (rt->state == RTSP_STATE_PAUSED) {
1326 snprintf(cmd, sizeof(cmd),
1327 "Range: npt=%0.3f-\r\n",
1328 (double)rt->seek_timestamp / AV_TIME_BASE);
1330 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1331 if (reply->status_code != RTSP_STATUS_OK) {
1334 if (reply->range_start != AV_NOPTS_VALUE &&
1335 rt->transport == RTSP_TRANSPORT_RTP) {
1336 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1337 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1338 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1339 AVStream *st = NULL;
1340 if (rtsp_st->stream_index >= 0)
1341 st = s->streams[rtsp_st->stream_index];
1342 rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
1343 rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
1345 rtpctx->range_start_offset = av_rescale_q(reply->range_start,
1351 rt->state = RTSP_STATE_STREAMING;
1355 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1357 RTSPState *rt = s->priv_data;
1359 unsigned char *content = NULL;
1362 /* describe the stream */
1363 snprintf(cmd, sizeof(cmd),
1364 "Accept: application/sdp\r\n");
1365 if (rt->server_type == RTSP_SERVER_REAL) {
1367 * The Require: attribute is needed for proper streaming from
1368 * Realmedia servers.
1371 "Require: com.real.retain-entity-for-setup\r\n",
1374 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1376 return AVERROR_INVALIDDATA;
1377 if (reply->status_code != RTSP_STATUS_OK) {
1379 return AVERROR_INVALIDDATA;
1382 /* now we got the SDP description, we parse it */
1383 ret = sdp_parse(s, (const char *)content);
1386 return AVERROR_INVALIDDATA;
1391 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1393 RTSPState *rt = s->priv_data;
1394 RTSPMessageHeader reply1, *reply = &reply1;
1397 AVFormatContext sdp_ctx, *ctx_array[1];
1399 rt->start_time = av_gettime();
1401 /* Announce the stream */
1402 sdp = av_mallocz(8192);
1404 return AVERROR(ENOMEM);
1405 /* We create the SDP based on the RTSP AVFormatContext where we
1406 * aren't allowed to change the filename field. (We create the SDP
1407 * based on the RTSP context since the contexts for the RTP streams
1408 * don't exist yet.) In order to specify a custom URL with the actual
1409 * peer IP instead of the originally specified hostname, we create
1410 * a temporary copy of the AVFormatContext, where the custom URL is set.
1412 * FIXME: Create the SDP without copying the AVFormatContext.
1413 * This either requires setting up the RTP stream AVFormatContexts
1414 * already here (complicating things immensely) or getting a more
1415 * flexible SDP creation interface.
1418 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1419 "rtsp", NULL, addr, -1, NULL);
1420 ctx_array[0] = &sdp_ctx;
1421 if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
1423 return AVERROR_INVALIDDATA;
1425 av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1426 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1427 "Content-Type: application/sdp\r\n",
1428 reply, NULL, sdp, strlen(sdp));
1430 if (reply->status_code != RTSP_STATUS_OK)
1431 return AVERROR_INVALIDDATA;
1433 /* Set up the RTSPStreams for each AVStream */
1434 for (i = 0; i < s->nb_streams; i++) {
1435 RTSPStream *rtsp_st;
1436 AVStream *st = s->streams[i];
1438 rtsp_st = av_mallocz(sizeof(RTSPStream));
1440 return AVERROR(ENOMEM);
1441 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1443 st->priv_data = rtsp_st;
1444 rtsp_st->stream_index = i;
1446 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1447 /* Note, this must match the relative uri set in the sdp content */
1448 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1455 int ff_rtsp_connect(AVFormatContext *s)
1457 RTSPState *rt = s->priv_data;
1458 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1459 char *option_list, *option, *filename;
1460 URLContext *rtsp_hd;
1461 int port, err, tcp_fd;
1462 RTSPMessageHeader reply1 = {}, *reply = &reply1;
1463 int lower_transport_mask = 0;
1464 char real_challenge[64];
1465 struct sockaddr_storage peer;
1466 socklen_t peer_len = sizeof(peer);
1468 if (!ff_network_init())
1469 return AVERROR(EIO);
1471 /* extract hostname and port */
1472 ff_url_split(NULL, 0, auth, sizeof(auth),
1473 host, sizeof(host), &port, path, sizeof(path), s->filename);
1475 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1478 port = RTSP_DEFAULT_PORT;
1480 /* search for options */
1481 option_list = strrchr(path, '?');
1483 /* Strip out the RTSP specific options, write out the rest of
1484 * the options back into the same string. */
1485 filename = option_list;
1486 while (option_list) {
1487 /* move the option pointer */
1488 option = ++option_list;
1489 option_list = strchr(option_list, '&');
1493 /* handle the options */
1494 if (!strcmp(option, "udp")) {
1495 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1496 } else if (!strcmp(option, "multicast")) {
1497 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1498 } else if (!strcmp(option, "tcp")) {
1499 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1501 /* Write options back into the buffer, using memmove instead
1502 * of strcpy since the strings may overlap. */
1503 int len = strlen(option);
1504 memmove(++filename, option, len);
1506 if (option_list) *filename = '&';
1512 if (!lower_transport_mask)
1513 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1516 /* Only UDP or TCP - UDP multicast isn't supported. */
1517 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1518 (1 << RTSP_LOWER_TRANSPORT_TCP);
1519 if (!lower_transport_mask) {
1520 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1521 "only UDP and TCP are supported for output.\n");
1522 err = AVERROR(EINVAL);
1527 /* Construct the URI used in request; this is similar to s->filename,
1528 * but with authentication credentials removed and RTSP specific options
1530 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1531 host, port, "%s", path);
1533 /* open the tcp connexion */
1534 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1535 if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
1539 rt->rtsp_hd = rtsp_hd;
1542 tcp_fd = url_get_file_handle(rtsp_hd);
1543 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1544 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1545 NULL, 0, NI_NUMERICHOST);
1548 /* request options supported by the server; this also detects server
1550 for (rt->server_type = RTSP_SERVER_RTP;;) {
1552 if (rt->server_type == RTSP_SERVER_REAL)
1555 * The following entries are required for proper
1556 * streaming from a Realmedia server. They are
1557 * interdependent in some way although we currently
1558 * don't quite understand how. Values were copied
1559 * from mplayer SVN r23589.
1560 * @param CompanyID is a 16-byte ID in base64
1561 * @param ClientChallenge is a 16-byte ID in hex
1563 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1564 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1565 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1566 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1568 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1569 if (reply->status_code != RTSP_STATUS_OK) {
1570 err = AVERROR_INVALIDDATA;
1574 /* detect server type if not standard-compliant RTP */
1575 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1576 rt->server_type = RTSP_SERVER_REAL;
1578 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1579 rt->server_type = RTSP_SERVER_WMS;
1580 } else if (rt->server_type == RTSP_SERVER_REAL)
1581 strcpy(real_challenge, reply->real_challenge);
1586 err = rtsp_setup_input_streams(s, reply);
1588 err = rtsp_setup_output_streams(s, host);
1593 int lower_transport = ff_log2_tab[lower_transport_mask &
1594 ~(lower_transport_mask - 1)];
1596 err = make_setup_request(s, host, port, lower_transport,
1597 rt->server_type == RTSP_SERVER_REAL ?
1598 real_challenge : NULL);
1601 lower_transport_mask &= ~(1 << lower_transport);
1602 if (lower_transport_mask == 0 && err == 1) {
1603 err = FF_NETERROR(EPROTONOSUPPORT);
1608 rt->state = RTSP_STATE_IDLE;
1609 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1612 ff_rtsp_close_streams(s);
1613 url_close(rt->rtsp_hd);
1614 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1615 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1616 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1626 #if CONFIG_RTSP_DEMUXER
1627 static int rtsp_read_header(AVFormatContext *s,
1628 AVFormatParameters *ap)
1630 RTSPState *rt = s->priv_data;
1633 ret = ff_rtsp_connect(s);
1637 if (ap->initial_pause) {
1638 /* do not start immediately */
1640 if (rtsp_read_play(s) < 0) {
1641 ff_rtsp_close_streams(s);
1642 url_close(rt->rtsp_hd);
1643 return AVERROR_INVALIDDATA;
1650 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1651 uint8_t *buf, int buf_size)
1653 RTSPState *rt = s->priv_data;
1654 RTSPStream *rtsp_st;
1656 int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1660 if (url_interrupt_cb())
1661 return AVERROR(EINTR);
1664 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1665 FD_SET(tcp_fd, &rfds);
1670 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1671 rtsp_st = rt->rtsp_streams[i];
1672 if (rtsp_st->rtp_handle) {
1673 /* currently, we cannot probe RTCP handle because of
1674 * blocking restrictions */
1675 fd = url_get_file_handle(rtsp_st->rtp_handle);
1682 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1683 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1686 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1687 rtsp_st = rt->rtsp_streams[i];
1688 if (rtsp_st->rtp_handle) {
1689 fd = url_get_file_handle(rtsp_st->rtp_handle);
1690 if (FD_ISSET(fd, &rfds)) {
1691 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1693 *prtsp_st = rtsp_st;
1699 #if CONFIG_RTSP_DEMUXER
1700 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1701 RTSPMessageHeader reply;
1703 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1706 /* XXX: parse message */
1707 if (rt->state != RTSP_STATE_STREAMING)
1711 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1712 return FF_NETERROR(ETIMEDOUT);
1713 } else if (n < 0 && errno != EINTR)
1714 return AVERROR(errno);
1718 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1719 uint8_t *buf, int buf_size)
1721 RTSPState *rt = s->priv_data;
1722 int id, len, i, ret;
1723 RTSPStream *rtsp_st;
1725 #ifdef DEBUG_RTP_TCP
1726 dprintf(s, "tcp_read_packet:\n");
1730 RTSPMessageHeader reply;
1732 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1735 if (ret == 1) /* received '$' */
1737 /* XXX: parse message */
1738 if (rt->state != RTSP_STATE_STREAMING)
1741 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1745 len = AV_RB16(buf + 1);
1746 #ifdef DEBUG_RTP_TCP
1747 dprintf(s, "id=%d len=%d\n", id, len);
1749 if (len > buf_size || len < 12)
1752 ret = url_read_complete(rt->rtsp_hd, buf, len);
1755 if (rt->transport == RTSP_TRANSPORT_RDT &&
1756 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1759 /* find the matching stream */
1760 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1761 rtsp_st = rt->rtsp_streams[i];
1762 if (id >= rtsp_st->interleaved_min &&
1763 id <= rtsp_st->interleaved_max)
1768 *prtsp_st = rtsp_st;
1772 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1774 RTSPState *rt = s->priv_data;
1776 uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
1777 RTSPStream *rtsp_st;
1779 /* get next frames from the same RTP packet */
1780 if (rt->cur_transport_priv) {
1781 if (rt->transport == RTSP_TRANSPORT_RDT) {
1782 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1784 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1786 rt->cur_transport_priv = NULL;
1788 } else if (ret == 1) {
1791 rt->cur_transport_priv = NULL;
1794 /* read next RTP packet */
1796 switch(rt->lower_transport) {
1798 #if CONFIG_RTSP_DEMUXER
1799 case RTSP_LOWER_TRANSPORT_TCP:
1800 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1803 case RTSP_LOWER_TRANSPORT_UDP:
1804 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1805 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1806 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1807 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1814 if (rt->transport == RTSP_TRANSPORT_RDT) {
1815 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1817 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1819 /* Either bad packet, or a RTCP packet. Check if the
1820 * first_rtcp_ntp_time field was initialized. */
1821 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1822 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1823 /* first_rtcp_ntp_time has been initialized for this stream,
1824 * copy the same value to all other uninitialized streams,
1825 * in order to map their timestamp origin to the same ntp time
1828 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1829 RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
1831 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1832 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1840 /* more packets may follow, so we save the RTP context */
1841 rt->cur_transport_priv = rtsp_st->transport_priv;
1846 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1848 RTSPState *rt = s->priv_data;
1850 RTSPMessageHeader reply1, *reply = &reply1;
1853 if (rt->server_type == RTSP_SERVER_REAL) {
1855 enum AVDiscard cache[MAX_STREAMS];
1857 for (i = 0; i < s->nb_streams; i++)
1858 cache[i] = s->streams[i]->discard;
1860 if (!rt->need_subscription) {
1861 if (memcmp (cache, rt->real_setup_cache,
1862 sizeof(enum AVDiscard) * s->nb_streams)) {
1863 snprintf(cmd, sizeof(cmd),
1864 "Unsubscribe: %s\r\n",
1865 rt->last_subscription);
1866 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1868 if (reply->status_code != RTSP_STATUS_OK)
1869 return AVERROR_INVALIDDATA;
1870 rt->need_subscription = 1;
1874 if (rt->need_subscription) {
1875 int r, rule_nr, first = 1;
1877 memcpy(rt->real_setup_cache, cache,
1878 sizeof(enum AVDiscard) * s->nb_streams);
1879 rt->last_subscription[0] = 0;
1881 snprintf(cmd, sizeof(cmd),
1883 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1885 for (r = 0; r < s->nb_streams; r++) {
1886 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1887 if (s->streams[r]->discard != AVDISCARD_ALL) {
1889 av_strlcat(rt->last_subscription, ",",
1890 sizeof(rt->last_subscription));
1891 ff_rdt_subscribe_rule(
1892 rt->last_subscription,
1893 sizeof(rt->last_subscription), i, rule_nr);
1900 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1901 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1903 if (reply->status_code != RTSP_STATUS_OK)
1904 return AVERROR_INVALIDDATA;
1905 rt->need_subscription = 0;
1907 if (rt->state == RTSP_STATE_STREAMING)
1912 ret = rtsp_fetch_packet(s, pkt);
1916 /* send dummy request to keep TCP connection alive */
1917 if ((rt->server_type == RTSP_SERVER_WMS ||
1918 rt->server_type == RTSP_SERVER_REAL) &&
1919 (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1920 if (rt->server_type == RTSP_SERVER_WMS) {
1921 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1923 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1930 /* pause the stream */
1931 static int rtsp_read_pause(AVFormatContext *s)
1933 RTSPState *rt = s->priv_data;
1934 RTSPMessageHeader reply1, *reply = &reply1;
1936 if (rt->state != RTSP_STATE_STREAMING)
1938 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1939 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1940 if (reply->status_code != RTSP_STATUS_OK) {
1944 rt->state = RTSP_STATE_PAUSED;
1948 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1949 int64_t timestamp, int flags)
1951 RTSPState *rt = s->priv_data;
1953 rt->seek_timestamp = av_rescale_q(timestamp,
1954 s->streams[stream_index]->time_base,
1958 case RTSP_STATE_IDLE:
1960 case RTSP_STATE_STREAMING:
1961 if (rtsp_read_pause(s) != 0)
1963 rt->state = RTSP_STATE_SEEKING;
1964 if (rtsp_read_play(s) != 0)
1967 case RTSP_STATE_PAUSED:
1968 rt->state = RTSP_STATE_IDLE;
1974 static int rtsp_read_close(AVFormatContext *s)
1976 RTSPState *rt = s->priv_data;
1979 /* NOTE: it is valid to flush the buffer here */
1980 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1981 url_fclose(&rt->rtsp_gb);
1984 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1986 ff_rtsp_close_streams(s);
1987 url_close(rt->rtsp_hd);
1992 AVInputFormat rtsp_demuxer = {
1994 NULL_IF_CONFIG_SMALL("RTSP input format"),
2001 .flags = AVFMT_NOFILE,
2002 .read_play = rtsp_read_play,
2003 .read_pause = rtsp_read_pause,
2007 static int sdp_probe(AVProbeData *p1)
2009 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2011 /* we look for a line beginning "c=IN IP4" */
2012 while (p < p_end && *p != '\0') {
2013 if (p + sizeof("c=IN IP4") - 1 < p_end &&
2014 av_strstart(p, "c=IN IP4", NULL))
2015 return AVPROBE_SCORE_MAX / 2;
2017 while (p < p_end - 1 && *p != '\n') p++;
2026 #define SDP_MAX_SIZE 8192
2028 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
2030 RTSPState *rt = s->priv_data;
2031 RTSPStream *rtsp_st;
2036 if (!ff_network_init())
2037 return AVERROR(EIO);
2039 /* read the whole sdp file */
2040 /* XXX: better loading */
2041 content = av_malloc(SDP_MAX_SIZE);
2042 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2045 return AVERROR_INVALIDDATA;
2047 content[size] ='\0';
2049 sdp_parse(s, content);
2052 /* open each RTP stream */
2053 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2054 rtsp_st = rt->rtsp_streams[i];
2056 ff_url_join(url, sizeof(url), "rtp", NULL,
2057 inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
2058 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2060 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2061 err = AVERROR_INVALIDDATA;
2064 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2069 ff_rtsp_close_streams(s);
2074 static int sdp_read_close(AVFormatContext *s)
2076 ff_rtsp_close_streams(s);
2081 AVInputFormat sdp_demuxer = {
2083 NULL_IF_CONFIG_SMALL("SDP"),