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"
25 #include "libavutil/random_seed.h"
30 #include <sys/select.h>
35 #include "os_support.h"
41 #include "rtpdec_formats.h"
42 #include "rtpenc_chain.h"
45 //#define DEBUG_RTP_TCP
47 /* Timeout values for socket select, in ms,
48 * and read_packet(), in seconds */
49 #define SELECT_TIMEOUT_MS 100
50 #define READ_PACKET_TIMEOUT_S 10
51 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
52 #define SDP_MAX_SIZE 16384
53 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
55 static void get_word_until_chars(char *buf, int buf_size,
56 const char *sep, const char **pp)
62 p += strspn(p, SPACE_CHARS);
64 while (!strchr(sep, *p) && *p != '\0') {
65 if ((q - buf) < buf_size - 1)
74 static void get_word_sep(char *buf, int buf_size, const char *sep,
77 if (**pp == '/') (*pp)++;
78 get_word_until_chars(buf, buf_size, sep, pp);
81 static void get_word(char *buf, int buf_size, const char **pp)
83 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
86 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
88 * Used for seeking in the rtp stream.
90 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
94 p += strspn(p, SPACE_CHARS);
95 if (!av_stristart(p, "npt=", &p))
98 *start = AV_NOPTS_VALUE;
99 *end = AV_NOPTS_VALUE;
101 get_word_sep(buf, sizeof(buf), "-", &p);
102 *start = parse_date(buf, 1);
105 get_word_sep(buf, sizeof(buf), "-", &p);
106 *end = parse_date(buf, 1);
108 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
109 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
112 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
114 struct addrinfo hints, *ai = NULL;
115 memset(&hints, 0, sizeof(hints));
116 hints.ai_flags = AI_NUMERICHOST;
117 if (getaddrinfo(buf, NULL, &hints, &ai))
119 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
125 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
126 RTSPStream *rtsp_st, AVCodecContext *codec)
130 codec->codec_id = handler->codec_id;
131 rtsp_st->dynamic_handler = handler;
133 rtsp_st->dynamic_protocol_context = handler->open();
136 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
137 static int sdp_parse_rtpmap(AVFormatContext *s,
138 AVCodecContext *codec, RTSPStream *rtsp_st,
139 int payload_type, const char *p)
146 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
147 * see if we can handle this kind of payload.
148 * The space should normally not be there but some Real streams or
149 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
150 * have a trailing space. */
151 get_word_sep(buf, sizeof(buf), "/ ", &p);
152 if (payload_type >= RTP_PT_PRIVATE) {
153 RTPDynamicProtocolHandler *handler =
154 ff_rtp_handler_find_by_name(buf, codec->codec_type);
155 init_rtp_handler(handler, rtsp_st, codec);
156 /* If no dynamic handler was found, check with the list of standard
157 * allocated types, if such a stream for some reason happens to
158 * use a private payload type. This isn't handled in rtpdec.c, since
159 * the format name from the rtpmap line never is passed into rtpdec. */
160 if (!rtsp_st->dynamic_handler)
161 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
163 /* We are in a standard case
164 * (from http://www.iana.org/assignments/rtp-parameters). */
165 /* search into AVRtpPayloadTypes[] */
166 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
169 c = avcodec_find_decoder(codec->codec_id);
175 get_word_sep(buf, sizeof(buf), "/", &p);
177 switch (codec->codec_type) {
178 case AVMEDIA_TYPE_AUDIO:
179 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
180 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
181 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
183 codec->sample_rate = i;
184 get_word_sep(buf, sizeof(buf), "/", &p);
188 // TODO: there is a bug here; if it is a mono stream, and
189 // less than 22000Hz, faad upconverts to stereo and twice
190 // the frequency. No problem, but the sample rate is being
191 // set here by the sdp line. Patch on its way. (rdm)
193 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
195 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
198 case AVMEDIA_TYPE_VIDEO:
199 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
207 /* parse the attribute line from the fmtp a line of an sdp response. This
208 * is broken out as a function because it is used in rtp_h264.c, which is
210 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
211 char *value, int value_size)
213 *p += strspn(*p, SPACE_CHARS);
215 get_word_sep(attr, attr_size, "=", p);
218 get_word_sep(value, value_size, ";", p);
226 typedef struct SDPParseState {
228 struct sockaddr_storage default_ip;
230 int skip_media; ///< set if an unknown m= line occurs
233 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
234 int letter, const char *buf)
236 RTSPState *rt = s->priv_data;
237 char buf1[64], st_type[64];
239 enum AVMediaType codec_type;
243 struct sockaddr_storage sdp_ip;
246 dprintf(s, "sdp: %c='%s'\n", letter, buf);
249 if (s1->skip_media && letter != 'm')
253 get_word(buf1, sizeof(buf1), &p);
254 if (strcmp(buf1, "IN") != 0)
256 get_word(buf1, sizeof(buf1), &p);
257 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
259 get_word_sep(buf1, sizeof(buf1), "/", &p);
260 if (get_sockaddr(buf1, &sdp_ip))
265 get_word_sep(buf1, sizeof(buf1), "/", &p);
268 if (s->nb_streams == 0) {
269 s1->default_ip = sdp_ip;
270 s1->default_ttl = ttl;
272 st = s->streams[s->nb_streams - 1];
273 rtsp_st = st->priv_data;
274 rtsp_st->sdp_ip = sdp_ip;
275 rtsp_st->sdp_ttl = ttl;
279 av_metadata_set2(&s->metadata, "title", p, 0);
282 if (s->nb_streams == 0) {
283 av_metadata_set2(&s->metadata, "comment", p, 0);
290 get_word(st_type, sizeof(st_type), &p);
291 if (!strcmp(st_type, "audio")) {
292 codec_type = AVMEDIA_TYPE_AUDIO;
293 } else if (!strcmp(st_type, "video")) {
294 codec_type = AVMEDIA_TYPE_VIDEO;
295 } else if (!strcmp(st_type, "application")) {
296 codec_type = AVMEDIA_TYPE_DATA;
301 rtsp_st = av_mallocz(sizeof(RTSPStream));
304 rtsp_st->stream_index = -1;
305 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
307 rtsp_st->sdp_ip = s1->default_ip;
308 rtsp_st->sdp_ttl = s1->default_ttl;
310 get_word(buf1, sizeof(buf1), &p); /* port */
311 rtsp_st->sdp_port = atoi(buf1);
313 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
315 /* XXX: handle list of formats */
316 get_word(buf1, sizeof(buf1), &p); /* format list */
317 rtsp_st->sdp_payload_type = atoi(buf1);
319 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
320 /* no corresponding stream */
322 st = av_new_stream(s, 0);
325 st->priv_data = rtsp_st;
326 rtsp_st->stream_index = st->index;
327 st->codec->codec_type = codec_type;
328 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
329 RTPDynamicProtocolHandler *handler;
330 /* if standard payload type, we can find the codec right now */
331 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
332 /* Even static payload types may need a custom depacketizer */
333 handler = ff_rtp_handler_find_by_id(
334 rtsp_st->sdp_payload_type, st->codec->codec_type);
335 init_rtp_handler(handler, rtsp_st, st->codec);
338 /* put a default control url */
339 av_strlcpy(rtsp_st->control_url, rt->control_uri,
340 sizeof(rtsp_st->control_url));
343 if (av_strstart(p, "control:", &p)) {
344 if (s->nb_streams == 0) {
345 if (!strncmp(p, "rtsp://", 7))
346 av_strlcpy(rt->control_uri, p,
347 sizeof(rt->control_uri));
350 /* get the control url */
351 st = s->streams[s->nb_streams - 1];
352 rtsp_st = st->priv_data;
354 /* XXX: may need to add full url resolution */
355 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
357 if (proto[0] == '\0') {
358 /* relative control URL */
359 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
360 av_strlcat(rtsp_st->control_url, "/",
361 sizeof(rtsp_st->control_url));
362 av_strlcat(rtsp_st->control_url, p,
363 sizeof(rtsp_st->control_url));
365 av_strlcpy(rtsp_st->control_url, p,
366 sizeof(rtsp_st->control_url));
368 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
369 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
370 get_word(buf1, sizeof(buf1), &p);
371 payload_type = atoi(buf1);
372 st = s->streams[s->nb_streams - 1];
373 rtsp_st = st->priv_data;
374 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
375 } else if (av_strstart(p, "fmtp:", &p) ||
376 av_strstart(p, "framesize:", &p)) {
377 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
378 // let dynamic protocol handlers have a stab at the line.
379 get_word(buf1, sizeof(buf1), &p);
380 payload_type = atoi(buf1);
381 for (i = 0; i < s->nb_streams; i++) {
383 rtsp_st = st->priv_data;
384 if (rtsp_st->sdp_payload_type == payload_type &&
385 rtsp_st->dynamic_handler &&
386 rtsp_st->dynamic_handler->parse_sdp_a_line)
387 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
388 rtsp_st->dynamic_protocol_context, buf);
390 } else if (av_strstart(p, "range:", &p)) {
393 // this is so that seeking on a streamed file can work.
394 rtsp_parse_range_npt(p, &start, &end);
395 s->start_time = start;
396 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
397 s->duration = (end == AV_NOPTS_VALUE) ?
398 AV_NOPTS_VALUE : end - start;
399 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
401 rt->transport = RTSP_TRANSPORT_RDT;
402 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
404 st = s->streams[s->nb_streams - 1];
405 st->codec->sample_rate = atoi(p);
407 if (rt->server_type == RTSP_SERVER_WMS)
408 ff_wms_parse_sdp_a_line(s, p);
409 if (s->nb_streams > 0) {
410 if (rt->server_type == RTSP_SERVER_REAL)
411 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
413 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
414 if (rtsp_st->dynamic_handler &&
415 rtsp_st->dynamic_handler->parse_sdp_a_line)
416 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
418 rtsp_st->dynamic_protocol_context, buf);
425 int ff_sdp_parse(AVFormatContext *s, const char *content)
429 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
430 * contain long SDP lines containing complete ASF Headers (several
431 * kB) or arrays of MDPR (RM stream descriptor) headers plus
432 * "rulebooks" describing their properties. Therefore, the SDP line
435 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
436 * in rtpdec_xiph.c. */
438 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
440 memset(s1, 0, sizeof(SDPParseState));
443 p += strspn(p, SPACE_CHARS);
451 /* get the content */
453 while (*p != '\n' && *p != '\r' && *p != '\0') {
454 if ((q - buf) < sizeof(buf) - 1)
459 sdp_parse_line(s, s1, letter, buf);
461 while (*p != '\n' && *p != '\0')
468 #endif /* CONFIG_RTPDEC */
470 /* close and free RTSP streams */
471 void ff_rtsp_close_streams(AVFormatContext *s)
473 RTSPState *rt = s->priv_data;
477 for (i = 0; i < rt->nb_rtsp_streams; i++) {
478 rtsp_st = rt->rtsp_streams[i];
480 if (rtsp_st->transport_priv) {
482 AVFormatContext *rtpctx = rtsp_st->transport_priv;
483 av_write_trailer(rtpctx);
484 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
486 url_close_dyn_buf(rtpctx->pb, &ptr);
489 url_fclose(rtpctx->pb);
491 av_metadata_free(&rtpctx->streams[0]->metadata);
492 av_metadata_free(&rtpctx->metadata);
493 av_free(rtpctx->streams[0]);
495 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
496 ff_rdt_parse_close(rtsp_st->transport_priv);
497 else if (CONFIG_RTPDEC)
498 rtp_parse_close(rtsp_st->transport_priv);
500 if (rtsp_st->rtp_handle)
501 url_close(rtsp_st->rtp_handle);
502 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
503 rtsp_st->dynamic_handler->close(
504 rtsp_st->dynamic_protocol_context);
507 av_free(rt->rtsp_streams);
509 av_close_input_stream (rt->asf_ctx);
512 av_free(rt->recvbuf);
515 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
517 RTSPState *rt = s->priv_data;
520 /* open the RTP context */
521 if (rtsp_st->stream_index >= 0)
522 st = s->streams[rtsp_st->stream_index];
524 s->ctx_flags |= AVFMTCTX_NOHEADER;
526 if (s->oformat && CONFIG_RTSP_MUXER) {
527 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
529 RTSP_TCP_MAX_PACKET_SIZE);
530 /* Ownership of rtp_handle is passed to the rtp mux context */
531 rtsp_st->rtp_handle = NULL;
532 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
533 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
534 rtsp_st->dynamic_protocol_context,
535 rtsp_st->dynamic_handler);
536 else if (CONFIG_RTPDEC)
537 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
538 rtsp_st->sdp_payload_type,
539 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
540 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
542 if (!rtsp_st->transport_priv) {
543 return AVERROR(ENOMEM);
544 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
545 if (rtsp_st->dynamic_handler) {
546 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
547 rtsp_st->dynamic_protocol_context,
548 rtsp_st->dynamic_handler);
555 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
556 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
562 p += strspn(p, SPACE_CHARS);
563 v = strtol(p, (char **)&p, 10);
567 v = strtol(p, (char **)&p, 10);
576 /* XXX: only one transport specification is parsed */
577 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
579 char transport_protocol[16];
581 char lower_transport[16];
583 RTSPTransportField *th;
586 reply->nb_transports = 0;
589 p += strspn(p, SPACE_CHARS);
593 th = &reply->transports[reply->nb_transports];
595 get_word_sep(transport_protocol, sizeof(transport_protocol),
597 if (!strcasecmp (transport_protocol, "rtp")) {
598 get_word_sep(profile, sizeof(profile), "/;,", &p);
599 lower_transport[0] = '\0';
600 /* rtp/avp/<protocol> */
602 get_word_sep(lower_transport, sizeof(lower_transport),
605 th->transport = RTSP_TRANSPORT_RTP;
606 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
607 !strcasecmp (transport_protocol, "x-real-rdt")) {
608 /* x-pn-tng/<protocol> */
609 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
611 th->transport = RTSP_TRANSPORT_RDT;
613 if (!strcasecmp(lower_transport, "TCP"))
614 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
616 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
620 /* get each parameter */
621 while (*p != '\0' && *p != ',') {
622 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
623 if (!strcmp(parameter, "port")) {
626 rtsp_parse_range(&th->port_min, &th->port_max, &p);
628 } else if (!strcmp(parameter, "client_port")) {
631 rtsp_parse_range(&th->client_port_min,
632 &th->client_port_max, &p);
634 } else if (!strcmp(parameter, "server_port")) {
637 rtsp_parse_range(&th->server_port_min,
638 &th->server_port_max, &p);
640 } else if (!strcmp(parameter, "interleaved")) {
643 rtsp_parse_range(&th->interleaved_min,
644 &th->interleaved_max, &p);
646 } else if (!strcmp(parameter, "multicast")) {
647 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
648 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
649 } else if (!strcmp(parameter, "ttl")) {
652 th->ttl = strtol(p, (char **)&p, 10);
654 } else if (!strcmp(parameter, "destination")) {
657 get_word_sep(buf, sizeof(buf), ";,", &p);
658 get_sockaddr(buf, &th->destination);
660 } else if (!strcmp(parameter, "source")) {
663 get_word_sep(buf, sizeof(buf), ";,", &p);
664 av_strlcpy(th->source, buf, sizeof(th->source));
668 while (*p != ';' && *p != '\0' && *p != ',')
676 reply->nb_transports++;
680 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
681 HTTPAuthState *auth_state)
685 /* NOTE: we do case independent match for broken servers */
687 if (av_stristart(p, "Session:", &p)) {
689 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
690 if (av_stristart(p, ";timeout=", &p) &&
691 (t = strtol(p, NULL, 10)) > 0) {
694 } else if (av_stristart(p, "Content-Length:", &p)) {
695 reply->content_length = strtol(p, NULL, 10);
696 } else if (av_stristart(p, "Transport:", &p)) {
697 rtsp_parse_transport(reply, p);
698 } else if (av_stristart(p, "CSeq:", &p)) {
699 reply->seq = strtol(p, NULL, 10);
700 } else if (av_stristart(p, "Range:", &p)) {
701 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
702 } else if (av_stristart(p, "RealChallenge1:", &p)) {
703 p += strspn(p, SPACE_CHARS);
704 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
705 } else if (av_stristart(p, "Server:", &p)) {
706 p += strspn(p, SPACE_CHARS);
707 av_strlcpy(reply->server, p, sizeof(reply->server));
708 } else if (av_stristart(p, "Notice:", &p) ||
709 av_stristart(p, "X-Notice:", &p)) {
710 reply->notice = strtol(p, NULL, 10);
711 } else if (av_stristart(p, "Location:", &p)) {
712 p += strspn(p, SPACE_CHARS);
713 av_strlcpy(reply->location, p , sizeof(reply->location));
714 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
715 p += strspn(p, SPACE_CHARS);
716 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
717 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
718 p += strspn(p, SPACE_CHARS);
719 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
720 } else if (av_stristart(p, "Content-Base:", &p)) {
721 p += strspn(p, SPACE_CHARS);
722 av_strlcpy(reply->content_base, p , sizeof(reply->content_base));
726 /* skip a RTP/TCP interleaved packet */
727 void ff_rtsp_skip_packet(AVFormatContext *s)
729 RTSPState *rt = s->priv_data;
733 ret = url_read_complete(rt->rtsp_hd, buf, 3);
736 len = AV_RB16(buf + 1);
738 dprintf(s, "skipping RTP packet len=%d\n", len);
743 if (len1 > sizeof(buf))
745 ret = url_read_complete(rt->rtsp_hd, buf, len1);
752 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
753 unsigned char **content_ptr,
754 int return_on_interleaved_data)
756 RTSPState *rt = s->priv_data;
757 char buf[4096], buf1[1024], *q;
760 int ret, content_length, line_count = 0;
761 unsigned char *content = NULL;
763 memset(reply, 0, sizeof(*reply));
765 /* parse reply (XXX: use buffers) */
766 rt->last_reply[0] = '\0';
770 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
772 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
779 /* XXX: only parse it if first char on line ? */
780 if (return_on_interleaved_data) {
783 ff_rtsp_skip_packet(s);
784 } else if (ch != '\r') {
785 if ((q - buf) < sizeof(buf) - 1)
791 dprintf(s, "line='%s'\n", buf);
793 /* test if last line */
797 if (line_count == 0) {
799 get_word(buf1, sizeof(buf1), &p);
800 get_word(buf1, sizeof(buf1), &p);
801 reply->status_code = atoi(buf1);
802 av_strlcpy(reply->reason, p, sizeof(reply->reason));
804 ff_rtsp_parse_line(reply, p, &rt->auth_state);
805 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
806 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
811 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
812 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
814 content_length = reply->content_length;
815 if (content_length > 0) {
816 /* leave some room for a trailing '\0' (useful for simple parsing) */
817 content = av_malloc(content_length + 1);
818 (void)url_read_complete(rt->rtsp_hd, content, content_length);
819 content[content_length] = '\0';
822 *content_ptr = content;
826 if (rt->seq != reply->seq) {
827 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
828 rt->seq, reply->seq);
832 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
833 reply->notice == 2104 /* Start-of-Stream Reached */ ||
834 reply->notice == 2306 /* Continuous Feed Terminated */) {
835 rt->state = RTSP_STATE_IDLE;
836 } else if (reply->notice >= 4400 && reply->notice < 5500) {
837 return AVERROR(EIO); /* data or server error */
838 } else if (reply->notice == 2401 /* Ticket Expired */ ||
839 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
840 return AVERROR(EPERM);
845 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
846 const char *method, const char *url,
848 const unsigned char *send_content,
849 int send_content_length)
851 RTSPState *rt = s->priv_data;
852 char buf[4096], *out_buf;
853 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
855 /* Add in RTSP headers */
858 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
860 av_strlcat(buf, headers, sizeof(buf));
861 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
862 if (rt->session_id[0] != '\0' && (!headers ||
863 !strstr(headers, "\nIf-Match:"))) {
864 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
867 char *str = ff_http_auth_create_response(&rt->auth_state,
868 rt->auth, url, method);
870 av_strlcat(buf, str, sizeof(buf));
873 if (send_content_length > 0 && send_content)
874 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
875 av_strlcat(buf, "\r\n", sizeof(buf));
877 /* base64 encode rtsp if tunneling */
878 if (rt->control_transport == RTSP_MODE_TUNNEL) {
879 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
883 dprintf(s, "Sending:\n%s--\n", buf);
885 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
886 if (send_content_length > 0 && send_content) {
887 if (rt->control_transport == RTSP_MODE_TUNNEL) {
888 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
889 "with content data not supported\n");
890 return AVERROR_PATCHWELCOME;
892 url_write(rt->rtsp_hd_out, send_content, send_content_length);
894 rt->last_cmd_time = av_gettime();
899 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
900 const char *url, const char *headers)
902 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
905 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
906 const char *headers, RTSPMessageHeader *reply,
907 unsigned char **content_ptr)
909 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
910 content_ptr, NULL, 0);
913 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
914 const char *method, const char *url,
916 RTSPMessageHeader *reply,
917 unsigned char **content_ptr,
918 const unsigned char *send_content,
919 int send_content_length)
921 RTSPState *rt = s->priv_data;
922 HTTPAuthType cur_auth_type;
926 cur_auth_type = rt->auth_state.auth_type;
927 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
929 send_content_length)))
932 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
935 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
936 rt->auth_state.auth_type != HTTP_AUTH_NONE)
939 if (reply->status_code > 400){
940 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
944 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
951 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
953 static int make_setup_request(AVFormatContext *s, const char *host, int port,
954 int lower_transport, const char *real_challenge)
956 RTSPState *rt = s->priv_data;
957 int rtx, j, i, err, interleave = 0;
959 RTSPMessageHeader reply1, *reply = &reply1;
961 const char *trans_pref;
963 if (rt->transport == RTSP_TRANSPORT_RDT)
964 trans_pref = "x-pn-tng";
966 trans_pref = "RTP/AVP";
968 /* default timeout: 1 minute */
971 /* for each stream, make the setup request */
972 /* XXX: we assume the same server is used for the control of each
975 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
976 char transport[2048];
979 * WMS serves all UDP data over a single connection, the RTX, which
980 * isn't necessarily the first in the SDP but has to be the first
981 * to be set up, else the second/third SETUP will fail with a 461.
983 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
984 rt->server_type == RTSP_SERVER_WMS) {
987 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
988 int len = strlen(rt->rtsp_streams[rtx]->control_url);
990 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
994 if (rtx == rt->nb_rtsp_streams)
995 return -1; /* no RTX found */
996 rtsp_st = rt->rtsp_streams[rtx];
998 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1000 rtsp_st = rt->rtsp_streams[i];
1003 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1006 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1007 port = reply->transports[0].client_port_min;
1011 /* first try in specified port range */
1012 if (RTSP_RTP_PORT_MIN != 0) {
1013 while (j <= RTSP_RTP_PORT_MAX) {
1014 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1015 "?localport=%d", j);
1016 /* we will use two ports per rtp stream (rtp and rtcp) */
1018 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1024 /* then try on any port */
1025 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1026 err = AVERROR_INVALIDDATA;
1032 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1034 snprintf(transport, sizeof(transport) - 1,
1035 "%s/UDP;", trans_pref);
1036 if (rt->server_type != RTSP_SERVER_REAL)
1037 av_strlcat(transport, "unicast;", sizeof(transport));
1038 av_strlcatf(transport, sizeof(transport),
1039 "client_port=%d", port);
1040 if (rt->transport == RTSP_TRANSPORT_RTP &&
1041 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1042 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1046 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1047 /** For WMS streams, the application streams are only used for
1048 * UDP. When trying to set it up for TCP streams, the server
1049 * will return an error. Therefore, we skip those streams. */
1050 if (rt->server_type == RTSP_SERVER_WMS &&
1051 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1054 snprintf(transport, sizeof(transport) - 1,
1055 "%s/TCP;", trans_pref);
1056 if (rt->server_type == RTSP_SERVER_WMS)
1057 av_strlcat(transport, "unicast;", sizeof(transport));
1058 av_strlcatf(transport, sizeof(transport),
1059 "interleaved=%d-%d",
1060 interleave, interleave + 1);
1064 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1065 snprintf(transport, sizeof(transport) - 1,
1066 "%s/UDP;multicast", trans_pref);
1069 av_strlcat(transport, ";mode=receive", sizeof(transport));
1070 } else if (rt->server_type == RTSP_SERVER_REAL ||
1071 rt->server_type == RTSP_SERVER_WMS)
1072 av_strlcat(transport, ";mode=play", sizeof(transport));
1073 snprintf(cmd, sizeof(cmd),
1074 "Transport: %s\r\n",
1076 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1077 char real_res[41], real_csum[9];
1078 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1080 av_strlcatf(cmd, sizeof(cmd),
1082 "RealChallenge2: %s, sd=%s\r\n",
1083 rt->session_id, real_res, real_csum);
1085 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1086 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1089 } else if (reply->status_code != RTSP_STATUS_OK ||
1090 reply->nb_transports != 1) {
1091 err = AVERROR_INVALIDDATA;
1095 /* XXX: same protocol for all streams is required */
1097 if (reply->transports[0].lower_transport != rt->lower_transport ||
1098 reply->transports[0].transport != rt->transport) {
1099 err = AVERROR_INVALIDDATA;
1103 rt->lower_transport = reply->transports[0].lower_transport;
1104 rt->transport = reply->transports[0].transport;
1107 /* close RTP connection if not chosen */
1108 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1109 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1110 url_close(rtsp_st->rtp_handle);
1111 rtsp_st->rtp_handle = NULL;
1114 switch(reply->transports[0].lower_transport) {
1115 case RTSP_LOWER_TRANSPORT_TCP:
1116 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1117 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1120 case RTSP_LOWER_TRANSPORT_UDP: {
1123 /* Use source address if specified */
1124 if (reply->transports[0].source[0]) {
1125 ff_url_join(url, sizeof(url), "rtp", NULL,
1126 reply->transports[0].source,
1127 reply->transports[0].server_port_min, NULL);
1129 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1130 reply->transports[0].server_port_min, NULL);
1132 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1133 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1134 err = AVERROR_INVALIDDATA;
1137 /* Try to initialize the connection state in a
1138 * potential NAT router by sending dummy packets.
1139 * RTP/RTCP dummy packets are used for RDT, too.
1141 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1143 rtp_send_punch_packets(rtsp_st->rtp_handle);
1146 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1147 char url[1024], namebuf[50];
1148 struct sockaddr_storage addr;
1151 if (reply->transports[0].destination.ss_family) {
1152 addr = reply->transports[0].destination;
1153 port = reply->transports[0].port_min;
1154 ttl = reply->transports[0].ttl;
1156 addr = rtsp_st->sdp_ip;
1157 port = rtsp_st->sdp_port;
1158 ttl = rtsp_st->sdp_ttl;
1160 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1161 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1162 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1163 port, "?ttl=%d", ttl);
1164 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1165 err = AVERROR_INVALIDDATA;
1172 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1176 if (reply->timeout > 0)
1177 rt->timeout = reply->timeout;
1179 if (rt->server_type == RTSP_SERVER_REAL)
1180 rt->need_subscription = 1;
1185 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1186 if (rt->rtsp_streams[i]->rtp_handle) {
1187 url_close(rt->rtsp_streams[i]->rtp_handle);
1188 rt->rtsp_streams[i]->rtp_handle = NULL;
1194 void ff_rtsp_close_connections(AVFormatContext *s)
1196 RTSPState *rt = s->priv_data;
1197 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1198 url_close(rt->rtsp_hd);
1199 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1202 int ff_rtsp_connect(AVFormatContext *s)
1204 RTSPState *rt = s->priv_data;
1205 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1206 char *option_list, *option, *filename;
1207 int port, err, tcp_fd;
1208 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1209 int lower_transport_mask = 0;
1210 char real_challenge[64];
1211 struct sockaddr_storage peer;
1212 socklen_t peer_len = sizeof(peer);
1214 if (!ff_network_init())
1215 return AVERROR(EIO);
1217 rt->control_transport = RTSP_MODE_PLAIN;
1218 /* extract hostname and port */
1219 av_url_split(NULL, 0, auth, sizeof(auth),
1220 host, sizeof(host), &port, path, sizeof(path), s->filename);
1222 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1225 port = RTSP_DEFAULT_PORT;
1227 /* search for options */
1228 option_list = strrchr(path, '?');
1230 /* Strip out the RTSP specific options, write out the rest of
1231 * the options back into the same string. */
1232 filename = option_list;
1233 while (option_list) {
1234 /* move the option pointer */
1235 option = ++option_list;
1236 option_list = strchr(option_list, '&');
1240 /* handle the options */
1241 if (!strcmp(option, "udp")) {
1242 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1243 } else if (!strcmp(option, "multicast")) {
1244 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1245 } else if (!strcmp(option, "tcp")) {
1246 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1247 } else if(!strcmp(option, "http")) {
1248 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1249 rt->control_transport = RTSP_MODE_TUNNEL;
1251 /* Write options back into the buffer, using memmove instead
1252 * of strcpy since the strings may overlap. */
1253 int len = strlen(option);
1254 memmove(++filename, option, len);
1256 if (option_list) *filename = '&';
1262 if (!lower_transport_mask)
1263 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1266 /* Only UDP or TCP - UDP multicast isn't supported. */
1267 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1268 (1 << RTSP_LOWER_TRANSPORT_TCP);
1269 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1270 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1271 "only UDP and TCP are supported for output.\n");
1272 err = AVERROR(EINVAL);
1277 /* Construct the URI used in request; this is similar to s->filename,
1278 * but with authentication credentials removed and RTSP specific options
1280 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1281 host, port, "%s", path);
1283 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1284 /* set up initial handshake for tunneling */
1285 char httpname[1024];
1286 char sessioncookie[17];
1289 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1290 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1291 av_get_random_seed(), av_get_random_seed());
1294 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1299 /* generate GET headers */
1300 snprintf(headers, sizeof(headers),
1301 "x-sessioncookie: %s\r\n"
1302 "Accept: application/x-rtsp-tunnelled\r\n"
1303 "Pragma: no-cache\r\n"
1304 "Cache-Control: no-cache\r\n",
1306 ff_http_set_headers(rt->rtsp_hd, headers);
1308 /* complete the connection */
1309 if (url_connect(rt->rtsp_hd)) {
1315 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1320 /* generate POST headers */
1321 snprintf(headers, sizeof(headers),
1322 "x-sessioncookie: %s\r\n"
1323 "Content-Type: application/x-rtsp-tunnelled\r\n"
1324 "Pragma: no-cache\r\n"
1325 "Cache-Control: no-cache\r\n"
1326 "Content-Length: 32767\r\n"
1327 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1329 ff_http_set_headers(rt->rtsp_hd_out, headers);
1330 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1332 /* Initialize the authentication state for the POST session. The HTTP
1333 * protocol implementation doesn't properly handle multi-pass
1334 * authentication for POST requests, since it would require one of
1336 * - implementing Expect: 100-continue, which many HTTP servers
1337 * don't support anyway, even less the RTSP servers that do HTTP
1339 * - sending the whole POST data until getting a 401 reply specifying
1340 * what authentication method to use, then resending all that data
1341 * - waiting for potential 401 replies directly after sending the
1342 * POST header (waiting for some unspecified time)
1343 * Therefore, we copy the full auth state, which works for both basic
1344 * and digest. (For digest, we would have to synchronize the nonce
1345 * count variable between the two sessions, if we'd do more requests
1346 * with the original session, though.)
1348 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1350 /* complete the connection */
1351 if (url_connect(rt->rtsp_hd_out)) {
1356 /* open the tcp connection */
1357 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1358 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1362 rt->rtsp_hd_out = rt->rtsp_hd;
1366 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1367 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1368 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1369 NULL, 0, NI_NUMERICHOST);
1372 /* request options supported by the server; this also detects server
1374 for (rt->server_type = RTSP_SERVER_RTP;;) {
1376 if (rt->server_type == RTSP_SERVER_REAL)
1379 * The following entries are required for proper
1380 * streaming from a Realmedia server. They are
1381 * interdependent in some way although we currently
1382 * don't quite understand how. Values were copied
1383 * from mplayer SVN r23589.
1384 * @param CompanyID is a 16-byte ID in base64
1385 * @param ClientChallenge is a 16-byte ID in hex
1387 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1388 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1389 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1390 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1392 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1393 if (reply->status_code != RTSP_STATUS_OK) {
1394 err = AVERROR_INVALIDDATA;
1398 /* detect server type if not standard-compliant RTP */
1399 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1400 rt->server_type = RTSP_SERVER_REAL;
1402 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1403 rt->server_type = RTSP_SERVER_WMS;
1404 } else if (rt->server_type == RTSP_SERVER_REAL)
1405 strcpy(real_challenge, reply->real_challenge);
1409 if (s->iformat && CONFIG_RTSP_DEMUXER)
1410 err = ff_rtsp_setup_input_streams(s, reply);
1411 else if (CONFIG_RTSP_MUXER)
1412 err = ff_rtsp_setup_output_streams(s, host);
1417 int lower_transport = ff_log2_tab[lower_transport_mask &
1418 ~(lower_transport_mask - 1)];
1420 err = make_setup_request(s, host, port, lower_transport,
1421 rt->server_type == RTSP_SERVER_REAL ?
1422 real_challenge : NULL);
1425 lower_transport_mask &= ~(1 << lower_transport);
1426 if (lower_transport_mask == 0 && err == 1) {
1427 err = FF_NETERROR(EPROTONOSUPPORT);
1432 rt->state = RTSP_STATE_IDLE;
1433 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1436 ff_rtsp_close_streams(s);
1437 ff_rtsp_close_connections(s);
1438 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1439 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1440 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1448 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1451 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1452 uint8_t *buf, int buf_size, int64_t wait_end)
1454 RTSPState *rt = s->priv_data;
1455 RTSPStream *rtsp_st;
1457 int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1461 if (url_interrupt_cb())
1462 return AVERROR(EINTR);
1463 if (wait_end && wait_end - av_gettime() < 0)
1464 return AVERROR(EAGAIN);
1467 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1468 FD_SET(tcp_fd, &rfds);
1473 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1474 rtsp_st = rt->rtsp_streams[i];
1475 if (rtsp_st->rtp_handle) {
1476 fd = url_get_file_handle(rtsp_st->rtp_handle);
1477 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1478 if (FFMAX(fd, fd_rtcp) > fd_max)
1479 fd_max = FFMAX(fd, fd_rtcp);
1481 FD_SET(fd_rtcp, &rfds);
1485 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1486 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1489 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1490 rtsp_st = rt->rtsp_streams[i];
1491 if (rtsp_st->rtp_handle) {
1492 fd = url_get_file_handle(rtsp_st->rtp_handle);
1493 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1494 if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
1495 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1497 *prtsp_st = rtsp_st;
1503 #if CONFIG_RTSP_DEMUXER
1504 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1505 RTSPMessageHeader reply;
1507 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1510 /* XXX: parse message */
1511 if (rt->state != RTSP_STATE_STREAMING)
1515 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1516 return FF_NETERROR(ETIMEDOUT);
1517 } else if (n < 0 && errno != EINTR)
1518 return AVERROR(errno);
1522 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1524 RTSPState *rt = s->priv_data;
1526 RTSPStream *rtsp_st, *first_queue_st = NULL;
1527 int64_t wait_end = 0;
1529 if (rt->nb_byes == rt->nb_rtsp_streams)
1532 /* get next frames from the same RTP packet */
1533 if (rt->cur_transport_priv) {
1534 if (rt->transport == RTSP_TRANSPORT_RDT) {
1535 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1537 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1539 rt->cur_transport_priv = NULL;
1541 } else if (ret == 1) {
1544 rt->cur_transport_priv = NULL;
1547 if (rt->transport == RTSP_TRANSPORT_RTP) {
1549 int64_t first_queue_time = 0;
1550 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1551 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1552 int64_t queue_time = ff_rtp_queued_packet_time(rtpctx);
1553 if (queue_time && (queue_time - first_queue_time < 0 ||
1554 !first_queue_time)) {
1555 first_queue_time = queue_time;
1556 first_queue_st = rt->rtsp_streams[i];
1559 if (first_queue_time)
1560 wait_end = first_queue_time + s->max_delay;
1563 /* read next RTP packet */
1566 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1568 return AVERROR(ENOMEM);
1571 switch(rt->lower_transport) {
1573 #if CONFIG_RTSP_DEMUXER
1574 case RTSP_LOWER_TRANSPORT_TCP:
1575 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1578 case RTSP_LOWER_TRANSPORT_UDP:
1579 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1580 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1581 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1582 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1585 if (len == AVERROR(EAGAIN) && first_queue_st &&
1586 rt->transport == RTSP_TRANSPORT_RTP) {
1587 rtsp_st = first_queue_st;
1588 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1595 if (rt->transport == RTSP_TRANSPORT_RDT) {
1596 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1598 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1600 /* Either bad packet, or a RTCP packet. Check if the
1601 * first_rtcp_ntp_time field was initialized. */
1602 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1603 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1604 /* first_rtcp_ntp_time has been initialized for this stream,
1605 * copy the same value to all other uninitialized streams,
1606 * in order to map their timestamp origin to the same ntp time
1609 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1610 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1612 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1613 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1616 if (ret == -RTCP_BYE) {
1619 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1620 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1622 if (rt->nb_byes == rt->nb_rtsp_streams)
1631 /* more packets may follow, so we save the RTP context */
1632 rt->cur_transport_priv = rtsp_st->transport_priv;
1636 #endif /* CONFIG_RTPDEC */
1638 #if CONFIG_SDP_DEMUXER
1639 static int sdp_probe(AVProbeData *p1)
1641 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1643 /* we look for a line beginning "c=IN IP" */
1644 while (p < p_end && *p != '\0') {
1645 if (p + sizeof("c=IN IP") - 1 < p_end &&
1646 av_strstart(p, "c=IN IP", NULL))
1647 return AVPROBE_SCORE_MAX / 2;
1649 while (p < p_end - 1 && *p != '\n') p++;
1658 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1660 RTSPState *rt = s->priv_data;
1661 RTSPStream *rtsp_st;
1666 if (!ff_network_init())
1667 return AVERROR(EIO);
1669 /* read the whole sdp file */
1670 /* XXX: better loading */
1671 content = av_malloc(SDP_MAX_SIZE);
1672 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1675 return AVERROR_INVALIDDATA;
1677 content[size] ='\0';
1679 ff_sdp_parse(s, content);
1682 /* open each RTP stream */
1683 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1685 rtsp_st = rt->rtsp_streams[i];
1687 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1688 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1689 ff_url_join(url, sizeof(url), "rtp", NULL,
1690 namebuf, rtsp_st->sdp_port,
1691 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1693 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1694 err = AVERROR_INVALIDDATA;
1697 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1702 ff_rtsp_close_streams(s);
1707 static int sdp_read_close(AVFormatContext *s)
1709 ff_rtsp_close_streams(s);
1714 AVInputFormat sdp_demuxer = {
1716 NULL_IF_CONFIG_SMALL("SDP"),
1720 ff_rtsp_fetch_packet,
1723 #endif /* CONFIG_SDP_DEMUXER */
1725 #if CONFIG_RTP_DEMUXER
1726 static int rtp_probe(AVProbeData *p)
1728 if (av_strstart(p->filename, "rtp:", NULL))
1729 return AVPROBE_SCORE_MAX;
1733 static int rtp_read_header(AVFormatContext *s,
1734 AVFormatParameters *ap)
1736 uint8_t recvbuf[1500];
1737 char host[500], sdp[500];
1739 URLContext* in = NULL;
1741 AVCodecContext codec;
1742 struct sockaddr_storage addr;
1744 socklen_t addrlen = sizeof(addr);
1746 if (!ff_network_init())
1747 return AVERROR(EIO);
1749 ret = url_open(&in, s->filename, URL_RDONLY);
1754 ret = url_read(in, recvbuf, sizeof(recvbuf));
1755 if (ret == AVERROR(EAGAIN))
1760 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1764 if ((recvbuf[0] & 0xc0) != 0x80) {
1765 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1770 payload_type = recvbuf[1] & 0x7f;
1773 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1777 memset(&codec, 0, sizeof(codec));
1778 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1779 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1780 "without an SDP file describing it\n",
1784 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1785 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1786 "properly you need an SDP file "
1790 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1791 NULL, 0, s->filename);
1793 snprintf(sdp, sizeof(sdp),
1794 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1795 addr.ss_family == AF_INET ? 4 : 6, host,
1796 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1797 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1798 port, payload_type);
1799 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1801 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1804 /* sdp_read_header initializes this again */
1807 ret = sdp_read_header(s, ap);
1818 AVInputFormat rtp_demuxer = {
1820 NULL_IF_CONFIG_SMALL("RTP input format"),
1824 ff_rtsp_fetch_packet,
1826 .flags = AVFMT_NOFILE,
1828 #endif /* CONFIG_RTP_DEMUXER */