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 /* if standard payload type, we can find the codec right now */
330 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
333 /* put a default control url */
334 av_strlcpy(rtsp_st->control_url, rt->control_uri,
335 sizeof(rtsp_st->control_url));
338 if (av_strstart(p, "control:", &p)) {
339 if (s->nb_streams == 0) {
340 if (!strncmp(p, "rtsp://", 7))
341 av_strlcpy(rt->control_uri, p,
342 sizeof(rt->control_uri));
345 /* get the control url */
346 st = s->streams[s->nb_streams - 1];
347 rtsp_st = st->priv_data;
349 /* XXX: may need to add full url resolution */
350 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
352 if (proto[0] == '\0') {
353 /* relative control URL */
354 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
355 av_strlcat(rtsp_st->control_url, "/",
356 sizeof(rtsp_st->control_url));
357 av_strlcat(rtsp_st->control_url, p,
358 sizeof(rtsp_st->control_url));
360 av_strlcpy(rtsp_st->control_url, p,
361 sizeof(rtsp_st->control_url));
363 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
364 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
365 get_word(buf1, sizeof(buf1), &p);
366 payload_type = atoi(buf1);
367 st = s->streams[s->nb_streams - 1];
368 rtsp_st = st->priv_data;
369 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
370 } else if (av_strstart(p, "fmtp:", &p) ||
371 av_strstart(p, "framesize:", &p)) {
372 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
373 // let dynamic protocol handlers have a stab at the line.
374 get_word(buf1, sizeof(buf1), &p);
375 payload_type = atoi(buf1);
376 for (i = 0; i < s->nb_streams; i++) {
378 rtsp_st = st->priv_data;
379 if (rtsp_st->sdp_payload_type == payload_type &&
380 rtsp_st->dynamic_handler &&
381 rtsp_st->dynamic_handler->parse_sdp_a_line)
382 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
383 rtsp_st->dynamic_protocol_context, buf);
385 } else if (av_strstart(p, "range:", &p)) {
388 // this is so that seeking on a streamed file can work.
389 rtsp_parse_range_npt(p, &start, &end);
390 s->start_time = start;
391 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
392 s->duration = (end == AV_NOPTS_VALUE) ?
393 AV_NOPTS_VALUE : end - start;
394 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
396 rt->transport = RTSP_TRANSPORT_RDT;
398 if (rt->server_type == RTSP_SERVER_WMS)
399 ff_wms_parse_sdp_a_line(s, p);
400 if (s->nb_streams > 0) {
401 if (rt->server_type == RTSP_SERVER_REAL)
402 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
404 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
405 if (rtsp_st->dynamic_handler &&
406 rtsp_st->dynamic_handler->parse_sdp_a_line)
407 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
409 rtsp_st->dynamic_protocol_context, buf);
416 int ff_sdp_parse(AVFormatContext *s, const char *content)
420 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
421 * contain long SDP lines containing complete ASF Headers (several
422 * kB) or arrays of MDPR (RM stream descriptor) headers plus
423 * "rulebooks" describing their properties. Therefore, the SDP line
426 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
427 * in rtpdec_xiph.c. */
429 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
431 memset(s1, 0, sizeof(SDPParseState));
434 p += strspn(p, SPACE_CHARS);
442 /* get the content */
444 while (*p != '\n' && *p != '\r' && *p != '\0') {
445 if ((q - buf) < sizeof(buf) - 1)
450 sdp_parse_line(s, s1, letter, buf);
452 while (*p != '\n' && *p != '\0')
459 #endif /* CONFIG_RTPDEC */
461 /* close and free RTSP streams */
462 void ff_rtsp_close_streams(AVFormatContext *s)
464 RTSPState *rt = s->priv_data;
468 for (i = 0; i < rt->nb_rtsp_streams; i++) {
469 rtsp_st = rt->rtsp_streams[i];
471 if (rtsp_st->transport_priv) {
473 AVFormatContext *rtpctx = rtsp_st->transport_priv;
474 av_write_trailer(rtpctx);
475 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
477 url_close_dyn_buf(rtpctx->pb, &ptr);
480 url_fclose(rtpctx->pb);
482 av_metadata_free(&rtpctx->streams[0]->metadata);
483 av_metadata_free(&rtpctx->metadata);
484 av_free(rtpctx->streams[0]);
486 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
487 ff_rdt_parse_close(rtsp_st->transport_priv);
488 else if (CONFIG_RTPDEC)
489 rtp_parse_close(rtsp_st->transport_priv);
491 if (rtsp_st->rtp_handle)
492 url_close(rtsp_st->rtp_handle);
493 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
494 rtsp_st->dynamic_handler->close(
495 rtsp_st->dynamic_protocol_context);
498 av_free(rt->rtsp_streams);
500 av_close_input_stream (rt->asf_ctx);
503 av_free(rt->recvbuf);
506 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
508 RTSPState *rt = s->priv_data;
511 /* open the RTP context */
512 if (rtsp_st->stream_index >= 0)
513 st = s->streams[rtsp_st->stream_index];
515 s->ctx_flags |= AVFMTCTX_NOHEADER;
517 if (s->oformat && CONFIG_RTSP_MUXER) {
518 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
520 RTSP_TCP_MAX_PACKET_SIZE);
521 /* Ownership of rtp_handle is passed to the rtp mux context */
522 rtsp_st->rtp_handle = NULL;
523 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
524 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
525 rtsp_st->dynamic_protocol_context,
526 rtsp_st->dynamic_handler);
527 else if (CONFIG_RTPDEC)
528 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
529 rtsp_st->sdp_payload_type,
530 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
531 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
533 if (!rtsp_st->transport_priv) {
534 return AVERROR(ENOMEM);
535 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
536 if (rtsp_st->dynamic_handler) {
537 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
538 rtsp_st->dynamic_protocol_context,
539 rtsp_st->dynamic_handler);
546 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
547 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
553 p += strspn(p, SPACE_CHARS);
554 v = strtol(p, (char **)&p, 10);
558 v = strtol(p, (char **)&p, 10);
567 /* XXX: only one transport specification is parsed */
568 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
570 char transport_protocol[16];
572 char lower_transport[16];
574 RTSPTransportField *th;
577 reply->nb_transports = 0;
580 p += strspn(p, SPACE_CHARS);
584 th = &reply->transports[reply->nb_transports];
586 get_word_sep(transport_protocol, sizeof(transport_protocol),
588 if (!strcasecmp (transport_protocol, "rtp")) {
589 get_word_sep(profile, sizeof(profile), "/;,", &p);
590 lower_transport[0] = '\0';
591 /* rtp/avp/<protocol> */
593 get_word_sep(lower_transport, sizeof(lower_transport),
596 th->transport = RTSP_TRANSPORT_RTP;
597 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
598 !strcasecmp (transport_protocol, "x-real-rdt")) {
599 /* x-pn-tng/<protocol> */
600 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
602 th->transport = RTSP_TRANSPORT_RDT;
604 if (!strcasecmp(lower_transport, "TCP"))
605 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
607 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
611 /* get each parameter */
612 while (*p != '\0' && *p != ',') {
613 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
614 if (!strcmp(parameter, "port")) {
617 rtsp_parse_range(&th->port_min, &th->port_max, &p);
619 } else if (!strcmp(parameter, "client_port")) {
622 rtsp_parse_range(&th->client_port_min,
623 &th->client_port_max, &p);
625 } else if (!strcmp(parameter, "server_port")) {
628 rtsp_parse_range(&th->server_port_min,
629 &th->server_port_max, &p);
631 } else if (!strcmp(parameter, "interleaved")) {
634 rtsp_parse_range(&th->interleaved_min,
635 &th->interleaved_max, &p);
637 } else if (!strcmp(parameter, "multicast")) {
638 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
639 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
640 } else if (!strcmp(parameter, "ttl")) {
643 th->ttl = strtol(p, (char **)&p, 10);
645 } else if (!strcmp(parameter, "destination")) {
648 get_word_sep(buf, sizeof(buf), ";,", &p);
649 get_sockaddr(buf, &th->destination);
651 } else if (!strcmp(parameter, "source")) {
654 get_word_sep(buf, sizeof(buf), ";,", &p);
655 av_strlcpy(th->source, buf, sizeof(th->source));
659 while (*p != ';' && *p != '\0' && *p != ',')
667 reply->nb_transports++;
671 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
672 HTTPAuthState *auth_state)
676 /* NOTE: we do case independent match for broken servers */
678 if (av_stristart(p, "Session:", &p)) {
680 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
681 if (av_stristart(p, ";timeout=", &p) &&
682 (t = strtol(p, NULL, 10)) > 0) {
685 } else if (av_stristart(p, "Content-Length:", &p)) {
686 reply->content_length = strtol(p, NULL, 10);
687 } else if (av_stristart(p, "Transport:", &p)) {
688 rtsp_parse_transport(reply, p);
689 } else if (av_stristart(p, "CSeq:", &p)) {
690 reply->seq = strtol(p, NULL, 10);
691 } else if (av_stristart(p, "Range:", &p)) {
692 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
693 } else if (av_stristart(p, "RealChallenge1:", &p)) {
694 p += strspn(p, SPACE_CHARS);
695 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
696 } else if (av_stristart(p, "Server:", &p)) {
697 p += strspn(p, SPACE_CHARS);
698 av_strlcpy(reply->server, p, sizeof(reply->server));
699 } else if (av_stristart(p, "Notice:", &p) ||
700 av_stristart(p, "X-Notice:", &p)) {
701 reply->notice = strtol(p, NULL, 10);
702 } else if (av_stristart(p, "Location:", &p)) {
703 p += strspn(p, SPACE_CHARS);
704 av_strlcpy(reply->location, p , sizeof(reply->location));
705 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
706 p += strspn(p, SPACE_CHARS);
707 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
708 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
709 p += strspn(p, SPACE_CHARS);
710 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
711 } else if (av_stristart(p, "Content-Base:", &p)) {
712 p += strspn(p, SPACE_CHARS);
713 av_strlcpy(reply->content_base, p , sizeof(reply->content_base));
717 /* skip a RTP/TCP interleaved packet */
718 void ff_rtsp_skip_packet(AVFormatContext *s)
720 RTSPState *rt = s->priv_data;
724 ret = url_read_complete(rt->rtsp_hd, buf, 3);
727 len = AV_RB16(buf + 1);
729 dprintf(s, "skipping RTP packet len=%d\n", len);
734 if (len1 > sizeof(buf))
736 ret = url_read_complete(rt->rtsp_hd, buf, len1);
743 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
744 unsigned char **content_ptr,
745 int return_on_interleaved_data)
747 RTSPState *rt = s->priv_data;
748 char buf[4096], buf1[1024], *q;
751 int ret, content_length, line_count = 0;
752 unsigned char *content = NULL;
754 memset(reply, 0, sizeof(*reply));
756 /* parse reply (XXX: use buffers) */
757 rt->last_reply[0] = '\0';
761 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
763 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
770 /* XXX: only parse it if first char on line ? */
771 if (return_on_interleaved_data) {
774 ff_rtsp_skip_packet(s);
775 } else if (ch != '\r') {
776 if ((q - buf) < sizeof(buf) - 1)
782 dprintf(s, "line='%s'\n", buf);
784 /* test if last line */
788 if (line_count == 0) {
790 get_word(buf1, sizeof(buf1), &p);
791 get_word(buf1, sizeof(buf1), &p);
792 reply->status_code = atoi(buf1);
793 av_strlcpy(reply->reason, p, sizeof(reply->reason));
795 ff_rtsp_parse_line(reply, p, &rt->auth_state);
796 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
797 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
802 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
803 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
805 content_length = reply->content_length;
806 if (content_length > 0) {
807 /* leave some room for a trailing '\0' (useful for simple parsing) */
808 content = av_malloc(content_length + 1);
809 (void)url_read_complete(rt->rtsp_hd, content, content_length);
810 content[content_length] = '\0';
813 *content_ptr = content;
817 if (rt->seq != reply->seq) {
818 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
819 rt->seq, reply->seq);
823 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
824 reply->notice == 2104 /* Start-of-Stream Reached */ ||
825 reply->notice == 2306 /* Continuous Feed Terminated */) {
826 rt->state = RTSP_STATE_IDLE;
827 } else if (reply->notice >= 4400 && reply->notice < 5500) {
828 return AVERROR(EIO); /* data or server error */
829 } else if (reply->notice == 2401 /* Ticket Expired */ ||
830 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
831 return AVERROR(EPERM);
836 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
837 const char *method, const char *url,
839 const unsigned char *send_content,
840 int send_content_length)
842 RTSPState *rt = s->priv_data;
843 char buf[4096], *out_buf;
844 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
846 /* Add in RTSP headers */
849 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
851 av_strlcat(buf, headers, sizeof(buf));
852 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
853 if (rt->session_id[0] != '\0' && (!headers ||
854 !strstr(headers, "\nIf-Match:"))) {
855 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
858 char *str = ff_http_auth_create_response(&rt->auth_state,
859 rt->auth, url, method);
861 av_strlcat(buf, str, sizeof(buf));
864 if (send_content_length > 0 && send_content)
865 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
866 av_strlcat(buf, "\r\n", sizeof(buf));
868 /* base64 encode rtsp if tunneling */
869 if (rt->control_transport == RTSP_MODE_TUNNEL) {
870 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
874 dprintf(s, "Sending:\n%s--\n", buf);
876 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
877 if (send_content_length > 0 && send_content) {
878 if (rt->control_transport == RTSP_MODE_TUNNEL) {
879 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
880 "with content data not supported\n");
881 return AVERROR_PATCHWELCOME;
883 url_write(rt->rtsp_hd_out, send_content, send_content_length);
885 rt->last_cmd_time = av_gettime();
890 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
891 const char *url, const char *headers)
893 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
896 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
897 const char *headers, RTSPMessageHeader *reply,
898 unsigned char **content_ptr)
900 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
901 content_ptr, NULL, 0);
904 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
905 const char *method, const char *url,
907 RTSPMessageHeader *reply,
908 unsigned char **content_ptr,
909 const unsigned char *send_content,
910 int send_content_length)
912 RTSPState *rt = s->priv_data;
913 HTTPAuthType cur_auth_type;
917 cur_auth_type = rt->auth_state.auth_type;
918 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
920 send_content_length)))
923 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
926 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
927 rt->auth_state.auth_type != HTTP_AUTH_NONE)
930 if (reply->status_code > 400){
931 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
935 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
942 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
944 static int make_setup_request(AVFormatContext *s, const char *host, int port,
945 int lower_transport, const char *real_challenge)
947 RTSPState *rt = s->priv_data;
948 int rtx, j, i, err, interleave = 0;
950 RTSPMessageHeader reply1, *reply = &reply1;
952 const char *trans_pref;
954 if (rt->transport == RTSP_TRANSPORT_RDT)
955 trans_pref = "x-pn-tng";
957 trans_pref = "RTP/AVP";
959 /* default timeout: 1 minute */
962 /* for each stream, make the setup request */
963 /* XXX: we assume the same server is used for the control of each
966 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
967 char transport[2048];
970 * WMS serves all UDP data over a single connection, the RTX, which
971 * isn't necessarily the first in the SDP but has to be the first
972 * to be set up, else the second/third SETUP will fail with a 461.
974 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
975 rt->server_type == RTSP_SERVER_WMS) {
978 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
979 int len = strlen(rt->rtsp_streams[rtx]->control_url);
981 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
985 if (rtx == rt->nb_rtsp_streams)
986 return -1; /* no RTX found */
987 rtsp_st = rt->rtsp_streams[rtx];
989 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
991 rtsp_st = rt->rtsp_streams[i];
994 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
997 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
998 port = reply->transports[0].client_port_min;
1002 /* first try in specified port range */
1003 if (RTSP_RTP_PORT_MIN != 0) {
1004 while (j <= RTSP_RTP_PORT_MAX) {
1005 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1006 "?localport=%d", j);
1007 /* we will use two ports per rtp stream (rtp and rtcp) */
1009 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1015 /* then try on any port */
1016 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1017 err = AVERROR_INVALIDDATA;
1023 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1025 snprintf(transport, sizeof(transport) - 1,
1026 "%s/UDP;", trans_pref);
1027 if (rt->server_type != RTSP_SERVER_REAL)
1028 av_strlcat(transport, "unicast;", sizeof(transport));
1029 av_strlcatf(transport, sizeof(transport),
1030 "client_port=%d", port);
1031 if (rt->transport == RTSP_TRANSPORT_RTP &&
1032 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1033 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1037 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1038 /** For WMS streams, the application streams are only used for
1039 * UDP. When trying to set it up for TCP streams, the server
1040 * will return an error. Therefore, we skip those streams. */
1041 if (rt->server_type == RTSP_SERVER_WMS &&
1042 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1045 snprintf(transport, sizeof(transport) - 1,
1046 "%s/TCP;", trans_pref);
1047 if (rt->server_type == RTSP_SERVER_WMS)
1048 av_strlcat(transport, "unicast;", sizeof(transport));
1049 av_strlcatf(transport, sizeof(transport),
1050 "interleaved=%d-%d",
1051 interleave, interleave + 1);
1055 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1056 snprintf(transport, sizeof(transport) - 1,
1057 "%s/UDP;multicast", trans_pref);
1060 av_strlcat(transport, ";mode=receive", sizeof(transport));
1061 } else if (rt->server_type == RTSP_SERVER_REAL ||
1062 rt->server_type == RTSP_SERVER_WMS)
1063 av_strlcat(transport, ";mode=play", sizeof(transport));
1064 snprintf(cmd, sizeof(cmd),
1065 "Transport: %s\r\n",
1067 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1068 char real_res[41], real_csum[9];
1069 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1071 av_strlcatf(cmd, sizeof(cmd),
1073 "RealChallenge2: %s, sd=%s\r\n",
1074 rt->session_id, real_res, real_csum);
1076 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1077 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1080 } else if (reply->status_code != RTSP_STATUS_OK ||
1081 reply->nb_transports != 1) {
1082 err = AVERROR_INVALIDDATA;
1086 /* XXX: same protocol for all streams is required */
1088 if (reply->transports[0].lower_transport != rt->lower_transport ||
1089 reply->transports[0].transport != rt->transport) {
1090 err = AVERROR_INVALIDDATA;
1094 rt->lower_transport = reply->transports[0].lower_transport;
1095 rt->transport = reply->transports[0].transport;
1098 /* close RTP connection if not chosen */
1099 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1100 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1101 url_close(rtsp_st->rtp_handle);
1102 rtsp_st->rtp_handle = NULL;
1105 switch(reply->transports[0].lower_transport) {
1106 case RTSP_LOWER_TRANSPORT_TCP:
1107 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1108 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1111 case RTSP_LOWER_TRANSPORT_UDP: {
1114 /* Use source address if specified */
1115 if (reply->transports[0].source[0]) {
1116 ff_url_join(url, sizeof(url), "rtp", NULL,
1117 reply->transports[0].source,
1118 reply->transports[0].server_port_min, NULL);
1120 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1121 reply->transports[0].server_port_min, NULL);
1123 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1124 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1125 err = AVERROR_INVALIDDATA;
1128 /* Try to initialize the connection state in a
1129 * potential NAT router by sending dummy packets.
1130 * RTP/RTCP dummy packets are used for RDT, too.
1132 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1134 rtp_send_punch_packets(rtsp_st->rtp_handle);
1137 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1138 char url[1024], namebuf[50];
1139 struct sockaddr_storage addr;
1142 if (reply->transports[0].destination.ss_family) {
1143 addr = reply->transports[0].destination;
1144 port = reply->transports[0].port_min;
1145 ttl = reply->transports[0].ttl;
1147 addr = rtsp_st->sdp_ip;
1148 port = rtsp_st->sdp_port;
1149 ttl = rtsp_st->sdp_ttl;
1151 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1152 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1153 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1154 port, "?ttl=%d", ttl);
1155 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1156 err = AVERROR_INVALIDDATA;
1163 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1167 if (reply->timeout > 0)
1168 rt->timeout = reply->timeout;
1170 if (rt->server_type == RTSP_SERVER_REAL)
1171 rt->need_subscription = 1;
1176 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1177 if (rt->rtsp_streams[i]->rtp_handle) {
1178 url_close(rt->rtsp_streams[i]->rtp_handle);
1179 rt->rtsp_streams[i]->rtp_handle = NULL;
1185 void ff_rtsp_close_connections(AVFormatContext *s)
1187 RTSPState *rt = s->priv_data;
1188 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1189 url_close(rt->rtsp_hd);
1190 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1193 int ff_rtsp_connect(AVFormatContext *s)
1195 RTSPState *rt = s->priv_data;
1196 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1197 char *option_list, *option, *filename;
1198 int port, err, tcp_fd;
1199 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1200 int lower_transport_mask = 0;
1201 char real_challenge[64];
1202 struct sockaddr_storage peer;
1203 socklen_t peer_len = sizeof(peer);
1205 if (!ff_network_init())
1206 return AVERROR(EIO);
1208 rt->control_transport = RTSP_MODE_PLAIN;
1209 /* extract hostname and port */
1210 av_url_split(NULL, 0, auth, sizeof(auth),
1211 host, sizeof(host), &port, path, sizeof(path), s->filename);
1213 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1216 port = RTSP_DEFAULT_PORT;
1218 /* search for options */
1219 option_list = strrchr(path, '?');
1221 /* Strip out the RTSP specific options, write out the rest of
1222 * the options back into the same string. */
1223 filename = option_list;
1224 while (option_list) {
1225 /* move the option pointer */
1226 option = ++option_list;
1227 option_list = strchr(option_list, '&');
1231 /* handle the options */
1232 if (!strcmp(option, "udp")) {
1233 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1234 } else if (!strcmp(option, "multicast")) {
1235 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1236 } else if (!strcmp(option, "tcp")) {
1237 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1238 } else if(!strcmp(option, "http")) {
1239 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1240 rt->control_transport = RTSP_MODE_TUNNEL;
1242 /* Write options back into the buffer, using memmove instead
1243 * of strcpy since the strings may overlap. */
1244 int len = strlen(option);
1245 memmove(++filename, option, len);
1247 if (option_list) *filename = '&';
1253 if (!lower_transport_mask)
1254 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1257 /* Only UDP or TCP - UDP multicast isn't supported. */
1258 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1259 (1 << RTSP_LOWER_TRANSPORT_TCP);
1260 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1261 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1262 "only UDP and TCP are supported for output.\n");
1263 err = AVERROR(EINVAL);
1268 /* Construct the URI used in request; this is similar to s->filename,
1269 * but with authentication credentials removed and RTSP specific options
1271 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1272 host, port, "%s", path);
1274 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1275 /* set up initial handshake for tunneling */
1276 char httpname[1024];
1277 char sessioncookie[17];
1280 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1281 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1282 av_get_random_seed(), av_get_random_seed());
1285 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1290 /* generate GET headers */
1291 snprintf(headers, sizeof(headers),
1292 "x-sessioncookie: %s\r\n"
1293 "Accept: application/x-rtsp-tunnelled\r\n"
1294 "Pragma: no-cache\r\n"
1295 "Cache-Control: no-cache\r\n",
1297 ff_http_set_headers(rt->rtsp_hd, headers);
1299 /* complete the connection */
1300 if (url_connect(rt->rtsp_hd)) {
1306 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1311 /* generate POST headers */
1312 snprintf(headers, sizeof(headers),
1313 "x-sessioncookie: %s\r\n"
1314 "Content-Type: application/x-rtsp-tunnelled\r\n"
1315 "Pragma: no-cache\r\n"
1316 "Cache-Control: no-cache\r\n"
1317 "Content-Length: 32767\r\n"
1318 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1320 ff_http_set_headers(rt->rtsp_hd_out, headers);
1321 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1323 /* Initialize the authentication state for the POST session. The HTTP
1324 * protocol implementation doesn't properly handle multi-pass
1325 * authentication for POST requests, since it would require one of
1327 * - implementing Expect: 100-continue, which many HTTP servers
1328 * don't support anyway, even less the RTSP servers that do HTTP
1330 * - sending the whole POST data until getting a 401 reply specifying
1331 * what authentication method to use, then resending all that data
1332 * - waiting for potential 401 replies directly after sending the
1333 * POST header (waiting for some unspecified time)
1334 * Therefore, we copy the full auth state, which works for both basic
1335 * and digest. (For digest, we would have to synchronize the nonce
1336 * count variable between the two sessions, if we'd do more requests
1337 * with the original session, though.)
1339 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1341 /* complete the connection */
1342 if (url_connect(rt->rtsp_hd_out)) {
1347 /* open the tcp connection */
1348 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1349 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1353 rt->rtsp_hd_out = rt->rtsp_hd;
1357 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1358 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1359 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1360 NULL, 0, NI_NUMERICHOST);
1363 /* request options supported by the server; this also detects server
1365 for (rt->server_type = RTSP_SERVER_RTP;;) {
1367 if (rt->server_type == RTSP_SERVER_REAL)
1370 * The following entries are required for proper
1371 * streaming from a Realmedia server. They are
1372 * interdependent in some way although we currently
1373 * don't quite understand how. Values were copied
1374 * from mplayer SVN r23589.
1375 * @param CompanyID is a 16-byte ID in base64
1376 * @param ClientChallenge is a 16-byte ID in hex
1378 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1379 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1380 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1381 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1383 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1384 if (reply->status_code != RTSP_STATUS_OK) {
1385 err = AVERROR_INVALIDDATA;
1389 /* detect server type if not standard-compliant RTP */
1390 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1391 rt->server_type = RTSP_SERVER_REAL;
1393 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1394 rt->server_type = RTSP_SERVER_WMS;
1395 } else if (rt->server_type == RTSP_SERVER_REAL)
1396 strcpy(real_challenge, reply->real_challenge);
1400 if (s->iformat && CONFIG_RTSP_DEMUXER)
1401 err = ff_rtsp_setup_input_streams(s, reply);
1402 else if (CONFIG_RTSP_MUXER)
1403 err = ff_rtsp_setup_output_streams(s, host);
1408 int lower_transport = ff_log2_tab[lower_transport_mask &
1409 ~(lower_transport_mask - 1)];
1411 err = make_setup_request(s, host, port, lower_transport,
1412 rt->server_type == RTSP_SERVER_REAL ?
1413 real_challenge : NULL);
1416 lower_transport_mask &= ~(1 << lower_transport);
1417 if (lower_transport_mask == 0 && err == 1) {
1418 err = FF_NETERROR(EPROTONOSUPPORT);
1423 rt->state = RTSP_STATE_IDLE;
1424 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1427 ff_rtsp_close_streams(s);
1428 ff_rtsp_close_connections(s);
1429 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1430 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1431 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1439 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1442 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1443 uint8_t *buf, int buf_size, int64_t wait_end)
1445 RTSPState *rt = s->priv_data;
1446 RTSPStream *rtsp_st;
1448 int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1452 if (url_interrupt_cb())
1453 return AVERROR(EINTR);
1454 if (wait_end && wait_end - av_gettime() < 0)
1455 return AVERROR(EAGAIN);
1458 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1459 FD_SET(tcp_fd, &rfds);
1464 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1465 rtsp_st = rt->rtsp_streams[i];
1466 if (rtsp_st->rtp_handle) {
1467 fd = url_get_file_handle(rtsp_st->rtp_handle);
1468 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1469 if (FFMAX(fd, fd_rtcp) > fd_max)
1470 fd_max = FFMAX(fd, fd_rtcp);
1472 FD_SET(fd_rtcp, &rfds);
1476 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1477 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1480 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1481 rtsp_st = rt->rtsp_streams[i];
1482 if (rtsp_st->rtp_handle) {
1483 fd = url_get_file_handle(rtsp_st->rtp_handle);
1484 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1485 if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
1486 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1488 *prtsp_st = rtsp_st;
1494 #if CONFIG_RTSP_DEMUXER
1495 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1496 RTSPMessageHeader reply;
1498 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1501 /* XXX: parse message */
1502 if (rt->state != RTSP_STATE_STREAMING)
1506 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1507 return FF_NETERROR(ETIMEDOUT);
1508 } else if (n < 0 && errno != EINTR)
1509 return AVERROR(errno);
1513 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1515 RTSPState *rt = s->priv_data;
1517 RTSPStream *rtsp_st, *first_queue_st = NULL;
1518 int64_t wait_end = 0;
1520 if (rt->nb_byes == rt->nb_rtsp_streams)
1523 /* get next frames from the same RTP packet */
1524 if (rt->cur_transport_priv) {
1525 if (rt->transport == RTSP_TRANSPORT_RDT) {
1526 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1528 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1530 rt->cur_transport_priv = NULL;
1532 } else if (ret == 1) {
1535 rt->cur_transport_priv = NULL;
1538 if (rt->transport == RTSP_TRANSPORT_RTP) {
1540 int64_t first_queue_time = 0;
1541 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1542 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1543 int64_t queue_time = ff_rtp_queued_packet_time(rtpctx);
1544 if (queue_time && (queue_time - first_queue_time < 0 ||
1545 !first_queue_time)) {
1546 first_queue_time = queue_time;
1547 first_queue_st = rt->rtsp_streams[i];
1550 if (first_queue_time)
1551 wait_end = first_queue_time + s->max_delay;
1554 /* read next RTP packet */
1557 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1559 return AVERROR(ENOMEM);
1562 switch(rt->lower_transport) {
1564 #if CONFIG_RTSP_DEMUXER
1565 case RTSP_LOWER_TRANSPORT_TCP:
1566 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1569 case RTSP_LOWER_TRANSPORT_UDP:
1570 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1571 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1572 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1573 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1576 if (len == AVERROR(EAGAIN) && first_queue_st &&
1577 rt->transport == RTSP_TRANSPORT_RTP) {
1578 rtsp_st = first_queue_st;
1579 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1586 if (rt->transport == RTSP_TRANSPORT_RDT) {
1587 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1589 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1591 /* Either bad packet, or a RTCP packet. Check if the
1592 * first_rtcp_ntp_time field was initialized. */
1593 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1594 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1595 /* first_rtcp_ntp_time has been initialized for this stream,
1596 * copy the same value to all other uninitialized streams,
1597 * in order to map their timestamp origin to the same ntp time
1600 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1601 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1603 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1604 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1607 if (ret == -RTCP_BYE) {
1610 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1611 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1613 if (rt->nb_byes == rt->nb_rtsp_streams)
1622 /* more packets may follow, so we save the RTP context */
1623 rt->cur_transport_priv = rtsp_st->transport_priv;
1627 #endif /* CONFIG_RTPDEC */
1629 #if CONFIG_SDP_DEMUXER
1630 static int sdp_probe(AVProbeData *p1)
1632 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1634 /* we look for a line beginning "c=IN IP" */
1635 while (p < p_end && *p != '\0') {
1636 if (p + sizeof("c=IN IP") - 1 < p_end &&
1637 av_strstart(p, "c=IN IP", NULL))
1638 return AVPROBE_SCORE_MAX / 2;
1640 while (p < p_end - 1 && *p != '\n') p++;
1649 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1651 RTSPState *rt = s->priv_data;
1652 RTSPStream *rtsp_st;
1657 if (!ff_network_init())
1658 return AVERROR(EIO);
1660 /* read the whole sdp file */
1661 /* XXX: better loading */
1662 content = av_malloc(SDP_MAX_SIZE);
1663 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1666 return AVERROR_INVALIDDATA;
1668 content[size] ='\0';
1670 ff_sdp_parse(s, content);
1673 /* open each RTP stream */
1674 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1676 rtsp_st = rt->rtsp_streams[i];
1678 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1679 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1680 ff_url_join(url, sizeof(url), "rtp", NULL,
1681 namebuf, rtsp_st->sdp_port,
1682 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1684 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1685 err = AVERROR_INVALIDDATA;
1688 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1693 ff_rtsp_close_streams(s);
1698 static int sdp_read_close(AVFormatContext *s)
1700 ff_rtsp_close_streams(s);
1705 AVInputFormat sdp_demuxer = {
1707 NULL_IF_CONFIG_SMALL("SDP"),
1711 ff_rtsp_fetch_packet,
1714 #endif /* CONFIG_SDP_DEMUXER */
1716 #if CONFIG_RTP_DEMUXER
1717 static int rtp_probe(AVProbeData *p)
1719 if (av_strstart(p->filename, "rtp:", NULL))
1720 return AVPROBE_SCORE_MAX;
1724 static int rtp_read_header(AVFormatContext *s,
1725 AVFormatParameters *ap)
1727 uint8_t recvbuf[1500];
1728 char host[500], sdp[500];
1730 URLContext* in = NULL;
1732 AVCodecContext codec;
1733 struct sockaddr_storage addr;
1735 socklen_t addrlen = sizeof(addr);
1737 if (!ff_network_init())
1738 return AVERROR(EIO);
1740 ret = url_open(&in, s->filename, URL_RDONLY);
1745 ret = url_read(in, recvbuf, sizeof(recvbuf));
1746 if (ret == AVERROR(EAGAIN))
1751 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1755 if ((recvbuf[0] & 0xc0) != 0x80) {
1756 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1761 payload_type = recvbuf[1] & 0x7f;
1764 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1768 memset(&codec, 0, sizeof(codec));
1769 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1770 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1771 "without an SDP file describing it\n",
1775 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1776 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1777 "properly you need an SDP file "
1781 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1782 NULL, 0, s->filename);
1784 snprintf(sdp, sizeof(sdp),
1785 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1786 addr.ss_family == AF_INET ? 4 : 6, host,
1787 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1788 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1789 port, payload_type);
1790 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1792 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1795 /* sdp_read_header initializes this again */
1798 ret = sdp_read_header(s, ap);
1809 AVInputFormat rtp_demuxer = {
1811 NULL_IF_CONFIG_SMALL("RTP input format"),
1815 ff_rtsp_fetch_packet,
1817 .flags = AVFMT_NOFILE,
1819 #endif /* CONFIG_RTP_DEMUXER */