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 AVStream *st, RTSPStream *rtsp_st,
139 int payload_type, const char *p)
141 AVCodecContext *codec = st->codec;
147 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
148 * see if we can handle this kind of payload.
149 * The space should normally not be there but some Real streams or
150 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
151 * have a trailing space. */
152 get_word_sep(buf, sizeof(buf), "/ ", &p);
153 if (payload_type >= RTP_PT_PRIVATE) {
154 RTPDynamicProtocolHandler *handler =
155 ff_rtp_handler_find_by_name(buf, codec->codec_type);
156 init_rtp_handler(handler, rtsp_st, codec);
157 /* If no dynamic handler was found, check with the list of standard
158 * allocated types, if such a stream for some reason happens to
159 * use a private payload type. This isn't handled in rtpdec.c, since
160 * the format name from the rtpmap line never is passed into rtpdec. */
161 if (!rtsp_st->dynamic_handler)
162 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
164 /* We are in a standard case
165 * (from http://www.iana.org/assignments/rtp-parameters). */
166 /* search into AVRtpPayloadTypes[] */
167 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
170 c = avcodec_find_decoder(codec->codec_id);
176 get_word_sep(buf, sizeof(buf), "/", &p);
178 switch (codec->codec_type) {
179 case AVMEDIA_TYPE_AUDIO:
180 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
181 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
182 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
184 codec->sample_rate = i;
185 av_set_pts_info(st, 32, 1, codec->sample_rate);
186 get_word_sep(buf, sizeof(buf), "/", &p);
190 // TODO: there is a bug here; if it is a mono stream, and
191 // less than 22000Hz, faad upconverts to stereo and twice
192 // the frequency. No problem, but the sample rate is being
193 // set here by the sdp line. Patch on its way. (rdm)
195 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
197 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
200 case AVMEDIA_TYPE_VIDEO:
201 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
203 av_set_pts_info(st, 32, 1, i);
211 /* parse the attribute line from the fmtp a line of an sdp response. This
212 * is broken out as a function because it is used in rtp_h264.c, which is
214 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
215 char *value, int value_size)
217 *p += strspn(*p, SPACE_CHARS);
219 get_word_sep(attr, attr_size, "=", p);
222 get_word_sep(value, value_size, ";", p);
230 typedef struct SDPParseState {
232 struct sockaddr_storage default_ip;
234 int skip_media; ///< set if an unknown m= line occurs
237 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
238 int letter, const char *buf)
240 RTSPState *rt = s->priv_data;
241 char buf1[64], st_type[64];
243 enum AVMediaType codec_type;
247 struct sockaddr_storage sdp_ip;
250 dprintf(s, "sdp: %c='%s'\n", letter, buf);
253 if (s1->skip_media && letter != 'm')
257 get_word(buf1, sizeof(buf1), &p);
258 if (strcmp(buf1, "IN") != 0)
260 get_word(buf1, sizeof(buf1), &p);
261 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
263 get_word_sep(buf1, sizeof(buf1), "/", &p);
264 if (get_sockaddr(buf1, &sdp_ip))
269 get_word_sep(buf1, sizeof(buf1), "/", &p);
272 if (s->nb_streams == 0) {
273 s1->default_ip = sdp_ip;
274 s1->default_ttl = ttl;
276 st = s->streams[s->nb_streams - 1];
277 rtsp_st = st->priv_data;
278 rtsp_st->sdp_ip = sdp_ip;
279 rtsp_st->sdp_ttl = ttl;
283 av_metadata_set2(&s->metadata, "title", p, 0);
286 if (s->nb_streams == 0) {
287 av_metadata_set2(&s->metadata, "comment", p, 0);
294 get_word(st_type, sizeof(st_type), &p);
295 if (!strcmp(st_type, "audio")) {
296 codec_type = AVMEDIA_TYPE_AUDIO;
297 } else if (!strcmp(st_type, "video")) {
298 codec_type = AVMEDIA_TYPE_VIDEO;
299 } else if (!strcmp(st_type, "application")) {
300 codec_type = AVMEDIA_TYPE_DATA;
305 rtsp_st = av_mallocz(sizeof(RTSPStream));
308 rtsp_st->stream_index = -1;
309 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
311 rtsp_st->sdp_ip = s1->default_ip;
312 rtsp_st->sdp_ttl = s1->default_ttl;
314 get_word(buf1, sizeof(buf1), &p); /* port */
315 rtsp_st->sdp_port = atoi(buf1);
317 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
319 /* XXX: handle list of formats */
320 get_word(buf1, sizeof(buf1), &p); /* format list */
321 rtsp_st->sdp_payload_type = atoi(buf1);
323 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
324 /* no corresponding stream */
326 st = av_new_stream(s, 0);
329 st->priv_data = rtsp_st;
330 rtsp_st->stream_index = st->index;
331 st->codec->codec_type = codec_type;
332 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
333 RTPDynamicProtocolHandler *handler;
334 /* if standard payload type, we can find the codec right now */
335 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
336 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
337 st->codec->sample_rate > 0)
338 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
339 /* Even static payload types may need a custom depacketizer */
340 handler = ff_rtp_handler_find_by_id(
341 rtsp_st->sdp_payload_type, st->codec->codec_type);
342 init_rtp_handler(handler, rtsp_st, st->codec);
345 /* put a default control url */
346 av_strlcpy(rtsp_st->control_url, rt->control_uri,
347 sizeof(rtsp_st->control_url));
350 if (av_strstart(p, "control:", &p)) {
351 if (s->nb_streams == 0) {
352 if (!strncmp(p, "rtsp://", 7))
353 av_strlcpy(rt->control_uri, p,
354 sizeof(rt->control_uri));
357 /* get the control url */
358 st = s->streams[s->nb_streams - 1];
359 rtsp_st = st->priv_data;
361 /* XXX: may need to add full url resolution */
362 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
364 if (proto[0] == '\0') {
365 /* relative control URL */
366 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
367 av_strlcat(rtsp_st->control_url, "/",
368 sizeof(rtsp_st->control_url));
369 av_strlcat(rtsp_st->control_url, p,
370 sizeof(rtsp_st->control_url));
372 av_strlcpy(rtsp_st->control_url, p,
373 sizeof(rtsp_st->control_url));
375 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
376 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
377 get_word(buf1, sizeof(buf1), &p);
378 payload_type = atoi(buf1);
379 st = s->streams[s->nb_streams - 1];
380 rtsp_st = st->priv_data;
381 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
382 } else if (av_strstart(p, "fmtp:", &p) ||
383 av_strstart(p, "framesize:", &p)) {
384 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
385 // let dynamic protocol handlers have a stab at the line.
386 get_word(buf1, sizeof(buf1), &p);
387 payload_type = atoi(buf1);
388 for (i = 0; i < s->nb_streams; i++) {
390 rtsp_st = st->priv_data;
391 if (rtsp_st->sdp_payload_type == payload_type &&
392 rtsp_st->dynamic_handler &&
393 rtsp_st->dynamic_handler->parse_sdp_a_line)
394 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
395 rtsp_st->dynamic_protocol_context, buf);
397 } else if (av_strstart(p, "range:", &p)) {
400 // this is so that seeking on a streamed file can work.
401 rtsp_parse_range_npt(p, &start, &end);
402 s->start_time = start;
403 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
404 s->duration = (end == AV_NOPTS_VALUE) ?
405 AV_NOPTS_VALUE : end - start;
406 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
408 rt->transport = RTSP_TRANSPORT_RDT;
409 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
411 st = s->streams[s->nb_streams - 1];
412 st->codec->sample_rate = atoi(p);
414 if (rt->server_type == RTSP_SERVER_WMS)
415 ff_wms_parse_sdp_a_line(s, p);
416 if (s->nb_streams > 0) {
417 if (rt->server_type == RTSP_SERVER_REAL)
418 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
420 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
421 if (rtsp_st->dynamic_handler &&
422 rtsp_st->dynamic_handler->parse_sdp_a_line)
423 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
425 rtsp_st->dynamic_protocol_context, buf);
432 int ff_sdp_parse(AVFormatContext *s, const char *content)
436 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
437 * contain long SDP lines containing complete ASF Headers (several
438 * kB) or arrays of MDPR (RM stream descriptor) headers plus
439 * "rulebooks" describing their properties. Therefore, the SDP line
442 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
443 * in rtpdec_xiph.c. */
445 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
447 memset(s1, 0, sizeof(SDPParseState));
450 p += strspn(p, SPACE_CHARS);
458 /* get the content */
460 while (*p != '\n' && *p != '\r' && *p != '\0') {
461 if ((q - buf) < sizeof(buf) - 1)
466 sdp_parse_line(s, s1, letter, buf);
468 while (*p != '\n' && *p != '\0')
475 #endif /* CONFIG_RTPDEC */
477 /* close and free RTSP streams */
478 void ff_rtsp_close_streams(AVFormatContext *s)
480 RTSPState *rt = s->priv_data;
484 for (i = 0; i < rt->nb_rtsp_streams; i++) {
485 rtsp_st = rt->rtsp_streams[i];
487 if (rtsp_st->transport_priv) {
489 AVFormatContext *rtpctx = rtsp_st->transport_priv;
490 av_write_trailer(rtpctx);
491 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
493 url_close_dyn_buf(rtpctx->pb, &ptr);
496 url_fclose(rtpctx->pb);
498 av_metadata_free(&rtpctx->streams[0]->metadata);
499 av_metadata_free(&rtpctx->metadata);
500 av_free(rtpctx->streams[0]);
502 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
503 ff_rdt_parse_close(rtsp_st->transport_priv);
504 else if (CONFIG_RTPDEC)
505 rtp_parse_close(rtsp_st->transport_priv);
507 if (rtsp_st->rtp_handle)
508 url_close(rtsp_st->rtp_handle);
509 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
510 rtsp_st->dynamic_handler->close(
511 rtsp_st->dynamic_protocol_context);
514 av_free(rt->rtsp_streams);
516 av_close_input_stream (rt->asf_ctx);
519 av_free(rt->recvbuf);
522 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
524 RTSPState *rt = s->priv_data;
527 /* open the RTP context */
528 if (rtsp_st->stream_index >= 0)
529 st = s->streams[rtsp_st->stream_index];
531 s->ctx_flags |= AVFMTCTX_NOHEADER;
533 if (s->oformat && CONFIG_RTSP_MUXER) {
534 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
536 RTSP_TCP_MAX_PACKET_SIZE);
537 /* Ownership of rtp_handle is passed to the rtp mux context */
538 rtsp_st->rtp_handle = NULL;
539 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
540 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
541 rtsp_st->dynamic_protocol_context,
542 rtsp_st->dynamic_handler);
543 else if (CONFIG_RTPDEC)
544 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
545 rtsp_st->sdp_payload_type,
546 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
547 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
549 if (!rtsp_st->transport_priv) {
550 return AVERROR(ENOMEM);
551 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
552 if (rtsp_st->dynamic_handler) {
553 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
554 rtsp_st->dynamic_protocol_context,
555 rtsp_st->dynamic_handler);
562 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
563 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
569 p += strspn(p, SPACE_CHARS);
570 v = strtol(p, (char **)&p, 10);
574 v = strtol(p, (char **)&p, 10);
583 /* XXX: only one transport specification is parsed */
584 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
586 char transport_protocol[16];
588 char lower_transport[16];
590 RTSPTransportField *th;
593 reply->nb_transports = 0;
596 p += strspn(p, SPACE_CHARS);
600 th = &reply->transports[reply->nb_transports];
602 get_word_sep(transport_protocol, sizeof(transport_protocol),
604 if (!strcasecmp (transport_protocol, "rtp")) {
605 get_word_sep(profile, sizeof(profile), "/;,", &p);
606 lower_transport[0] = '\0';
607 /* rtp/avp/<protocol> */
609 get_word_sep(lower_transport, sizeof(lower_transport),
612 th->transport = RTSP_TRANSPORT_RTP;
613 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
614 !strcasecmp (transport_protocol, "x-real-rdt")) {
615 /* x-pn-tng/<protocol> */
616 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
618 th->transport = RTSP_TRANSPORT_RDT;
620 if (!strcasecmp(lower_transport, "TCP"))
621 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
623 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
627 /* get each parameter */
628 while (*p != '\0' && *p != ',') {
629 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
630 if (!strcmp(parameter, "port")) {
633 rtsp_parse_range(&th->port_min, &th->port_max, &p);
635 } else if (!strcmp(parameter, "client_port")) {
638 rtsp_parse_range(&th->client_port_min,
639 &th->client_port_max, &p);
641 } else if (!strcmp(parameter, "server_port")) {
644 rtsp_parse_range(&th->server_port_min,
645 &th->server_port_max, &p);
647 } else if (!strcmp(parameter, "interleaved")) {
650 rtsp_parse_range(&th->interleaved_min,
651 &th->interleaved_max, &p);
653 } else if (!strcmp(parameter, "multicast")) {
654 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
655 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
656 } else if (!strcmp(parameter, "ttl")) {
659 th->ttl = strtol(p, (char **)&p, 10);
661 } else if (!strcmp(parameter, "destination")) {
664 get_word_sep(buf, sizeof(buf), ";,", &p);
665 get_sockaddr(buf, &th->destination);
667 } else if (!strcmp(parameter, "source")) {
670 get_word_sep(buf, sizeof(buf), ";,", &p);
671 av_strlcpy(th->source, buf, sizeof(th->source));
675 while (*p != ';' && *p != '\0' && *p != ',')
683 reply->nb_transports++;
687 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
688 HTTPAuthState *auth_state)
692 /* NOTE: we do case independent match for broken servers */
694 if (av_stristart(p, "Session:", &p)) {
696 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
697 if (av_stristart(p, ";timeout=", &p) &&
698 (t = strtol(p, NULL, 10)) > 0) {
701 } else if (av_stristart(p, "Content-Length:", &p)) {
702 reply->content_length = strtol(p, NULL, 10);
703 } else if (av_stristart(p, "Transport:", &p)) {
704 rtsp_parse_transport(reply, p);
705 } else if (av_stristart(p, "CSeq:", &p)) {
706 reply->seq = strtol(p, NULL, 10);
707 } else if (av_stristart(p, "Range:", &p)) {
708 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
709 } else if (av_stristart(p, "RealChallenge1:", &p)) {
710 p += strspn(p, SPACE_CHARS);
711 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
712 } else if (av_stristart(p, "Server:", &p)) {
713 p += strspn(p, SPACE_CHARS);
714 av_strlcpy(reply->server, p, sizeof(reply->server));
715 } else if (av_stristart(p, "Notice:", &p) ||
716 av_stristart(p, "X-Notice:", &p)) {
717 reply->notice = strtol(p, NULL, 10);
718 } else if (av_stristart(p, "Location:", &p)) {
719 p += strspn(p, SPACE_CHARS);
720 av_strlcpy(reply->location, p , sizeof(reply->location));
721 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
722 p += strspn(p, SPACE_CHARS);
723 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
724 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
725 p += strspn(p, SPACE_CHARS);
726 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
727 } else if (av_stristart(p, "Content-Base:", &p)) {
728 p += strspn(p, SPACE_CHARS);
729 av_strlcpy(reply->content_base, p , sizeof(reply->content_base));
733 /* skip a RTP/TCP interleaved packet */
734 void ff_rtsp_skip_packet(AVFormatContext *s)
736 RTSPState *rt = s->priv_data;
740 ret = url_read_complete(rt->rtsp_hd, buf, 3);
743 len = AV_RB16(buf + 1);
745 dprintf(s, "skipping RTP packet len=%d\n", len);
750 if (len1 > sizeof(buf))
752 ret = url_read_complete(rt->rtsp_hd, buf, len1);
759 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
760 unsigned char **content_ptr,
761 int return_on_interleaved_data)
763 RTSPState *rt = s->priv_data;
764 char buf[4096], buf1[1024], *q;
767 int ret, content_length, line_count = 0;
768 unsigned char *content = NULL;
770 memset(reply, 0, sizeof(*reply));
772 /* parse reply (XXX: use buffers) */
773 rt->last_reply[0] = '\0';
777 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
779 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
786 /* XXX: only parse it if first char on line ? */
787 if (return_on_interleaved_data) {
790 ff_rtsp_skip_packet(s);
791 } else if (ch != '\r') {
792 if ((q - buf) < sizeof(buf) - 1)
798 dprintf(s, "line='%s'\n", buf);
800 /* test if last line */
804 if (line_count == 0) {
806 get_word(buf1, sizeof(buf1), &p);
807 get_word(buf1, sizeof(buf1), &p);
808 reply->status_code = atoi(buf1);
809 av_strlcpy(reply->reason, p, sizeof(reply->reason));
811 ff_rtsp_parse_line(reply, p, &rt->auth_state);
812 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
813 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
818 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
819 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
821 content_length = reply->content_length;
822 if (content_length > 0) {
823 /* leave some room for a trailing '\0' (useful for simple parsing) */
824 content = av_malloc(content_length + 1);
825 (void)url_read_complete(rt->rtsp_hd, content, content_length);
826 content[content_length] = '\0';
829 *content_ptr = content;
833 if (rt->seq != reply->seq) {
834 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
835 rt->seq, reply->seq);
839 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
840 reply->notice == 2104 /* Start-of-Stream Reached */ ||
841 reply->notice == 2306 /* Continuous Feed Terminated */) {
842 rt->state = RTSP_STATE_IDLE;
843 } else if (reply->notice >= 4400 && reply->notice < 5500) {
844 return AVERROR(EIO); /* data or server error */
845 } else if (reply->notice == 2401 /* Ticket Expired */ ||
846 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
847 return AVERROR(EPERM);
852 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
853 const char *method, const char *url,
855 const unsigned char *send_content,
856 int send_content_length)
858 RTSPState *rt = s->priv_data;
859 char buf[4096], *out_buf;
860 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
862 /* Add in RTSP headers */
865 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
867 av_strlcat(buf, headers, sizeof(buf));
868 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
869 if (rt->session_id[0] != '\0' && (!headers ||
870 !strstr(headers, "\nIf-Match:"))) {
871 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
874 char *str = ff_http_auth_create_response(&rt->auth_state,
875 rt->auth, url, method);
877 av_strlcat(buf, str, sizeof(buf));
880 if (send_content_length > 0 && send_content)
881 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
882 av_strlcat(buf, "\r\n", sizeof(buf));
884 /* base64 encode rtsp if tunneling */
885 if (rt->control_transport == RTSP_MODE_TUNNEL) {
886 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
890 dprintf(s, "Sending:\n%s--\n", buf);
892 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
893 if (send_content_length > 0 && send_content) {
894 if (rt->control_transport == RTSP_MODE_TUNNEL) {
895 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
896 "with content data not supported\n");
897 return AVERROR_PATCHWELCOME;
899 url_write(rt->rtsp_hd_out, send_content, send_content_length);
901 rt->last_cmd_time = av_gettime();
906 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
907 const char *url, const char *headers)
909 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
912 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
913 const char *headers, RTSPMessageHeader *reply,
914 unsigned char **content_ptr)
916 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
917 content_ptr, NULL, 0);
920 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
921 const char *method, const char *url,
923 RTSPMessageHeader *reply,
924 unsigned char **content_ptr,
925 const unsigned char *send_content,
926 int send_content_length)
928 RTSPState *rt = s->priv_data;
929 HTTPAuthType cur_auth_type;
933 cur_auth_type = rt->auth_state.auth_type;
934 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
936 send_content_length)))
939 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
942 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
943 rt->auth_state.auth_type != HTTP_AUTH_NONE)
946 if (reply->status_code > 400){
947 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
951 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
958 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
960 static int make_setup_request(AVFormatContext *s, const char *host, int port,
961 int lower_transport, const char *real_challenge)
963 RTSPState *rt = s->priv_data;
964 int rtx, j, i, err, interleave = 0;
966 RTSPMessageHeader reply1, *reply = &reply1;
968 const char *trans_pref;
970 if (rt->transport == RTSP_TRANSPORT_RDT)
971 trans_pref = "x-pn-tng";
973 trans_pref = "RTP/AVP";
975 /* default timeout: 1 minute */
978 /* for each stream, make the setup request */
979 /* XXX: we assume the same server is used for the control of each
982 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
983 char transport[2048];
986 * WMS serves all UDP data over a single connection, the RTX, which
987 * isn't necessarily the first in the SDP but has to be the first
988 * to be set up, else the second/third SETUP will fail with a 461.
990 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
991 rt->server_type == RTSP_SERVER_WMS) {
994 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
995 int len = strlen(rt->rtsp_streams[rtx]->control_url);
997 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1001 if (rtx == rt->nb_rtsp_streams)
1002 return -1; /* no RTX found */
1003 rtsp_st = rt->rtsp_streams[rtx];
1005 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1007 rtsp_st = rt->rtsp_streams[i];
1010 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1013 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1014 port = reply->transports[0].client_port_min;
1018 /* first try in specified port range */
1019 if (RTSP_RTP_PORT_MIN != 0) {
1020 while (j <= RTSP_RTP_PORT_MAX) {
1021 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1022 "?localport=%d", j);
1023 /* we will use two ports per rtp stream (rtp and rtcp) */
1025 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1031 /* then try on any port */
1032 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1033 err = AVERROR_INVALIDDATA;
1039 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1041 snprintf(transport, sizeof(transport) - 1,
1042 "%s/UDP;", trans_pref);
1043 if (rt->server_type != RTSP_SERVER_REAL)
1044 av_strlcat(transport, "unicast;", sizeof(transport));
1045 av_strlcatf(transport, sizeof(transport),
1046 "client_port=%d", port);
1047 if (rt->transport == RTSP_TRANSPORT_RTP &&
1048 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1049 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1053 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1054 /** For WMS streams, the application streams are only used for
1055 * UDP. When trying to set it up for TCP streams, the server
1056 * will return an error. Therefore, we skip those streams. */
1057 if (rt->server_type == RTSP_SERVER_WMS &&
1058 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1061 snprintf(transport, sizeof(transport) - 1,
1062 "%s/TCP;", trans_pref);
1063 if (rt->server_type == RTSP_SERVER_WMS)
1064 av_strlcat(transport, "unicast;", sizeof(transport));
1065 av_strlcatf(transport, sizeof(transport),
1066 "interleaved=%d-%d",
1067 interleave, interleave + 1);
1071 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1072 snprintf(transport, sizeof(transport) - 1,
1073 "%s/UDP;multicast", trans_pref);
1076 av_strlcat(transport, ";mode=receive", sizeof(transport));
1077 } else if (rt->server_type == RTSP_SERVER_REAL ||
1078 rt->server_type == RTSP_SERVER_WMS)
1079 av_strlcat(transport, ";mode=play", sizeof(transport));
1080 snprintf(cmd, sizeof(cmd),
1081 "Transport: %s\r\n",
1083 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1084 char real_res[41], real_csum[9];
1085 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1087 av_strlcatf(cmd, sizeof(cmd),
1089 "RealChallenge2: %s, sd=%s\r\n",
1090 rt->session_id, real_res, real_csum);
1092 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1093 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1096 } else if (reply->status_code != RTSP_STATUS_OK ||
1097 reply->nb_transports != 1) {
1098 err = AVERROR_INVALIDDATA;
1102 /* XXX: same protocol for all streams is required */
1104 if (reply->transports[0].lower_transport != rt->lower_transport ||
1105 reply->transports[0].transport != rt->transport) {
1106 err = AVERROR_INVALIDDATA;
1110 rt->lower_transport = reply->transports[0].lower_transport;
1111 rt->transport = reply->transports[0].transport;
1114 /* close RTP connection if not chosen */
1115 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1116 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1117 url_close(rtsp_st->rtp_handle);
1118 rtsp_st->rtp_handle = NULL;
1121 switch(reply->transports[0].lower_transport) {
1122 case RTSP_LOWER_TRANSPORT_TCP:
1123 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1124 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1127 case RTSP_LOWER_TRANSPORT_UDP: {
1130 /* Use source address if specified */
1131 if (reply->transports[0].source[0]) {
1132 ff_url_join(url, sizeof(url), "rtp", NULL,
1133 reply->transports[0].source,
1134 reply->transports[0].server_port_min, NULL);
1136 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1137 reply->transports[0].server_port_min, NULL);
1139 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1140 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1141 err = AVERROR_INVALIDDATA;
1144 /* Try to initialize the connection state in a
1145 * potential NAT router by sending dummy packets.
1146 * RTP/RTCP dummy packets are used for RDT, too.
1148 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1150 rtp_send_punch_packets(rtsp_st->rtp_handle);
1153 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1154 char url[1024], namebuf[50];
1155 struct sockaddr_storage addr;
1158 if (reply->transports[0].destination.ss_family) {
1159 addr = reply->transports[0].destination;
1160 port = reply->transports[0].port_min;
1161 ttl = reply->transports[0].ttl;
1163 addr = rtsp_st->sdp_ip;
1164 port = rtsp_st->sdp_port;
1165 ttl = rtsp_st->sdp_ttl;
1167 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1168 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1169 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1170 port, "?ttl=%d", ttl);
1171 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1172 err = AVERROR_INVALIDDATA;
1179 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1183 if (reply->timeout > 0)
1184 rt->timeout = reply->timeout;
1186 if (rt->server_type == RTSP_SERVER_REAL)
1187 rt->need_subscription = 1;
1192 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1193 if (rt->rtsp_streams[i]->rtp_handle) {
1194 url_close(rt->rtsp_streams[i]->rtp_handle);
1195 rt->rtsp_streams[i]->rtp_handle = NULL;
1201 void ff_rtsp_close_connections(AVFormatContext *s)
1203 RTSPState *rt = s->priv_data;
1204 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1205 url_close(rt->rtsp_hd);
1206 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1209 int ff_rtsp_connect(AVFormatContext *s)
1211 RTSPState *rt = s->priv_data;
1212 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1213 char *option_list, *option, *filename;
1214 int port, err, tcp_fd;
1215 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1216 int lower_transport_mask = 0;
1217 char real_challenge[64];
1218 struct sockaddr_storage peer;
1219 socklen_t peer_len = sizeof(peer);
1221 if (!ff_network_init())
1222 return AVERROR(EIO);
1224 rt->control_transport = RTSP_MODE_PLAIN;
1225 /* extract hostname and port */
1226 av_url_split(NULL, 0, auth, sizeof(auth),
1227 host, sizeof(host), &port, path, sizeof(path), s->filename);
1229 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1232 port = RTSP_DEFAULT_PORT;
1234 /* search for options */
1235 option_list = strrchr(path, '?');
1237 /* Strip out the RTSP specific options, write out the rest of
1238 * the options back into the same string. */
1239 filename = option_list;
1240 while (option_list) {
1241 /* move the option pointer */
1242 option = ++option_list;
1243 option_list = strchr(option_list, '&');
1247 /* handle the options */
1248 if (!strcmp(option, "udp")) {
1249 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1250 } else if (!strcmp(option, "multicast")) {
1251 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1252 } else if (!strcmp(option, "tcp")) {
1253 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1254 } else if(!strcmp(option, "http")) {
1255 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1256 rt->control_transport = RTSP_MODE_TUNNEL;
1258 /* Write options back into the buffer, using memmove instead
1259 * of strcpy since the strings may overlap. */
1260 int len = strlen(option);
1261 memmove(++filename, option, len);
1263 if (option_list) *filename = '&';
1269 if (!lower_transport_mask)
1270 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1273 /* Only UDP or TCP - UDP multicast isn't supported. */
1274 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1275 (1 << RTSP_LOWER_TRANSPORT_TCP);
1276 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1277 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1278 "only UDP and TCP are supported for output.\n");
1279 err = AVERROR(EINVAL);
1284 /* Construct the URI used in request; this is similar to s->filename,
1285 * but with authentication credentials removed and RTSP specific options
1287 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1288 host, port, "%s", path);
1290 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1291 /* set up initial handshake for tunneling */
1292 char httpname[1024];
1293 char sessioncookie[17];
1296 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1297 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1298 av_get_random_seed(), av_get_random_seed());
1301 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1306 /* generate GET headers */
1307 snprintf(headers, sizeof(headers),
1308 "x-sessioncookie: %s\r\n"
1309 "Accept: application/x-rtsp-tunnelled\r\n"
1310 "Pragma: no-cache\r\n"
1311 "Cache-Control: no-cache\r\n",
1313 ff_http_set_headers(rt->rtsp_hd, headers);
1315 /* complete the connection */
1316 if (url_connect(rt->rtsp_hd)) {
1322 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1327 /* generate POST headers */
1328 snprintf(headers, sizeof(headers),
1329 "x-sessioncookie: %s\r\n"
1330 "Content-Type: application/x-rtsp-tunnelled\r\n"
1331 "Pragma: no-cache\r\n"
1332 "Cache-Control: no-cache\r\n"
1333 "Content-Length: 32767\r\n"
1334 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1336 ff_http_set_headers(rt->rtsp_hd_out, headers);
1337 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1339 /* Initialize the authentication state for the POST session. The HTTP
1340 * protocol implementation doesn't properly handle multi-pass
1341 * authentication for POST requests, since it would require one of
1343 * - implementing Expect: 100-continue, which many HTTP servers
1344 * don't support anyway, even less the RTSP servers that do HTTP
1346 * - sending the whole POST data until getting a 401 reply specifying
1347 * what authentication method to use, then resending all that data
1348 * - waiting for potential 401 replies directly after sending the
1349 * POST header (waiting for some unspecified time)
1350 * Therefore, we copy the full auth state, which works for both basic
1351 * and digest. (For digest, we would have to synchronize the nonce
1352 * count variable between the two sessions, if we'd do more requests
1353 * with the original session, though.)
1355 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1357 /* complete the connection */
1358 if (url_connect(rt->rtsp_hd_out)) {
1363 /* open the tcp connection */
1364 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1365 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1369 rt->rtsp_hd_out = rt->rtsp_hd;
1373 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1374 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1375 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1376 NULL, 0, NI_NUMERICHOST);
1379 /* request options supported by the server; this also detects server
1381 for (rt->server_type = RTSP_SERVER_RTP;;) {
1383 if (rt->server_type == RTSP_SERVER_REAL)
1386 * The following entries are required for proper
1387 * streaming from a Realmedia server. They are
1388 * interdependent in some way although we currently
1389 * don't quite understand how. Values were copied
1390 * from mplayer SVN r23589.
1391 * @param CompanyID is a 16-byte ID in base64
1392 * @param ClientChallenge is a 16-byte ID in hex
1394 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1395 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1396 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1397 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1399 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1400 if (reply->status_code != RTSP_STATUS_OK) {
1401 err = AVERROR_INVALIDDATA;
1405 /* detect server type if not standard-compliant RTP */
1406 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1407 rt->server_type = RTSP_SERVER_REAL;
1409 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1410 rt->server_type = RTSP_SERVER_WMS;
1411 } else if (rt->server_type == RTSP_SERVER_REAL)
1412 strcpy(real_challenge, reply->real_challenge);
1416 if (s->iformat && CONFIG_RTSP_DEMUXER)
1417 err = ff_rtsp_setup_input_streams(s, reply);
1418 else if (CONFIG_RTSP_MUXER)
1419 err = ff_rtsp_setup_output_streams(s, host);
1424 int lower_transport = ff_log2_tab[lower_transport_mask &
1425 ~(lower_transport_mask - 1)];
1427 err = make_setup_request(s, host, port, lower_transport,
1428 rt->server_type == RTSP_SERVER_REAL ?
1429 real_challenge : NULL);
1432 lower_transport_mask &= ~(1 << lower_transport);
1433 if (lower_transport_mask == 0 && err == 1) {
1434 err = FF_NETERROR(EPROTONOSUPPORT);
1439 rt->state = RTSP_STATE_IDLE;
1440 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1443 ff_rtsp_close_streams(s);
1444 ff_rtsp_close_connections(s);
1445 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1446 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1447 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1455 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1458 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1459 uint8_t *buf, int buf_size, int64_t wait_end)
1461 RTSPState *rt = s->priv_data;
1462 RTSPStream *rtsp_st;
1464 int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1468 if (url_interrupt_cb())
1469 return AVERROR(EINTR);
1470 if (wait_end && wait_end - av_gettime() < 0)
1471 return AVERROR(EAGAIN);
1474 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1475 FD_SET(tcp_fd, &rfds);
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 (FFMAX(fd, fd_rtcp) > fd_max)
1486 fd_max = FFMAX(fd, fd_rtcp);
1488 FD_SET(fd_rtcp, &rfds);
1492 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1493 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1496 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1497 rtsp_st = rt->rtsp_streams[i];
1498 if (rtsp_st->rtp_handle) {
1499 fd = url_get_file_handle(rtsp_st->rtp_handle);
1500 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1501 if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
1502 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1504 *prtsp_st = rtsp_st;
1510 #if CONFIG_RTSP_DEMUXER
1511 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1512 RTSPMessageHeader reply;
1514 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1517 /* XXX: parse message */
1518 if (rt->state != RTSP_STATE_STREAMING)
1522 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1523 return FF_NETERROR(ETIMEDOUT);
1524 } else if (n < 0 && errno != EINTR)
1525 return AVERROR(errno);
1529 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1531 RTSPState *rt = s->priv_data;
1533 RTSPStream *rtsp_st, *first_queue_st = NULL;
1534 int64_t wait_end = 0;
1536 if (rt->nb_byes == rt->nb_rtsp_streams)
1539 /* get next frames from the same RTP packet */
1540 if (rt->cur_transport_priv) {
1541 if (rt->transport == RTSP_TRANSPORT_RDT) {
1542 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1544 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1546 rt->cur_transport_priv = NULL;
1548 } else if (ret == 1) {
1551 rt->cur_transport_priv = NULL;
1554 if (rt->transport == RTSP_TRANSPORT_RTP) {
1556 int64_t first_queue_time = 0;
1557 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1558 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1559 int64_t queue_time = ff_rtp_queued_packet_time(rtpctx);
1560 if (queue_time && (queue_time - first_queue_time < 0 ||
1561 !first_queue_time)) {
1562 first_queue_time = queue_time;
1563 first_queue_st = rt->rtsp_streams[i];
1566 if (first_queue_time)
1567 wait_end = first_queue_time + s->max_delay;
1570 /* read next RTP packet */
1573 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1575 return AVERROR(ENOMEM);
1578 switch(rt->lower_transport) {
1580 #if CONFIG_RTSP_DEMUXER
1581 case RTSP_LOWER_TRANSPORT_TCP:
1582 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1585 case RTSP_LOWER_TRANSPORT_UDP:
1586 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1587 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1588 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1589 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1592 if (len == AVERROR(EAGAIN) && first_queue_st &&
1593 rt->transport == RTSP_TRANSPORT_RTP) {
1594 rtsp_st = first_queue_st;
1595 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1602 if (rt->transport == RTSP_TRANSPORT_RDT) {
1603 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1605 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1607 /* Either bad packet, or a RTCP packet. Check if the
1608 * first_rtcp_ntp_time field was initialized. */
1609 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1610 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1611 /* first_rtcp_ntp_time has been initialized for this stream,
1612 * copy the same value to all other uninitialized streams,
1613 * in order to map their timestamp origin to the same ntp time
1616 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1617 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1619 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1620 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1623 if (ret == -RTCP_BYE) {
1626 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1627 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1629 if (rt->nb_byes == rt->nb_rtsp_streams)
1638 /* more packets may follow, so we save the RTP context */
1639 rt->cur_transport_priv = rtsp_st->transport_priv;
1643 #endif /* CONFIG_RTPDEC */
1645 #if CONFIG_SDP_DEMUXER
1646 static int sdp_probe(AVProbeData *p1)
1648 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1650 /* we look for a line beginning "c=IN IP" */
1651 while (p < p_end && *p != '\0') {
1652 if (p + sizeof("c=IN IP") - 1 < p_end &&
1653 av_strstart(p, "c=IN IP", NULL))
1654 return AVPROBE_SCORE_MAX / 2;
1656 while (p < p_end - 1 && *p != '\n') p++;
1665 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1667 RTSPState *rt = s->priv_data;
1668 RTSPStream *rtsp_st;
1673 if (!ff_network_init())
1674 return AVERROR(EIO);
1676 /* read the whole sdp file */
1677 /* XXX: better loading */
1678 content = av_malloc(SDP_MAX_SIZE);
1679 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1682 return AVERROR_INVALIDDATA;
1684 content[size] ='\0';
1686 ff_sdp_parse(s, content);
1689 /* open each RTP stream */
1690 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1692 rtsp_st = rt->rtsp_streams[i];
1694 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1695 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1696 ff_url_join(url, sizeof(url), "rtp", NULL,
1697 namebuf, rtsp_st->sdp_port,
1698 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1700 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1701 err = AVERROR_INVALIDDATA;
1704 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1709 ff_rtsp_close_streams(s);
1714 static int sdp_read_close(AVFormatContext *s)
1716 ff_rtsp_close_streams(s);
1721 AVInputFormat sdp_demuxer = {
1723 NULL_IF_CONFIG_SMALL("SDP"),
1727 ff_rtsp_fetch_packet,
1730 #endif /* CONFIG_SDP_DEMUXER */
1732 #if CONFIG_RTP_DEMUXER
1733 static int rtp_probe(AVProbeData *p)
1735 if (av_strstart(p->filename, "rtp:", NULL))
1736 return AVPROBE_SCORE_MAX;
1740 static int rtp_read_header(AVFormatContext *s,
1741 AVFormatParameters *ap)
1743 uint8_t recvbuf[1500];
1744 char host[500], sdp[500];
1746 URLContext* in = NULL;
1748 AVCodecContext codec;
1749 struct sockaddr_storage addr;
1751 socklen_t addrlen = sizeof(addr);
1753 if (!ff_network_init())
1754 return AVERROR(EIO);
1756 ret = url_open(&in, s->filename, URL_RDONLY);
1761 ret = url_read(in, recvbuf, sizeof(recvbuf));
1762 if (ret == AVERROR(EAGAIN))
1767 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1771 if ((recvbuf[0] & 0xc0) != 0x80) {
1772 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1777 payload_type = recvbuf[1] & 0x7f;
1780 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1784 memset(&codec, 0, sizeof(codec));
1785 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1786 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1787 "without an SDP file describing it\n",
1791 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1792 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1793 "properly you need an SDP file "
1797 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1798 NULL, 0, s->filename);
1800 snprintf(sdp, sizeof(sdp),
1801 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1802 addr.ss_family == AF_INET ? 4 : 6, host,
1803 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1804 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1805 port, payload_type);
1806 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1808 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1811 /* sdp_read_header initializes this again */
1814 ret = sdp_read_header(s, ap);
1825 AVInputFormat rtp_demuxer = {
1827 NULL_IF_CONFIG_SMALL("RTP input format"),
1831 ff_rtsp_fetch_packet,
1833 .flags = AVFMT_NOFILE,
1835 #endif /* CONFIG_RTP_DEMUXER */