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/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/bprint.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/parseutils.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/dict.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/time.h"
34 #include "avio_internal.h"
41 #include "os_support.h"
48 #include "rtpdec_formats.h"
49 #include "rtpenc_chain.h"
54 /* Default timeout values for read packet in seconds */
55 #define READ_PACKET_TIMEOUT_S 10
56 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
57 #define DEFAULT_REORDERING_DELAY 100000
59 #define OFFSET(x) offsetof(RTSPState, x)
60 #define DEC AV_OPT_FLAG_DECODING_PARAM
61 #define ENC AV_OPT_FLAG_ENCODING_PARAM
63 #define RTSP_FLAG_OPTS(name, longname) \
64 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
65 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
67 #define RTSP_MEDIATYPE_OPTS(name, longname) \
68 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
69 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
70 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
71 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
72 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
74 #define COMMON_OPTS() \
75 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
76 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
77 { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
82 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
90 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
91 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
92 { "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_SATIP_RAW}, 0, 0, DEC, "rtsp_flags" },
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97 #if FF_API_OLD_RTSP_OPTIONS
98 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC|AV_OPT_FLAG_DEPRECATED },
99 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
101 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
104 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
105 #if FF_API_OLD_RTSP_OPTIONS
106 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC|AV_OPT_FLAG_DEPRECATED },
111 static const AVOption sdp_options[] = {
112 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
113 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
114 { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
115 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
116 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
121 static const AVOption rtp_options[] = {
122 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
123 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
124 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
130 static AVDictionary *map_to_opts(RTSPState *rt)
132 AVDictionary *opts = NULL;
135 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
136 av_dict_set(&opts, "buffer_size", buf, 0);
137 snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
138 av_dict_set(&opts, "pkt_size", buf, 0);
143 static void get_word_until_chars(char *buf, int buf_size,
144 const char *sep, const char **pp)
150 p += strspn(p, SPACE_CHARS);
152 while (!strchr(sep, *p) && *p != '\0') {
153 if ((q - buf) < buf_size - 1)
162 static void get_word_sep(char *buf, int buf_size, const char *sep,
165 if (**pp == '/') (*pp)++;
166 get_word_until_chars(buf, buf_size, sep, pp);
169 static void get_word(char *buf, int buf_size, const char **pp)
171 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
174 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
176 * Used for seeking in the rtp stream.
178 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
182 p += strspn(p, SPACE_CHARS);
183 if (!av_stristart(p, "npt=", &p))
186 *start = AV_NOPTS_VALUE;
187 *end = AV_NOPTS_VALUE;
189 get_word_sep(buf, sizeof(buf), "-", &p);
190 if (av_parse_time(start, buf, 1) < 0)
194 get_word_sep(buf, sizeof(buf), "-", &p);
195 if (av_parse_time(end, buf, 1) < 0)
196 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
200 static int get_sockaddr(AVFormatContext *s,
201 const char *buf, struct sockaddr_storage *sock)
203 struct addrinfo hints = { 0 }, *ai = NULL;
206 hints.ai_flags = AI_NUMERICHOST;
207 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
208 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
213 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
219 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
220 RTSPStream *rtsp_st, AVStream *st)
222 AVCodecParameters *par = st ? st->codecpar : NULL;
226 par->codec_id = handler->codec_id;
227 rtsp_st->dynamic_handler = handler;
229 st->need_parsing = handler->need_parsing;
230 if (handler->priv_data_size) {
231 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
232 if (!rtsp_st->dynamic_protocol_context)
233 rtsp_st->dynamic_handler = NULL;
237 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
240 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
241 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
242 rtsp_st->dynamic_protocol_context);
244 if (rtsp_st->dynamic_protocol_context) {
245 if (rtsp_st->dynamic_handler->close)
246 rtsp_st->dynamic_handler->close(
247 rtsp_st->dynamic_protocol_context);
248 av_free(rtsp_st->dynamic_protocol_context);
250 rtsp_st->dynamic_protocol_context = NULL;
251 rtsp_st->dynamic_handler = NULL;
256 static int init_satip_stream(AVFormatContext *s)
258 RTSPState *rt = s->priv_data;
259 RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream));
261 return AVERROR(ENOMEM);
262 dynarray_add(&rt->rtsp_streams,
263 &rt->nb_rtsp_streams, rtsp_st);
265 rtsp_st->sdp_payload_type = 33; // MP2T
266 av_strlcpy(rtsp_st->control_url,
267 rt->control_uri, sizeof(rtsp_st->control_url));
269 if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) {
270 AVStream *st = avformat_new_stream(s, NULL);
272 return AVERROR(ENOMEM);
273 st->id = rt->nb_rtsp_streams - 1;
274 rtsp_st->stream_index = st->index;
275 st->codecpar->codec_type = AVMEDIA_TYPE_DATA;
276 st->codecpar->codec_id = AV_CODEC_ID_MPEG2TS;
278 rtsp_st->stream_index = -1;
279 init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
280 finalize_rtp_handler_init(s, rtsp_st, NULL);
285 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
286 static int sdp_parse_rtpmap(AVFormatContext *s,
287 AVStream *st, RTSPStream *rtsp_st,
288 int payload_type, const char *p)
290 AVCodecParameters *par = st->codecpar;
293 const AVCodecDescriptor *desc;
296 /* See if we can handle this kind of payload.
297 * The space should normally not be there but some Real streams or
298 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
299 * have a trailing space. */
300 get_word_sep(buf, sizeof(buf), "/ ", &p);
301 if (payload_type < RTP_PT_PRIVATE) {
302 /* We are in a standard case
303 * (from http://www.iana.org/assignments/rtp-parameters). */
304 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
307 if (par->codec_id == AV_CODEC_ID_NONE) {
308 const RTPDynamicProtocolHandler *handler =
309 ff_rtp_handler_find_by_name(buf, par->codec_type);
310 init_rtp_handler(handler, rtsp_st, st);
311 /* If no dynamic handler was found, check with the list of standard
312 * allocated types, if such a stream for some reason happens to
313 * use a private payload type. This isn't handled in rtpdec.c, since
314 * the format name from the rtpmap line never is passed into rtpdec. */
315 if (!rtsp_st->dynamic_handler)
316 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
319 desc = avcodec_descriptor_get(par->codec_id);
320 if (desc && desc->name)
325 get_word_sep(buf, sizeof(buf), "/", &p);
327 switch (par->codec_type) {
328 case AVMEDIA_TYPE_AUDIO:
329 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
330 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
331 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
333 par->sample_rate = i;
334 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
335 get_word_sep(buf, sizeof(buf), "/", &p);
340 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
342 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
345 case AVMEDIA_TYPE_VIDEO:
346 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
348 avpriv_set_pts_info(st, 32, 1, i);
353 finalize_rtp_handler_init(s, rtsp_st, st);
357 /* parse the attribute line from the fmtp a line of an sdp response. This
358 * is broken out as a function because it is used in rtp_h264.c, which is
360 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
361 char *value, int value_size)
363 *p += strspn(*p, SPACE_CHARS);
365 get_word_sep(attr, attr_size, "=", p);
368 get_word_sep(value, value_size, ";", p);
376 typedef struct SDPParseState {
378 struct sockaddr_storage default_ip;
380 int skip_media; ///< set if an unknown m= line occurs
381 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
382 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
383 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
384 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
387 char delayed_fmtp[2048];
390 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
391 struct RTSPSource ***dest, int *dest_count)
393 RTSPSource *rtsp_src, *rtsp_src2;
395 for (i = 0; i < count; i++) {
397 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
400 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
401 dynarray_add(dest, dest_count, rtsp_src2);
405 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
406 int payload_type, const char *line)
410 for (i = 0; i < rt->nb_rtsp_streams; i++) {
411 RTSPStream *rtsp_st = rt->rtsp_streams[i];
412 if (rtsp_st->sdp_payload_type == payload_type &&
413 rtsp_st->dynamic_handler &&
414 rtsp_st->dynamic_handler->parse_sdp_a_line) {
415 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
416 rtsp_st->dynamic_protocol_context, line);
421 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
422 int letter, const char *buf)
424 RTSPState *rt = s->priv_data;
425 char buf1[64], st_type[64];
427 enum AVMediaType codec_type;
431 RTSPSource *rtsp_src;
432 struct sockaddr_storage sdp_ip;
435 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
438 if (s1->skip_media && letter != 'm')
442 get_word(buf1, sizeof(buf1), &p);
443 if (strcmp(buf1, "IN") != 0)
445 get_word(buf1, sizeof(buf1), &p);
446 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
448 get_word_sep(buf1, sizeof(buf1), "/", &p);
449 if (get_sockaddr(s, buf1, &sdp_ip))
454 get_word_sep(buf1, sizeof(buf1), "/", &p);
457 if (s->nb_streams == 0) {
458 s1->default_ip = sdp_ip;
459 s1->default_ttl = ttl;
461 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
462 rtsp_st->sdp_ip = sdp_ip;
463 rtsp_st->sdp_ttl = ttl;
467 av_dict_set(&s->metadata, "title", p, 0);
470 if (s->nb_streams == 0) {
471 av_dict_set(&s->metadata, "comment", p, 0);
480 codec_type = AVMEDIA_TYPE_UNKNOWN;
481 get_word(st_type, sizeof(st_type), &p);
482 if (!strcmp(st_type, "audio")) {
483 codec_type = AVMEDIA_TYPE_AUDIO;
484 } else if (!strcmp(st_type, "video")) {
485 codec_type = AVMEDIA_TYPE_VIDEO;
486 } else if (!strcmp(st_type, "application")) {
487 codec_type = AVMEDIA_TYPE_DATA;
488 } else if (!strcmp(st_type, "text")) {
489 codec_type = AVMEDIA_TYPE_SUBTITLE;
491 if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
492 !(rt->media_type_mask & (1 << codec_type)) ||
493 rt->nb_rtsp_streams >= s->max_streams
498 rtsp_st = av_mallocz(sizeof(RTSPStream));
501 rtsp_st->stream_index = -1;
502 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
504 rtsp_st->sdp_ip = s1->default_ip;
505 rtsp_st->sdp_ttl = s1->default_ttl;
507 copy_default_source_addrs(s1->default_include_source_addrs,
508 s1->nb_default_include_source_addrs,
509 &rtsp_st->include_source_addrs,
510 &rtsp_st->nb_include_source_addrs);
511 copy_default_source_addrs(s1->default_exclude_source_addrs,
512 s1->nb_default_exclude_source_addrs,
513 &rtsp_st->exclude_source_addrs,
514 &rtsp_st->nb_exclude_source_addrs);
516 get_word(buf1, sizeof(buf1), &p); /* port */
517 rtsp_st->sdp_port = atoi(buf1);
519 get_word(buf1, sizeof(buf1), &p); /* protocol */
520 if (!strcmp(buf1, "udp"))
521 rt->transport = RTSP_TRANSPORT_RAW;
522 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
523 rtsp_st->feedback = 1;
525 /* XXX: handle list of formats */
526 get_word(buf1, sizeof(buf1), &p); /* format list */
527 rtsp_st->sdp_payload_type = atoi(buf1);
529 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
530 /* no corresponding stream */
531 if (rt->transport == RTSP_TRANSPORT_RAW) {
532 if (CONFIG_RTPDEC && !rt->ts)
533 rt->ts = avpriv_mpegts_parse_open(s);
535 const RTPDynamicProtocolHandler *handler;
536 handler = ff_rtp_handler_find_by_id(
537 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
538 init_rtp_handler(handler, rtsp_st, NULL);
539 finalize_rtp_handler_init(s, rtsp_st, NULL);
541 } else if (rt->server_type == RTSP_SERVER_WMS &&
542 codec_type == AVMEDIA_TYPE_DATA) {
543 /* RTX stream, a stream that carries all the other actual
544 * audio/video streams. Don't expose this to the callers. */
546 st = avformat_new_stream(s, NULL);
549 st->id = rt->nb_rtsp_streams - 1;
550 rtsp_st->stream_index = st->index;
551 st->codecpar->codec_type = codec_type;
552 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
553 const RTPDynamicProtocolHandler *handler;
554 /* if standard payload type, we can find the codec right now */
555 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
556 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
557 st->codecpar->sample_rate > 0)
558 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
559 /* Even static payload types may need a custom depacketizer */
560 handler = ff_rtp_handler_find_by_id(
561 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
562 init_rtp_handler(handler, rtsp_st, st);
563 finalize_rtp_handler_init(s, rtsp_st, st);
565 if (rt->default_lang[0])
566 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
568 /* put a default control url */
569 av_strlcpy(rtsp_st->control_url, rt->control_uri,
570 sizeof(rtsp_st->control_url));
573 if (av_strstart(p, "control:", &p)) {
574 if (rt->nb_rtsp_streams == 0) {
575 if (!strncmp(p, "rtsp://", 7))
576 av_strlcpy(rt->control_uri, p,
577 sizeof(rt->control_uri));
580 /* get the control url */
581 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
583 /* XXX: may need to add full url resolution */
584 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
586 if (proto[0] == '\0') {
587 /* relative control URL */
588 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
589 av_strlcat(rtsp_st->control_url, "/",
590 sizeof(rtsp_st->control_url));
591 av_strlcat(rtsp_st->control_url, p,
592 sizeof(rtsp_st->control_url));
594 av_strlcpy(rtsp_st->control_url, p,
595 sizeof(rtsp_st->control_url));
597 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
598 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
599 get_word(buf1, sizeof(buf1), &p);
600 payload_type = atoi(buf1);
601 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
602 if (rtsp_st->stream_index >= 0) {
603 st = s->streams[rtsp_st->stream_index];
604 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
608 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
610 } else if (av_strstart(p, "fmtp:", &p) ||
611 av_strstart(p, "framesize:", &p)) {
612 // let dynamic protocol handlers have a stab at the line.
613 get_word(buf1, sizeof(buf1), &p);
614 payload_type = atoi(buf1);
615 if (s1->seen_rtpmap) {
616 parse_fmtp(s, rt, payload_type, buf);
619 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
621 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
622 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
623 get_word(buf1, sizeof(buf1), &p);
624 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
625 } else if (av_strstart(p, "range:", &p)) {
628 // this is so that seeking on a streamed file can work.
629 rtsp_parse_range_npt(p, &start, &end);
630 s->start_time = start;
631 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
632 s->duration = (end == AV_NOPTS_VALUE) ?
633 AV_NOPTS_VALUE : end - start;
634 } else if (av_strstart(p, "lang:", &p)) {
635 if (s->nb_streams > 0) {
636 get_word(buf1, sizeof(buf1), &p);
637 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
638 if (rtsp_st->stream_index >= 0) {
639 st = s->streams[rtsp_st->stream_index];
640 av_dict_set(&st->metadata, "language", buf1, 0);
643 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
644 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
646 rt->transport = RTSP_TRANSPORT_RDT;
647 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
649 st = s->streams[s->nb_streams - 1];
650 st->codecpar->sample_rate = atoi(p);
651 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
653 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
654 get_word(buf1, sizeof(buf1), &p); // ignore tag
655 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
656 p += strspn(p, SPACE_CHARS);
657 if (av_strstart(p, "inline:", &p))
658 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
659 } else if (av_strstart(p, "source-filter:", &p)) {
661 get_word(buf1, sizeof(buf1), &p);
662 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
664 exclude = !strcmp(buf1, "excl");
666 get_word(buf1, sizeof(buf1), &p);
667 if (strcmp(buf1, "IN") != 0)
669 get_word(buf1, sizeof(buf1), &p);
670 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
672 // not checking that the destination address actually matches or is wildcard
673 get_word(buf1, sizeof(buf1), &p);
676 rtsp_src = av_mallocz(sizeof(*rtsp_src));
679 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
681 if (s->nb_streams == 0) {
682 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
684 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
685 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
688 if (s->nb_streams == 0) {
689 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
691 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
692 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
697 if (rt->server_type == RTSP_SERVER_WMS)
698 ff_wms_parse_sdp_a_line(s, p);
699 if (s->nb_streams > 0) {
700 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
702 if (rt->server_type == RTSP_SERVER_REAL)
703 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
705 if (rtsp_st->dynamic_handler &&
706 rtsp_st->dynamic_handler->parse_sdp_a_line)
707 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
708 rtsp_st->stream_index,
709 rtsp_st->dynamic_protocol_context, buf);
716 int ff_sdp_parse(AVFormatContext *s, const char *content)
720 char buf[SDP_MAX_SIZE], *q;
721 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
725 p += strspn(p, SPACE_CHARS);
733 /* get the content */
735 while (*p != '\n' && *p != '\r' && *p != '\0') {
736 if ((q - buf) < sizeof(buf) - 1)
741 sdp_parse_line(s, s1, letter, buf);
743 while (*p != '\n' && *p != '\0')
749 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
750 av_freep(&s1->default_include_source_addrs[i]);
751 av_freep(&s1->default_include_source_addrs);
752 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
753 av_freep(&s1->default_exclude_source_addrs[i]);
754 av_freep(&s1->default_exclude_source_addrs);
758 #endif /* CONFIG_RTPDEC */
760 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
762 RTSPState *rt = s->priv_data;
765 for (i = 0; i < rt->nb_rtsp_streams; i++) {
766 RTSPStream *rtsp_st = rt->rtsp_streams[i];
769 if (rtsp_st->transport_priv) {
771 AVFormatContext *rtpctx = rtsp_st->transport_priv;
772 av_write_trailer(rtpctx);
773 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
774 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
775 ff_rtsp_tcp_write_packet(s, rtsp_st);
776 ffio_free_dyn_buf(&rtpctx->pb);
778 avio_closep(&rtpctx->pb);
780 avformat_free_context(rtpctx);
781 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
782 ff_rdt_parse_close(rtsp_st->transport_priv);
783 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
784 ff_rtp_parse_close(rtsp_st->transport_priv);
786 rtsp_st->transport_priv = NULL;
787 ffurl_closep(&rtsp_st->rtp_handle);
791 /* close and free RTSP streams */
792 void ff_rtsp_close_streams(AVFormatContext *s)
794 RTSPState *rt = s->priv_data;
798 ff_rtsp_undo_setup(s, 0);
799 for (i = 0; i < rt->nb_rtsp_streams; i++) {
800 rtsp_st = rt->rtsp_streams[i];
802 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
803 if (rtsp_st->dynamic_handler->close)
804 rtsp_st->dynamic_handler->close(
805 rtsp_st->dynamic_protocol_context);
806 av_free(rtsp_st->dynamic_protocol_context);
808 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
809 av_freep(&rtsp_st->include_source_addrs[j]);
810 av_freep(&rtsp_st->include_source_addrs);
811 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
812 av_freep(&rtsp_st->exclude_source_addrs[j]);
813 av_freep(&rtsp_st->exclude_source_addrs);
818 av_freep(&rt->rtsp_streams);
820 avformat_close_input(&rt->asf_ctx);
822 if (CONFIG_RTPDEC && rt->ts)
823 avpriv_mpegts_parse_close(rt->ts);
825 av_freep(&rt->recvbuf);
828 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
830 RTSPState *rt = s->priv_data;
832 int reordering_queue_size = rt->reordering_queue_size;
833 if (reordering_queue_size < 0) {
834 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
835 reordering_queue_size = 0;
837 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
840 /* open the RTP context */
841 if (rtsp_st->stream_index >= 0)
842 st = s->streams[rtsp_st->stream_index];
844 s->ctx_flags |= AVFMTCTX_NOHEADER;
846 if (CONFIG_RTSP_MUXER && s->oformat && st) {
847 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
848 s, st, rtsp_st->rtp_handle,
849 RTSP_TCP_MAX_PACKET_SIZE,
850 rtsp_st->stream_index);
851 /* Ownership of rtp_handle is passed to the rtp mux context */
852 rtsp_st->rtp_handle = NULL;
855 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
856 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
857 return 0; // Don't need to open any parser here
858 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
859 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
860 rtsp_st->dynamic_protocol_context,
861 rtsp_st->dynamic_handler);
862 else if (CONFIG_RTPDEC)
863 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
864 rtsp_st->sdp_payload_type,
865 reordering_queue_size);
867 if (!rtsp_st->transport_priv) {
868 return AVERROR(ENOMEM);
869 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
871 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
872 rtpctx->ssrc = rtsp_st->ssrc;
873 if (rtsp_st->dynamic_handler) {
874 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
875 rtsp_st->dynamic_protocol_context,
876 rtsp_st->dynamic_handler);
878 if (rtsp_st->crypto_suite[0])
879 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
880 rtsp_st->crypto_suite,
881 rtsp_st->crypto_params);
887 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
888 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
895 q += strspn(q, SPACE_CHARS);
896 v = strtol(q, &p, 10);
900 v = strtol(p, &p, 10);
909 /* XXX: only one transport specification is parsed */
910 static void rtsp_parse_transport(AVFormatContext *s,
911 RTSPMessageHeader *reply, const char *p)
913 char transport_protocol[16];
915 char lower_transport[16];
917 RTSPTransportField *th;
920 reply->nb_transports = 0;
923 p += strspn(p, SPACE_CHARS);
927 th = &reply->transports[reply->nb_transports];
929 get_word_sep(transport_protocol, sizeof(transport_protocol),
931 if (!av_strcasecmp (transport_protocol, "rtp")) {
932 get_word_sep(profile, sizeof(profile), "/;,", &p);
933 lower_transport[0] = '\0';
934 /* rtp/avp/<protocol> */
936 get_word_sep(lower_transport, sizeof(lower_transport),
939 th->transport = RTSP_TRANSPORT_RTP;
940 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
941 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
942 /* x-pn-tng/<protocol> */
943 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
945 th->transport = RTSP_TRANSPORT_RDT;
946 } else if (!av_strcasecmp(transport_protocol, "raw")) {
947 get_word_sep(profile, sizeof(profile), "/;,", &p);
948 lower_transport[0] = '\0';
949 /* raw/raw/<protocol> */
951 get_word_sep(lower_transport, sizeof(lower_transport),
954 th->transport = RTSP_TRANSPORT_RAW;
956 if (!av_strcasecmp(lower_transport, "TCP"))
957 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
959 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
963 /* get each parameter */
964 while (*p != '\0' && *p != ',') {
965 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
966 if (!strcmp(parameter, "port")) {
969 rtsp_parse_range(&th->port_min, &th->port_max, &p);
971 } else if (!strcmp(parameter, "client_port")) {
974 rtsp_parse_range(&th->client_port_min,
975 &th->client_port_max, &p);
977 } else if (!strcmp(parameter, "server_port")) {
980 rtsp_parse_range(&th->server_port_min,
981 &th->server_port_max, &p);
983 } else if (!strcmp(parameter, "interleaved")) {
986 rtsp_parse_range(&th->interleaved_min,
987 &th->interleaved_max, &p);
989 } else if (!strcmp(parameter, "multicast")) {
990 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
991 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
992 } else if (!strcmp(parameter, "ttl")) {
996 th->ttl = strtol(p, &end, 10);
999 } else if (!strcmp(parameter, "destination")) {
1002 get_word_sep(buf, sizeof(buf), ";,", &p);
1003 get_sockaddr(s, buf, &th->destination);
1005 } else if (!strcmp(parameter, "source")) {
1008 get_word_sep(buf, sizeof(buf), ";,", &p);
1009 av_strlcpy(th->source, buf, sizeof(th->source));
1011 } else if (!strcmp(parameter, "mode")) {
1014 get_word_sep(buf, sizeof(buf), ";, ", &p);
1015 if (!strcmp(buf, "record") ||
1016 !strcmp(buf, "receive"))
1017 th->mode_record = 1;
1021 while (*p != ';' && *p != '\0' && *p != ',')
1029 reply->nb_transports++;
1030 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1035 static void handle_rtp_info(RTSPState *rt, const char *url,
1036 uint32_t seq, uint32_t rtptime)
1039 if (!rtptime || !url[0])
1041 if (rt->transport != RTSP_TRANSPORT_RTP)
1043 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1044 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1045 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1048 if (!strcmp(rtsp_st->control_url, url)) {
1049 rtpctx->base_timestamp = rtptime;
1055 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1058 char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
1059 uint32_t seq = 0, rtptime = 0;
1062 p += strspn(p, SPACE_CHARS);
1065 get_word_sep(key, sizeof(key), "=", &p);
1069 get_word_sep(value, sizeof(value), ";, ", &p);
1071 if (!strcmp(key, "url"))
1072 av_strlcpy(url, value, sizeof(url));
1073 else if (!strcmp(key, "seq"))
1074 seq = strtoul(value, NULL, 10);
1075 else if (!strcmp(key, "rtptime"))
1076 rtptime = strtoul(value, NULL, 10);
1078 handle_rtp_info(rt, url, seq, rtptime);
1087 handle_rtp_info(rt, url, seq, rtptime);
1090 void ff_rtsp_parse_line(AVFormatContext *s,
1091 RTSPMessageHeader *reply, const char *buf,
1092 RTSPState *rt, const char *method)
1096 /* NOTE: we do case independent match for broken servers */
1098 if (av_stristart(p, "Session:", &p)) {
1100 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1101 if (av_stristart(p, ";timeout=", &p) &&
1102 (t = strtol(p, NULL, 10)) > 0) {
1105 } else if (av_stristart(p, "Content-Length:", &p)) {
1106 reply->content_length = strtol(p, NULL, 10);
1107 } else if (av_stristart(p, "Transport:", &p)) {
1108 rtsp_parse_transport(s, reply, p);
1109 } else if (av_stristart(p, "CSeq:", &p)) {
1110 reply->seq = strtol(p, NULL, 10);
1111 } else if (av_stristart(p, "Range:", &p)) {
1112 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1113 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1114 p += strspn(p, SPACE_CHARS);
1115 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1116 } else if (av_stristart(p, "Server:", &p)) {
1117 p += strspn(p, SPACE_CHARS);
1118 av_strlcpy(reply->server, p, sizeof(reply->server));
1119 } else if (av_stristart(p, "Notice:", &p) ||
1120 av_stristart(p, "X-Notice:", &p)) {
1121 reply->notice = strtol(p, NULL, 10);
1122 } else if (av_stristart(p, "Location:", &p)) {
1123 p += strspn(p, SPACE_CHARS);
1124 av_strlcpy(reply->location, p , sizeof(reply->location));
1125 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1126 p += strspn(p, SPACE_CHARS);
1127 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1128 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1129 p += strspn(p, SPACE_CHARS);
1130 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1131 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1132 p += strspn(p, SPACE_CHARS);
1133 if (method && !strcmp(method, "DESCRIBE"))
1134 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1135 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1136 p += strspn(p, SPACE_CHARS);
1137 if (method && !strcmp(method, "PLAY"))
1138 rtsp_parse_rtp_info(rt, p);
1139 } else if (av_stristart(p, "Public:", &p) && rt) {
1140 if (strstr(p, "GET_PARAMETER") &&
1141 method && !strcmp(method, "OPTIONS"))
1142 rt->get_parameter_supported = 1;
1143 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1144 p += strspn(p, SPACE_CHARS);
1145 rt->accept_dynamic_rate = atoi(p);
1146 } else if (av_stristart(p, "Content-Type:", &p)) {
1147 p += strspn(p, SPACE_CHARS);
1148 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1149 } else if (av_stristart(p, "com.ses.streamID:", &p)) {
1150 p += strspn(p, SPACE_CHARS);
1151 av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
1155 /* skip a RTP/TCP interleaved packet */
1156 void ff_rtsp_skip_packet(AVFormatContext *s)
1158 RTSPState *rt = s->priv_data;
1160 uint8_t buf[MAX_URL_SIZE];
1162 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1165 len = AV_RB16(buf + 1);
1167 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1172 if (len1 > sizeof(buf))
1174 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1181 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1182 unsigned char **content_ptr,
1183 int return_on_interleaved_data, const char *method)
1185 RTSPState *rt = s->priv_data;
1186 char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
1189 int ret, content_length, line_count = 0, request = 0;
1190 unsigned char *content = NULL;
1196 memset(reply, 0, sizeof(*reply));
1198 /* parse reply (XXX: use buffers) */
1199 rt->last_reply[0] = '\0';
1203 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1204 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1209 if (ch == '$' && q == buf) {
1210 if (return_on_interleaved_data) {
1213 ff_rtsp_skip_packet(s);
1214 } else if (ch != '\r') {
1215 if ((q - buf) < sizeof(buf) - 1)
1221 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1223 /* test if last line */
1227 if (line_count == 0) {
1228 /* get reply code */
1229 get_word(buf1, sizeof(buf1), &p);
1230 if (!strncmp(buf1, "RTSP/", 5)) {
1231 get_word(buf1, sizeof(buf1), &p);
1232 reply->status_code = atoi(buf1);
1233 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1235 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1236 get_word(buf1, sizeof(buf1), &p); // object
1240 ff_rtsp_parse_line(s, reply, p, rt, method);
1241 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1242 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1247 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1248 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1250 content_length = reply->content_length;
1251 if (content_length > 0) {
1252 /* leave some room for a trailing '\0' (useful for simple parsing) */
1253 content = av_malloc(content_length + 1);
1255 return AVERROR(ENOMEM);
1256 if (ffurl_read_complete(rt->rtsp_hd, content, content_length) != content_length)
1257 return AVERROR(EIO);
1258 content[content_length] = '\0';
1261 *content_ptr = content;
1266 char buf[MAX_URL_SIZE];
1267 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1268 const char* ptr = buf;
1270 if (!strcmp(reply->reason, "OPTIONS")) {
1271 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1273 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1274 if (reply->session_id[0])
1275 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1278 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1280 av_strlcat(buf, "\r\n", sizeof(buf));
1282 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1283 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1286 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1288 rt->last_cmd_time = av_gettime_relative();
1289 /* Even if the request from the server had data, it is not the data
1290 * that the caller wants or expects. The memory could also be leaked
1291 * if the actual following reply has content data. */
1293 av_freep(content_ptr);
1294 /* If method is set, this is called from ff_rtsp_send_cmd,
1295 * where a reply to exactly this request is awaited. For
1296 * callers from within packet receiving, we just want to
1297 * return to the caller and go back to receiving packets. */
1303 if (rt->seq != reply->seq) {
1304 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1305 rt->seq, reply->seq);
1309 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1310 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1311 reply->notice == 2306 /* Continuous Feed Terminated */) {
1312 rt->state = RTSP_STATE_IDLE;
1313 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1314 return AVERROR(EIO); /* data or server error */
1315 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1316 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1317 return AVERROR(EPERM);
1323 * Send a command to the RTSP server without waiting for the reply.
1325 * @param s RTSP (de)muxer context
1326 * @param method the method for the request
1327 * @param url the target url for the request
1328 * @param headers extra header lines to include in the request
1329 * @param send_content if non-null, the data to send as request body content
1330 * @param send_content_length the length of the send_content data, or 0 if
1331 * send_content is null
1333 * @return zero if success, nonzero otherwise
1335 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1336 const char *method, const char *url,
1337 const char *headers,
1338 const unsigned char *send_content,
1339 int send_content_length)
1341 RTSPState *rt = s->priv_data;
1342 char buf[MAX_URL_SIZE], *out_buf;
1343 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1345 if (!rt->rtsp_hd_out)
1346 return AVERROR(ENOTCONN);
1348 /* Add in RTSP headers */
1351 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1353 av_strlcat(buf, headers, sizeof(buf));
1354 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1355 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1356 if (rt->session_id[0] != '\0' && (!headers ||
1357 !strstr(headers, "\nIf-Match:"))) {
1358 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1361 char *str = ff_http_auth_create_response(&rt->auth_state,
1362 rt->auth, url, method);
1364 av_strlcat(buf, str, sizeof(buf));
1367 if (send_content_length > 0 && send_content)
1368 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1369 av_strlcat(buf, "\r\n", sizeof(buf));
1371 /* base64 encode rtsp if tunneling */
1372 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1373 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1374 out_buf = base64buf;
1377 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1379 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1380 if (send_content_length > 0 && send_content) {
1381 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1382 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1383 return AVERROR_PATCHWELCOME;
1385 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1387 rt->last_cmd_time = av_gettime_relative();
1392 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1393 const char *url, const char *headers)
1395 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1398 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1399 const char *headers, RTSPMessageHeader *reply,
1400 unsigned char **content_ptr)
1402 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1403 content_ptr, NULL, 0);
1406 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1407 const char *method, const char *url,
1409 RTSPMessageHeader *reply,
1410 unsigned char **content_ptr,
1411 const unsigned char *send_content,
1412 int send_content_length)
1414 RTSPState *rt = s->priv_data;
1415 HTTPAuthType cur_auth_type;
1416 int ret, attempts = 0;
1419 cur_auth_type = rt->auth_state.auth_type;
1420 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1422 send_content_length)))
1425 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1429 if (reply->status_code == 401 &&
1430 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1431 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1434 if (reply->status_code > 400){
1435 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1439 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1445 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1446 int lower_transport, const char *real_challenge)
1448 RTSPState *rt = s->priv_data;
1449 int rtx = 0, j, i, err, interleave = 0, port_off;
1450 RTSPStream *rtsp_st;
1451 RTSPMessageHeader reply1, *reply = &reply1;
1452 char cmd[MAX_URL_SIZE];
1453 const char *trans_pref;
1455 if (rt->transport == RTSP_TRANSPORT_RDT)
1456 trans_pref = "x-pn-tng";
1457 else if (rt->transport == RTSP_TRANSPORT_RAW)
1458 trans_pref = "RAW/RAW";
1460 trans_pref = "RTP/AVP";
1462 /* default timeout: 1 minute */
1465 /* Choose a random starting offset within the first half of the
1466 * port range, to allow for a number of ports to try even if the offset
1467 * happens to be at the end of the random range. */
1468 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1469 /* even random offset */
1470 port_off -= port_off & 0x01;
1472 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1473 char transport[MAX_URL_SIZE];
1476 * WMS serves all UDP data over a single connection, the RTX, which
1477 * isn't necessarily the first in the SDP but has to be the first
1478 * to be set up, else the second/third SETUP will fail with a 461.
1480 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1481 rt->server_type == RTSP_SERVER_WMS) {
1484 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1485 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1487 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1491 if (rtx == rt->nb_rtsp_streams)
1492 return -1; /* no RTX found */
1493 rtsp_st = rt->rtsp_streams[rtx];
1495 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1497 rtsp_st = rt->rtsp_streams[i];
1500 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1503 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1504 port = reply->transports[0].client_port_min;
1508 /* first try in specified port range */
1509 while (j <= rt->rtp_port_max) {
1510 AVDictionary *opts = map_to_opts(rt);
1512 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1513 "?localport=%d", j);
1514 /* we will use two ports per rtp stream (rtp and rtcp) */
1516 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1517 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1519 av_dict_free(&opts);
1524 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1529 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1531 av_strlcpy(transport, trans_pref, sizeof(transport));
1532 av_strlcat(transport,
1533 rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
1535 if (rt->server_type != RTSP_SERVER_REAL)
1536 av_strlcat(transport, "unicast;", sizeof(transport));
1537 av_strlcatf(transport, sizeof(transport),
1538 "client_port=%d", port);
1539 if (rt->transport == RTSP_TRANSPORT_RTP &&
1540 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1541 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1545 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1546 /* For WMS streams, the application streams are only used for
1547 * UDP. When trying to set it up for TCP streams, the server
1548 * will return an error. Therefore, we skip those streams. */
1549 if (rt->server_type == RTSP_SERVER_WMS &&
1550 (rtsp_st->stream_index < 0 ||
1551 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1554 snprintf(transport, sizeof(transport) - 1,
1555 "%s/TCP;", trans_pref);
1556 if (rt->transport != RTSP_TRANSPORT_RDT)
1557 av_strlcat(transport, "unicast;", sizeof(transport));
1558 av_strlcatf(transport, sizeof(transport),
1559 "interleaved=%d-%d",
1560 interleave, interleave + 1);
1564 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1565 snprintf(transport, sizeof(transport) - 1,
1566 "%s/UDP;multicast", trans_pref);
1569 av_strlcat(transport, ";mode=record", sizeof(transport));
1570 } else if (rt->server_type == RTSP_SERVER_REAL ||
1571 rt->server_type == RTSP_SERVER_WMS)
1572 av_strlcat(transport, ";mode=play", sizeof(transport));
1573 snprintf(cmd, sizeof(cmd),
1574 "Transport: %s\r\n",
1576 if (rt->accept_dynamic_rate)
1577 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1578 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1579 char real_res[41], real_csum[9];
1580 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1582 av_strlcatf(cmd, sizeof(cmd),
1584 "RealChallenge2: %s, sd=%s\r\n",
1585 rt->session_id, real_res, real_csum);
1587 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1588 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1591 } else if (reply->status_code != RTSP_STATUS_OK ||
1592 reply->nb_transports != 1) {
1593 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1597 if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
1598 char proto[128], host[128], path[512], auth[128];
1600 av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
1601 &port, path, sizeof(path), rt->control_uri);
1602 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
1603 port, "/stream=%s", reply->stream_id);
1606 /* XXX: same protocol for all streams is required */
1608 if (reply->transports[0].lower_transport != rt->lower_transport ||
1609 reply->transports[0].transport != rt->transport) {
1610 err = AVERROR_INVALIDDATA;
1614 rt->lower_transport = reply->transports[0].lower_transport;
1615 rt->transport = reply->transports[0].transport;
1618 /* Fail if the server responded with another lower transport mode
1619 * than what we requested. */
1620 if (reply->transports[0].lower_transport != lower_transport) {
1621 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1622 err = AVERROR_INVALIDDATA;
1626 switch(reply->transports[0].lower_transport) {
1627 case RTSP_LOWER_TRANSPORT_TCP:
1628 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1629 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1632 case RTSP_LOWER_TRANSPORT_UDP: {
1633 char url[MAX_URL_SIZE], options[30] = "";
1634 const char *peer = host;
1636 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1637 av_strlcpy(options, "?connect=1", sizeof(options));
1638 /* Use source address if specified */
1639 if (reply->transports[0].source[0])
1640 peer = reply->transports[0].source;
1641 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1642 reply->transports[0].server_port_min, "%s", options);
1643 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1644 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1645 err = AVERROR_INVALIDDATA;
1650 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1651 char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
1652 struct sockaddr_storage addr;
1654 AVDictionary *opts = map_to_opts(rt);
1656 if (reply->transports[0].destination.ss_family) {
1657 addr = reply->transports[0].destination;
1658 port = reply->transports[0].port_min;
1659 ttl = reply->transports[0].ttl;
1661 addr = rtsp_st->sdp_ip;
1662 port = rtsp_st->sdp_port;
1663 ttl = rtsp_st->sdp_ttl;
1666 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1667 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1668 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1669 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1670 port, "%s", optbuf);
1671 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1672 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1673 av_dict_free(&opts);
1676 err = AVERROR_INVALIDDATA;
1683 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1687 if (rt->nb_rtsp_streams && reply->timeout > 0)
1688 rt->timeout = reply->timeout;
1690 if (rt->server_type == RTSP_SERVER_REAL)
1691 rt->need_subscription = 1;
1696 ff_rtsp_undo_setup(s, 0);
1700 void ff_rtsp_close_connections(AVFormatContext *s)
1702 RTSPState *rt = s->priv_data;
1703 if (rt->rtsp_hd_out != rt->rtsp_hd)
1704 ffurl_closep(&rt->rtsp_hd_out);
1705 rt->rtsp_hd_out = NULL;
1706 ffurl_closep(&rt->rtsp_hd);
1709 int ff_rtsp_connect(AVFormatContext *s)
1711 RTSPState *rt = s->priv_data;
1712 char proto[128], host[1024], path[1024];
1713 char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
1714 const char *lower_rtsp_proto = "tcp";
1715 int port, err, tcp_fd;
1716 RTSPMessageHeader reply1, *reply = &reply1;
1717 int lower_transport_mask = 0;
1718 int default_port = RTSP_DEFAULT_PORT;
1719 int https_tunnel = 0;
1720 char real_challenge[64] = "";
1721 struct sockaddr_storage peer;
1722 socklen_t peer_len = sizeof(peer);
1724 if (rt->rtp_port_max < rt->rtp_port_min) {
1725 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1726 "than min port %d\n", rt->rtp_port_max,
1728 return AVERROR(EINVAL);
1731 if (!ff_network_init())
1732 return AVERROR(EIO);
1734 if (s->max_delay < 0) /* Not set by the caller */
1735 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1737 rt->control_transport = RTSP_MODE_PLAIN;
1738 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1739 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1740 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1741 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1742 rt->control_transport = RTSP_MODE_TUNNEL;
1744 /* Only pass through valid flags from here */
1745 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1748 memset(&reply1, 0, sizeof(reply1));
1749 /* extract hostname and port */
1750 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1751 host, sizeof(host), &port, path, sizeof(path), s->url);
1753 if (!strcmp(proto, "rtsps")) {
1754 lower_rtsp_proto = "tls";
1755 default_port = RTSPS_DEFAULT_PORT;
1756 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1757 } else if (!strcmp(proto, "satip")) {
1758 av_strlcpy(proto, "rtsp", sizeof(proto));
1759 rt->server_type = RTSP_SERVER_SATIP;
1763 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1766 port = default_port;
1768 lower_transport_mask = rt->lower_transport_mask;
1770 if (!lower_transport_mask)
1771 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1774 /* Only UDP or TCP - UDP multicast isn't supported. */
1775 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1776 (1 << RTSP_LOWER_TRANSPORT_TCP);
1777 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1778 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1779 "only UDP and TCP are supported for output.\n");
1780 err = AVERROR(EINVAL);
1785 /* Construct the URI used in request; this is similar to s->url,
1786 * but with authentication credentials removed and RTSP specific options
1788 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1789 host, port, "%s", path);
1791 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1792 /* set up initial handshake for tunneling */
1793 char httpname[1024];
1794 char sessioncookie[17];
1796 AVDictionary *options = NULL;
1798 av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1800 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1801 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1802 av_get_random_seed(), av_get_random_seed());
1805 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1806 &s->interrupt_callback) < 0) {
1811 /* generate GET headers */
1812 snprintf(headers, sizeof(headers),
1813 "x-sessioncookie: %s\r\n"
1814 "Accept: application/x-rtsp-tunnelled\r\n"
1815 "Pragma: no-cache\r\n"
1816 "Cache-Control: no-cache\r\n",
1818 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1820 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1821 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1822 if (!rt->rtsp_hd->protocol_whitelist) {
1823 err = AVERROR(ENOMEM);
1828 /* complete the connection */
1829 if (ffurl_connect(rt->rtsp_hd, &options)) {
1830 av_dict_free(&options);
1836 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1837 &s->interrupt_callback) < 0 ) {
1842 /* generate POST headers */
1843 snprintf(headers, sizeof(headers),
1844 "x-sessioncookie: %s\r\n"
1845 "Content-Type: application/x-rtsp-tunnelled\r\n"
1846 "Pragma: no-cache\r\n"
1847 "Cache-Control: no-cache\r\n"
1848 "Content-Length: 32767\r\n"
1849 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1851 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1852 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1853 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1855 /* Initialize the authentication state for the POST session. The HTTP
1856 * protocol implementation doesn't properly handle multi-pass
1857 * authentication for POST requests, since it would require one of
1859 * - implementing Expect: 100-continue, which many HTTP servers
1860 * don't support anyway, even less the RTSP servers that do HTTP
1862 * - sending the whole POST data until getting a 401 reply specifying
1863 * what authentication method to use, then resending all that data
1864 * - waiting for potential 401 replies directly after sending the
1865 * POST header (waiting for some unspecified time)
1866 * Therefore, we copy the full auth state, which works for both basic
1867 * and digest. (For digest, we would have to synchronize the nonce
1868 * count variable between the two sessions, if we'd do more requests
1869 * with the original session, though.)
1871 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1873 /* complete the connection */
1874 if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1875 av_dict_free(&options);
1879 av_dict_free(&options);
1882 /* open the tcp connection */
1883 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1885 "?timeout=%d", rt->stimeout);
1886 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1887 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1891 rt->rtsp_hd_out = rt->rtsp_hd;
1895 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1900 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1901 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1902 NULL, 0, NI_NUMERICHOST);
1905 /* request options supported by the server; this also detects server
1907 if (rt->server_type != RTSP_SERVER_SATIP)
1908 rt->server_type = RTSP_SERVER_RTP;
1911 if (rt->server_type == RTSP_SERVER_REAL)
1914 * The following entries are required for proper
1915 * streaming from a Realmedia server. They are
1916 * interdependent in some way although we currently
1917 * don't quite understand how. Values were copied
1918 * from mplayer SVN r23589.
1919 * ClientChallenge is a 16-byte ID in hex
1920 * CompanyID is a 16-byte ID in base64
1922 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1923 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1924 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1925 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1927 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1928 if (reply->status_code != RTSP_STATUS_OK) {
1929 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1933 /* detect server type if not standard-compliant RTP */
1934 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1935 rt->server_type = RTSP_SERVER_REAL;
1937 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1938 rt->server_type = RTSP_SERVER_WMS;
1939 } else if (rt->server_type == RTSP_SERVER_REAL)
1940 strcpy(real_challenge, reply->real_challenge);
1944 #if CONFIG_RTSP_DEMUXER
1946 if (rt->server_type == RTSP_SERVER_SATIP)
1947 err = init_satip_stream(s);
1949 err = ff_rtsp_setup_input_streams(s, reply);
1952 if (CONFIG_RTSP_MUXER)
1953 err = ff_rtsp_setup_output_streams(s, host);
1960 int lower_transport = ff_log2_tab[lower_transport_mask &
1961 ~(lower_transport_mask - 1)];
1963 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1964 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1965 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1967 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1968 rt->server_type == RTSP_SERVER_REAL ?
1969 real_challenge : NULL);
1972 lower_transport_mask &= ~(1 << lower_transport);
1973 if (lower_transport_mask == 0 && err == 1) {
1974 err = AVERROR(EPROTONOSUPPORT);
1979 rt->lower_transport_mask = lower_transport_mask;
1980 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1981 rt->state = RTSP_STATE_IDLE;
1982 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1985 ff_rtsp_close_streams(s);
1986 ff_rtsp_close_connections(s);
1987 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1988 char *new_url = av_strdup(reply->location);
1990 err = AVERROR(ENOMEM);
1993 ff_format_set_url(s, new_url);
1994 rt->session_id[0] = '\0';
1995 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
2004 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
2007 static int parse_rtsp_message(AVFormatContext *s)
2009 RTSPState *rt = s->priv_data;
2012 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
2013 if (rt->state == RTSP_STATE_STREAMING) {
2014 return ff_rtsp_parse_streaming_commands(s);
2018 RTSPMessageHeader reply;
2019 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
2022 /* XXX: parse message */
2023 if (rt->state != RTSP_STATE_STREAMING)
2030 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
2031 uint8_t *buf, int buf_size, int64_t wait_end)
2033 RTSPState *rt = s->priv_data;
2034 RTSPStream *rtsp_st;
2036 struct pollfd *p = rt->p;
2037 int *fds = NULL, fdsnum, fdsidx;
2038 int runs = rt->initial_timeout * 1000LL / POLLING_TIME;
2041 p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
2043 return AVERROR(ENOMEM);
2046 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2047 p[rt->max_p++].events = POLLIN;
2049 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2050 rtsp_st = rt->rtsp_streams[i];
2051 if (rtsp_st->rtp_handle) {
2052 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2054 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2058 av_log(s, AV_LOG_ERROR,
2059 "Number of fds %d not supported\n", fdsnum);
2060 return AVERROR_INVALIDDATA;
2062 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2063 p[rt->max_p].fd = fds[fdsidx];
2064 p[rt->max_p++].events = POLLIN;
2072 if (ff_check_interrupt(&s->interrupt_callback))
2073 return AVERROR_EXIT;
2074 if (wait_end && wait_end - av_gettime_relative() < 0)
2075 return AVERROR(EAGAIN);
2076 n = poll(p, rt->max_p, POLLING_TIME);
2078 int j = rt->rtsp_hd ? 1 : 0;
2079 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2080 rtsp_st = rt->rtsp_streams[i];
2081 if (rtsp_st->rtp_handle) {
2082 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2083 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2085 *prtsp_st = rtsp_st;
2092 #if CONFIG_RTSP_DEMUXER
2093 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2094 if ((ret = parse_rtsp_message(s)) < 0) {
2099 } else if (n == 0 && rt->initial_timeout > 0 && --runs <= 0) {
2100 return AVERROR(ETIMEDOUT);
2101 } else if (n < 0 && errno != EINTR)
2102 return AVERROR(errno);
2106 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2107 const uint8_t *buf, int len)
2109 RTSPState *rt = s->priv_data;
2113 if (rt->nb_rtsp_streams == 1) {
2114 *rtsp_st = rt->rtsp_streams[0];
2117 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2118 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2120 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2121 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2124 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2125 *rtsp_st = rt->rtsp_streams[i];
2132 av_log(s, AV_LOG_WARNING,
2133 "Unable to pick stream for packet - SSRC not known for "
2135 return AVERROR(EAGAIN);
2138 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2139 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2140 *rtsp_st = rt->rtsp_streams[i];
2146 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2147 return AVERROR(EAGAIN);
2150 static int read_packet(AVFormatContext *s,
2151 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2154 RTSPState *rt = s->priv_data;
2157 switch(rt->lower_transport) {
2159 #if CONFIG_RTSP_DEMUXER
2160 case RTSP_LOWER_TRANSPORT_TCP:
2161 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2164 case RTSP_LOWER_TRANSPORT_UDP:
2165 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2166 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2167 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2168 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2170 case RTSP_LOWER_TRANSPORT_CUSTOM:
2171 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2172 wait_end && wait_end < av_gettime_relative())
2173 len = AVERROR(EAGAIN);
2175 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2176 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2177 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2178 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2188 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2190 RTSPState *rt = s->priv_data;
2192 RTSPStream *rtsp_st, *first_queue_st = NULL;
2193 int64_t wait_end = 0;
2195 if (rt->nb_byes == rt->nb_rtsp_streams)
2198 /* get next frames from the same RTP packet */
2199 if (rt->cur_transport_priv) {
2200 if (rt->transport == RTSP_TRANSPORT_RDT) {
2201 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2202 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2203 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2204 } else if (CONFIG_RTPDEC && rt->ts) {
2205 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2207 rt->recvbuf_pos += ret;
2208 ret = rt->recvbuf_pos < rt->recvbuf_len;
2213 rt->cur_transport_priv = NULL;
2215 } else if (ret == 1) {
2218 rt->cur_transport_priv = NULL;
2222 if (rt->transport == RTSP_TRANSPORT_RTP) {
2224 int64_t first_queue_time = 0;
2225 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2226 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2230 queue_time = ff_rtp_queued_packet_time(rtpctx);
2231 if (queue_time && (queue_time - first_queue_time < 0 ||
2232 !first_queue_time)) {
2233 first_queue_time = queue_time;
2234 first_queue_st = rt->rtsp_streams[i];
2237 if (first_queue_time) {
2238 wait_end = first_queue_time + s->max_delay;
2241 first_queue_st = NULL;
2245 /* read next RTP packet */
2247 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2249 return AVERROR(ENOMEM);
2252 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2253 if (len == AVERROR(EAGAIN) && first_queue_st &&
2254 rt->transport == RTSP_TRANSPORT_RTP) {
2255 av_log(s, AV_LOG_WARNING,
2256 "max delay reached. need to consume packet\n");
2257 rtsp_st = first_queue_st;
2258 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2264 if (rt->transport == RTSP_TRANSPORT_RDT) {
2265 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2266 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2267 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2268 if (rtsp_st->feedback) {
2269 AVIOContext *pb = NULL;
2270 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2272 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2275 /* Either bad packet, or a RTCP packet. Check if the
2276 * first_rtcp_ntp_time field was initialized. */
2277 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2278 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2279 /* first_rtcp_ntp_time has been initialized for this stream,
2280 * copy the same value to all other uninitialized streams,
2281 * in order to map their timestamp origin to the same ntp time
2284 AVStream *st = NULL;
2285 if (rtsp_st->stream_index >= 0)
2286 st = s->streams[rtsp_st->stream_index];
2287 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2288 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2289 AVStream *st2 = NULL;
2290 if (rt->rtsp_streams[i]->stream_index >= 0)
2291 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2292 if (rtpctx2 && st && st2 &&
2293 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2294 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2295 rtpctx2->rtcp_ts_offset = av_rescale_q(
2296 rtpctx->rtcp_ts_offset, st->time_base,
2300 // Make real NTP start time available in AVFormatContext
2301 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2302 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2304 s->start_time_realtime -=
2305 av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
2309 if (ret == -RTCP_BYE) {
2312 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2313 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2315 if (rt->nb_byes == rt->nb_rtsp_streams)
2319 } else if (CONFIG_RTPDEC && rt->ts) {
2320 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2323 rt->recvbuf_len = len;
2324 rt->recvbuf_pos = ret;
2325 rt->cur_transport_priv = rt->ts;
2332 return AVERROR_INVALIDDATA;
2338 /* more packets may follow, so we save the RTP context */
2339 rt->cur_transport_priv = rtsp_st->transport_priv;
2343 #endif /* CONFIG_RTPDEC */
2345 #if CONFIG_SDP_DEMUXER
2346 static int sdp_probe(const AVProbeData *p1)
2348 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2350 /* we look for a line beginning "c=IN IP" */
2351 while (p < p_end && *p != '\0') {
2352 if (sizeof("c=IN IP") - 1 < p_end - p &&
2353 av_strstart(p, "c=IN IP", NULL))
2354 return AVPROBE_SCORE_EXTENSION;
2356 while (p < p_end - 1 && *p != '\n') p++;
2365 static void append_source_addrs(char *buf, int size, const char *name,
2366 int count, struct RTSPSource **addrs)
2371 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2372 for (i = 1; i < count; i++)
2373 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2376 static int sdp_read_header(AVFormatContext *s)
2378 RTSPState *rt = s->priv_data;
2379 RTSPStream *rtsp_st;
2382 char url[MAX_URL_SIZE];
2384 if (!ff_network_init())
2385 return AVERROR(EIO);
2387 if (s->max_delay < 0) /* Not set by the caller */
2388 s->max_delay = DEFAULT_REORDERING_DELAY;
2389 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2390 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2392 /* read the whole sdp file */
2393 /* XXX: better loading */
2394 content = av_malloc(SDP_MAX_SIZE);
2397 return AVERROR(ENOMEM);
2399 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2403 return AVERROR_INVALIDDATA;
2405 content[size] ='\0';
2407 err = ff_sdp_parse(s, content);
2411 /* open each RTP stream */
2412 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2414 rtsp_st = rt->rtsp_streams[i];
2416 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2417 AVDictionary *opts = map_to_opts(rt);
2419 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2420 sizeof(rtsp_st->sdp_ip),
2421 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2423 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2425 av_dict_free(&opts);
2428 ff_url_join(url, sizeof(url), "rtp", NULL,
2429 namebuf, rtsp_st->sdp_port,
2430 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2431 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2432 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2433 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2435 append_source_addrs(url, sizeof(url), "sources",
2436 rtsp_st->nb_include_source_addrs,
2437 rtsp_st->include_source_addrs);
2438 append_source_addrs(url, sizeof(url), "block",
2439 rtsp_st->nb_exclude_source_addrs,
2440 rtsp_st->exclude_source_addrs);
2441 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2442 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2444 av_dict_free(&opts);
2447 err = AVERROR_INVALIDDATA;
2451 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2456 ff_rtsp_close_streams(s);
2461 static int sdp_read_close(AVFormatContext *s)
2463 ff_rtsp_close_streams(s);
2468 static const AVClass sdp_demuxer_class = {
2469 .class_name = "SDP demuxer",
2470 .item_name = av_default_item_name,
2471 .option = sdp_options,
2472 .version = LIBAVUTIL_VERSION_INT,
2475 AVInputFormat ff_sdp_demuxer = {
2477 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2478 .priv_data_size = sizeof(RTSPState),
2479 .read_probe = sdp_probe,
2480 .read_header = sdp_read_header,
2481 .read_packet = ff_rtsp_fetch_packet,
2482 .read_close = sdp_read_close,
2483 .priv_class = &sdp_demuxer_class,
2485 #endif /* CONFIG_SDP_DEMUXER */
2487 #if CONFIG_RTP_DEMUXER
2488 static int rtp_probe(const AVProbeData *p)
2490 if (av_strstart(p->filename, "rtp:", NULL))
2491 return AVPROBE_SCORE_MAX;
2495 static int rtp_read_header(AVFormatContext *s)
2497 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2498 char host[500], filters_buf[1000];
2500 URLContext* in = NULL;
2502 AVCodecParameters *par = NULL;
2503 struct sockaddr_storage addr;
2505 socklen_t addrlen = sizeof(addr);
2506 RTSPState *rt = s->priv_data;
2510 if (!ff_network_init())
2511 return AVERROR(EIO);
2513 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2514 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2519 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2520 if (ret == AVERROR(EAGAIN))
2525 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2529 if ((recvbuf[0] & 0xc0) != 0x80) {
2530 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2535 if (RTP_PT_IS_RTCP(recvbuf[1]))
2538 payload_type = recvbuf[1] & 0x7f;
2541 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2544 par = avcodec_parameters_alloc();
2546 ret = AVERROR(ENOMEM);
2550 if (ff_rtp_get_codec_info(par, payload_type)) {
2551 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2552 "without an SDP file describing it\n",
2554 ret = AVERROR_INVALIDDATA;
2557 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2558 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2559 "properly you need an SDP file "
2563 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2566 av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
2567 av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2568 addr.ss_family == AF_INET ? 4 : 6, host);
2570 p = strchr(s->url, '?');
2572 static const char filters[][2][8] = { { "sources", "incl" },
2573 { "block", "excl" } };
2576 for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2577 if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2579 while ((q = strchr(q, ',')) != NULL)
2581 av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2583 addr.ss_family == AF_INET ? 4 : 6, host,
2589 av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2590 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2591 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2592 port, payload_type);
2593 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2594 if (!av_bprint_is_complete(&sdp))
2596 avcodec_parameters_free(&par);
2598 ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2601 /* if sdp_read_header() fails then following ff_network_close() cancels out */
2602 /* ff_network_init() at the start of this function. Otherwise it cancels out */
2603 /* ff_network_init() inside sdp_read_header() */
2606 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2608 ret = sdp_read_header(s);
2610 av_bprint_finalize(&sdp, NULL);
2614 ret = AVERROR(ENOMEM);
2615 av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2616 av_bprint_finalize(&sdp, NULL);
2618 avcodec_parameters_free(&par);
2624 static const AVClass rtp_demuxer_class = {
2625 .class_name = "RTP demuxer",
2626 .item_name = av_default_item_name,
2627 .option = rtp_options,
2628 .version = LIBAVUTIL_VERSION_INT,
2631 AVInputFormat ff_rtp_demuxer = {
2633 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2634 .priv_data_size = sizeof(RTSPState),
2635 .read_probe = rtp_probe,
2636 .read_header = rtp_read_header,
2637 .read_packet = ff_rtsp_fetch_packet,
2638 .read_close = sdp_read_close,
2639 .flags = AVFMT_NOFILE,
2640 .priv_class = &rtp_demuxer_class,
2642 #endif /* CONFIG_RTP_DEMUXER */