3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; 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/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define RTSP_FLAG_OPTS(name, longname) \
66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
68 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76 const AVOption ff_rtsp_options
[] = {
77 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause
), AV_OPT_TYPE_INT
, {0}, 0, 1, DEC
},
78 FF_RTP_FLAG_OPTS(RTSPState
, rtp_muxer_flags
),
79 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask
), AV_OPT_TYPE_FLAGS
, {0}, INT_MIN
, INT_MAX
, DEC
|ENC
, "rtsp_transport" }, \
80 { "udp", "UDP", 0, AV_OPT_TYPE_CONST
, {1 << RTSP_LOWER_TRANSPORT_UDP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
81 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST
, {1 << RTSP_LOWER_TRANSPORT_TCP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
82 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST
, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST
}, 0, 0, DEC
, "rtsp_transport" },
83 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST
, {(1 << RTSP_LOWER_TRANSPORT_HTTP
)}, 0, 0, DEC
, "rtsp_transport" },
84 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
85 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
86 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min
), AV_OPT_TYPE_INT
, {RTSP_RTP_PORT_MIN
}, 0, 65535, DEC
|ENC
},
87 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max
), AV_OPT_TYPE_INT
, {RTSP_RTP_PORT_MAX
}, 0, 65535, DEC
|ENC
},
88 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout
), AV_OPT_TYPE_INT
, {-1}, INT_MIN
, INT_MAX
, DEC
},
92 static const AVOption sdp_options
[] = {
93 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
94 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
98 static const AVOption rtp_options
[] = {
99 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
103 static void get_word_until_chars(char *buf
, int buf_size
,
104 const char *sep
, const char **pp
)
110 p
+= strspn(p
, SPACE_CHARS
);
112 while (!strchr(sep
, *p
) && *p
!= '\0') {
113 if ((q
- buf
) < buf_size
- 1)
122 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
125 if (**pp
== '/') (*pp
)++;
126 get_word_until_chars(buf
, buf_size
, sep
, pp
);
129 static void get_word(char *buf
, int buf_size
, const char **pp
)
131 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
134 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
136 * Used for seeking in the rtp stream.
138 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
142 p
+= strspn(p
, SPACE_CHARS
);
143 if (!av_stristart(p
, "npt=", &p
))
146 *start
= AV_NOPTS_VALUE
;
147 *end
= AV_NOPTS_VALUE
;
149 get_word_sep(buf
, sizeof(buf
), "-", &p
);
150 av_parse_time(start
, buf
, 1);
153 get_word_sep(buf
, sizeof(buf
), "-", &p
);
154 av_parse_time(end
, buf
, 1);
156 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
157 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
160 static int get_sockaddr(const char *buf
, struct sockaddr_storage
*sock
)
162 struct addrinfo hints
= { 0 }, *ai
= NULL
;
163 hints
.ai_flags
= AI_NUMERICHOST
;
164 if (getaddrinfo(buf
, NULL
, &hints
, &ai
))
166 memcpy(sock
, ai
->ai_addr
, FFMIN(sizeof(*sock
), ai
->ai_addrlen
));
172 static void init_rtp_handler(RTPDynamicProtocolHandler
*handler
,
173 RTSPStream
*rtsp_st
, AVCodecContext
*codec
)
177 codec
->codec_id
= handler
->codec_id
;
178 rtsp_st
->dynamic_handler
= handler
;
179 if (handler
->alloc
) {
180 rtsp_st
->dynamic_protocol_context
= handler
->alloc();
181 if (!rtsp_st
->dynamic_protocol_context
)
182 rtsp_st
->dynamic_handler
= NULL
;
186 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
187 static int sdp_parse_rtpmap(AVFormatContext
*s
,
188 AVStream
*st
, RTSPStream
*rtsp_st
,
189 int payload_type
, const char *p
)
191 AVCodecContext
*codec
= st
->codec
;
197 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
198 * see if we can handle this kind of payload.
199 * The space should normally not be there but some Real streams or
200 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
201 * have a trailing space. */
202 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
203 if (payload_type
< RTP_PT_PRIVATE
) {
204 /* We are in a standard case
205 * (from http://www.iana.org/assignments/rtp-parameters). */
206 /* search into AVRtpPayloadTypes[] */
207 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
210 if (codec
->codec_id
== AV_CODEC_ID_NONE
) {
211 RTPDynamicProtocolHandler
*handler
=
212 ff_rtp_handler_find_by_name(buf
, codec
->codec_type
);
213 init_rtp_handler(handler
, rtsp_st
, codec
);
214 /* If no dynamic handler was found, check with the list of standard
215 * allocated types, if such a stream for some reason happens to
216 * use a private payload type. This isn't handled in rtpdec.c, since
217 * the format name from the rtpmap line never is passed into rtpdec. */
218 if (!rtsp_st
->dynamic_handler
)
219 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
222 c
= avcodec_find_decoder(codec
->codec_id
);
228 get_word_sep(buf
, sizeof(buf
), "/", &p
);
230 switch (codec
->codec_type
) {
231 case AVMEDIA_TYPE_AUDIO
:
232 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
233 codec
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
234 codec
->channels
= RTSP_DEFAULT_NB_AUDIO_CHANNELS
;
236 codec
->sample_rate
= i
;
237 avpriv_set_pts_info(st
, 32, 1, codec
->sample_rate
);
238 get_word_sep(buf
, sizeof(buf
), "/", &p
);
242 // TODO: there is a bug here; if it is a mono stream, and
243 // less than 22000Hz, faad upconverts to stereo and twice
244 // the frequency. No problem, but the sample rate is being
245 // set here by the sdp line. Patch on its way. (rdm)
247 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
249 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
252 case AVMEDIA_TYPE_VIDEO
:
253 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
255 avpriv_set_pts_info(st
, 32, 1, i
);
260 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_handler
->init
)
261 rtsp_st
->dynamic_handler
->init(s
, st
->index
,
262 rtsp_st
->dynamic_protocol_context
);
266 /* parse the attribute line from the fmtp a line of an sdp response. This
267 * is broken out as a function because it is used in rtp_h264.c, which is
269 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
270 char *value
, int value_size
)
272 *p
+= strspn(*p
, SPACE_CHARS
);
274 get_word_sep(attr
, attr_size
, "=", p
);
277 get_word_sep(value
, value_size
, ";", p
);
285 typedef struct SDPParseState
{
287 struct sockaddr_storage default_ip
;
289 int skip_media
; ///< set if an unknown m= line occurs
292 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
293 int letter
, const char *buf
)
295 RTSPState
*rt
= s
->priv_data
;
296 char buf1
[64], st_type
[64];
298 enum AVMediaType codec_type
;
302 struct sockaddr_storage sdp_ip
;
305 av_dlog(s
, "sdp: %c='%s'\n", letter
, buf
);
308 if (s1
->skip_media
&& letter
!= 'm')
312 get_word(buf1
, sizeof(buf1
), &p
);
313 if (strcmp(buf1
, "IN") != 0)
315 get_word(buf1
, sizeof(buf1
), &p
);
316 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6"))
318 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
319 if (get_sockaddr(buf1
, &sdp_ip
))
324 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
327 if (s
->nb_streams
== 0) {
328 s1
->default_ip
= sdp_ip
;
329 s1
->default_ttl
= ttl
;
331 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
332 rtsp_st
->sdp_ip
= sdp_ip
;
333 rtsp_st
->sdp_ttl
= ttl
;
337 av_dict_set(&s
->metadata
, "title", p
, 0);
340 if (s
->nb_streams
== 0) {
341 av_dict_set(&s
->metadata
, "comment", p
, 0);
348 codec_type
= AVMEDIA_TYPE_UNKNOWN
;
349 get_word(st_type
, sizeof(st_type
), &p
);
350 if (!strcmp(st_type
, "audio")) {
351 codec_type
= AVMEDIA_TYPE_AUDIO
;
352 } else if (!strcmp(st_type
, "video")) {
353 codec_type
= AVMEDIA_TYPE_VIDEO
;
354 } else if (!strcmp(st_type
, "application")) {
355 codec_type
= AVMEDIA_TYPE_DATA
;
357 if (codec_type
== AVMEDIA_TYPE_UNKNOWN
|| !(rt
->media_type_mask
& (1 << codec_type
))) {
361 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
364 rtsp_st
->stream_index
= -1;
365 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
367 rtsp_st
->sdp_ip
= s1
->default_ip
;
368 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
370 get_word(buf1
, sizeof(buf1
), &p
); /* port */
371 rtsp_st
->sdp_port
= atoi(buf1
);
373 get_word(buf1
, sizeof(buf1
), &p
); /* protocol (ignored) */
375 /* XXX: handle list of formats */
376 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
377 rtsp_st
->sdp_payload_type
= atoi(buf1
);
379 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
380 /* no corresponding stream */
381 } else if (rt
->server_type
== RTSP_SERVER_WMS
&&
382 codec_type
== AVMEDIA_TYPE_DATA
) {
383 /* RTX stream, a stream that carries all the other actual
384 * audio/video streams. Don't expose this to the callers. */
386 st
= avformat_new_stream(s
, NULL
);
389 st
->id
= rt
->nb_rtsp_streams
- 1;
390 rtsp_st
->stream_index
= st
->index
;
391 st
->codec
->codec_type
= codec_type
;
392 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
393 RTPDynamicProtocolHandler
*handler
;
394 /* if standard payload type, we can find the codec right now */
395 ff_rtp_get_codec_info(st
->codec
, rtsp_st
->sdp_payload_type
);
396 if (st
->codec
->codec_type
== AVMEDIA_TYPE_AUDIO
&&
397 st
->codec
->sample_rate
> 0)
398 avpriv_set_pts_info(st
, 32, 1, st
->codec
->sample_rate
);
399 /* Even static payload types may need a custom depacketizer */
400 handler
= ff_rtp_handler_find_by_id(
401 rtsp_st
->sdp_payload_type
, st
->codec
->codec_type
);
402 init_rtp_handler(handler
, rtsp_st
, st
->codec
);
403 if (handler
&& handler
->init
)
404 handler
->init(s
, st
->index
,
405 rtsp_st
->dynamic_protocol_context
);
408 /* put a default control url */
409 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
410 sizeof(rtsp_st
->control_url
));
413 if (av_strstart(p
, "control:", &p
)) {
414 if (s
->nb_streams
== 0) {
415 if (!strncmp(p
, "rtsp://", 7))
416 av_strlcpy(rt
->control_uri
, p
,
417 sizeof(rt
->control_uri
));
420 /* get the control url */
421 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
423 /* XXX: may need to add full url resolution */
424 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
426 if (proto
[0] == '\0') {
427 /* relative control URL */
428 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
429 av_strlcat(rtsp_st
->control_url
, "/",
430 sizeof(rtsp_st
->control_url
));
431 av_strlcat(rtsp_st
->control_url
, p
,
432 sizeof(rtsp_st
->control_url
));
434 av_strlcpy(rtsp_st
->control_url
, p
,
435 sizeof(rtsp_st
->control_url
));
437 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
438 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
439 get_word(buf1
, sizeof(buf1
), &p
);
440 payload_type
= atoi(buf1
);
441 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
442 if (rtsp_st
->stream_index
>= 0) {
443 st
= s
->streams
[rtsp_st
->stream_index
];
444 sdp_parse_rtpmap(s
, st
, rtsp_st
, payload_type
, p
);
446 } else if (av_strstart(p
, "fmtp:", &p
) ||
447 av_strstart(p
, "framesize:", &p
)) {
448 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
449 // let dynamic protocol handlers have a stab at the line.
450 get_word(buf1
, sizeof(buf1
), &p
);
451 payload_type
= atoi(buf1
);
452 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
453 rtsp_st
= rt
->rtsp_streams
[i
];
454 if (rtsp_st
->sdp_payload_type
== payload_type
&&
455 rtsp_st
->dynamic_handler
&&
456 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
457 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, i
,
458 rtsp_st
->dynamic_protocol_context
, buf
);
460 } else if (av_strstart(p
, "range:", &p
)) {
463 // this is so that seeking on a streamed file can work.
464 rtsp_parse_range_npt(p
, &start
, &end
);
465 s
->start_time
= start
;
466 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
467 s
->duration
= (end
== AV_NOPTS_VALUE
) ?
468 AV_NOPTS_VALUE
: end
- start
;
469 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
471 rt
->transport
= RTSP_TRANSPORT_RDT
;
472 } else if (av_strstart(p
, "SampleRate:integer;", &p
) &&
474 st
= s
->streams
[s
->nb_streams
- 1];
475 st
->codec
->sample_rate
= atoi(p
);
477 if (rt
->server_type
== RTSP_SERVER_WMS
)
478 ff_wms_parse_sdp_a_line(s
, p
);
479 if (s
->nb_streams
> 0) {
480 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
482 if (rt
->server_type
== RTSP_SERVER_REAL
)
483 ff_real_parse_sdp_a_line(s
, rtsp_st
->stream_index
, p
);
485 if (rtsp_st
->dynamic_handler
&&
486 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
487 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
488 rtsp_st
->stream_index
,
489 rtsp_st
->dynamic_protocol_context
, buf
);
496 int ff_sdp_parse(AVFormatContext
*s
, const char *content
)
498 RTSPState
*rt
= s
->priv_data
;
501 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
502 * contain long SDP lines containing complete ASF Headers (several
503 * kB) or arrays of MDPR (RM stream descriptor) headers plus
504 * "rulebooks" describing their properties. Therefore, the SDP line
507 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
508 * in rtpdec_xiph.c. */
510 SDPParseState sdp_parse_state
= { { 0 } }, *s1
= &sdp_parse_state
;
514 p
+= strspn(p
, SPACE_CHARS
);
522 /* get the content */
524 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
525 if ((q
- buf
) < sizeof(buf
) - 1)
530 sdp_parse_line(s
, s1
, letter
, buf
);
532 while (*p
!= '\n' && *p
!= '\0')
537 rt
->p
= av_malloc(sizeof(struct pollfd
)*2*(rt
->nb_rtsp_streams
+1));
538 if (!rt
->p
) return AVERROR(ENOMEM
);
541 #endif /* CONFIG_RTPDEC */
543 void ff_rtsp_undo_setup(AVFormatContext
*s
)
545 RTSPState
*rt
= s
->priv_data
;
548 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
549 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
552 if (rtsp_st
->transport_priv
) {
554 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
555 av_write_trailer(rtpctx
);
556 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
558 avio_close_dyn_buf(rtpctx
->pb
, &ptr
);
561 avio_close(rtpctx
->pb
);
563 avformat_free_context(rtpctx
);
564 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
565 ff_rdt_parse_close(rtsp_st
->transport_priv
);
566 else if (CONFIG_RTPDEC
)
567 ff_rtp_parse_close(rtsp_st
->transport_priv
);
569 rtsp_st
->transport_priv
= NULL
;
570 if (rtsp_st
->rtp_handle
)
571 ffurl_close(rtsp_st
->rtp_handle
);
572 rtsp_st
->rtp_handle
= NULL
;
576 /* close and free RTSP streams */
577 void ff_rtsp_close_streams(AVFormatContext
*s
)
579 RTSPState
*rt
= s
->priv_data
;
583 ff_rtsp_undo_setup(s
);
584 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
585 rtsp_st
= rt
->rtsp_streams
[i
];
587 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
)
588 rtsp_st
->dynamic_handler
->free(
589 rtsp_st
->dynamic_protocol_context
);
593 av_free(rt
->rtsp_streams
);
595 avformat_close_input(&rt
->asf_ctx
);
598 av_free(rt
->recvbuf
);
601 int ff_rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
603 RTSPState
*rt
= s
->priv_data
;
606 /* open the RTP context */
607 if (rtsp_st
->stream_index
>= 0)
608 st
= s
->streams
[rtsp_st
->stream_index
];
610 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
612 if (s
->oformat
&& CONFIG_RTSP_MUXER
) {
613 int ret
= ff_rtp_chain_mux_open(&rtsp_st
->transport_priv
, s
, st
,
615 RTSP_TCP_MAX_PACKET_SIZE
);
616 /* Ownership of rtp_handle is passed to the rtp mux context */
617 rtsp_st
->rtp_handle
= NULL
;
620 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
621 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
622 rtsp_st
->dynamic_protocol_context
,
623 rtsp_st
->dynamic_handler
);
624 else if (CONFIG_RTPDEC
)
625 rtsp_st
->transport_priv
= ff_rtp_parse_open(s
, st
, rtsp_st
->rtp_handle
,
626 rtsp_st
->sdp_payload_type
,
627 (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
|| !s
->max_delay
)
628 ?
0 : RTP_REORDER_QUEUE_DEFAULT_SIZE
);
630 if (!rtsp_st
->transport_priv
) {
631 return AVERROR(ENOMEM
);
632 } else if (rt
->transport
!= RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
) {
633 if (rtsp_st
->dynamic_handler
) {
634 ff_rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
635 rtsp_st
->dynamic_protocol_context
,
636 rtsp_st
->dynamic_handler
);
643 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
644 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
651 q
+= strspn(q
, SPACE_CHARS
);
652 v
= strtol(q
, &p
, 10);
656 v
= strtol(p
, &p
, 10);
665 /* XXX: only one transport specification is parsed */
666 static void rtsp_parse_transport(RTSPMessageHeader
*reply
, const char *p
)
668 char transport_protocol
[16];
670 char lower_transport
[16];
672 RTSPTransportField
*th
;
675 reply
->nb_transports
= 0;
678 p
+= strspn(p
, SPACE_CHARS
);
682 th
= &reply
->transports
[reply
->nb_transports
];
684 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
686 if (!av_strcasecmp (transport_protocol
, "rtp")) {
687 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
688 lower_transport
[0] = '\0';
689 /* rtp/avp/<protocol> */
691 get_word_sep(lower_transport
, sizeof(lower_transport
),
694 th
->transport
= RTSP_TRANSPORT_RTP
;
695 } else if (!av_strcasecmp (transport_protocol
, "x-pn-tng") ||
696 !av_strcasecmp (transport_protocol
, "x-real-rdt")) {
697 /* x-pn-tng/<protocol> */
698 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
700 th
->transport
= RTSP_TRANSPORT_RDT
;
702 if (!av_strcasecmp(lower_transport
, "TCP"))
703 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
705 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
709 /* get each parameter */
710 while (*p
!= '\0' && *p
!= ',') {
711 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
712 if (!strcmp(parameter
, "port")) {
715 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
717 } else if (!strcmp(parameter
, "client_port")) {
720 rtsp_parse_range(&th
->client_port_min
,
721 &th
->client_port_max
, &p
);
723 } else if (!strcmp(parameter
, "server_port")) {
726 rtsp_parse_range(&th
->server_port_min
,
727 &th
->server_port_max
, &p
);
729 } else if (!strcmp(parameter
, "interleaved")) {
732 rtsp_parse_range(&th
->interleaved_min
,
733 &th
->interleaved_max
, &p
);
735 } else if (!strcmp(parameter
, "multicast")) {
736 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
737 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
738 } else if (!strcmp(parameter
, "ttl")) {
741 th
->ttl
= strtol(p
, (char **)&p
, 10);
743 } else if (!strcmp(parameter
, "destination")) {
746 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
747 get_sockaddr(buf
, &th
->destination
);
749 } else if (!strcmp(parameter
, "source")) {
752 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
753 av_strlcpy(th
->source
, buf
, sizeof(th
->source
));
755 } else if (!strcmp(parameter
, "mode")) {
758 get_word_sep(buf
, sizeof(buf
), ";, ", &p
);
759 if (!strcmp(buf
, "record") ||
760 !strcmp(buf
, "receive"))
765 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
773 reply
->nb_transports
++;
777 static void handle_rtp_info(RTSPState
*rt
, const char *url
,
778 uint32_t seq
, uint32_t rtptime
)
781 if (!rtptime
|| !url
[0])
783 if (rt
->transport
!= RTSP_TRANSPORT_RTP
)
785 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
786 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
787 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
790 if (!strcmp(rtsp_st
->control_url
, url
)) {
791 rtpctx
->base_timestamp
= rtptime
;
797 static void rtsp_parse_rtp_info(RTSPState
*rt
, const char *p
)
800 char key
[20], value
[1024], url
[1024] = "";
801 uint32_t seq
= 0, rtptime
= 0;
804 p
+= strspn(p
, SPACE_CHARS
);
807 get_word_sep(key
, sizeof(key
), "=", &p
);
811 get_word_sep(value
, sizeof(value
), ";, ", &p
);
813 if (!strcmp(key
, "url"))
814 av_strlcpy(url
, value
, sizeof(url
));
815 else if (!strcmp(key
, "seq"))
816 seq
= strtoul(value
, NULL
, 10);
817 else if (!strcmp(key
, "rtptime"))
818 rtptime
= strtoul(value
, NULL
, 10);
820 handle_rtp_info(rt
, url
, seq
, rtptime
);
829 handle_rtp_info(rt
, url
, seq
, rtptime
);
832 void ff_rtsp_parse_line(RTSPMessageHeader
*reply
, const char *buf
,
833 RTSPState
*rt
, const char *method
)
837 /* NOTE: we do case independent match for broken servers */
839 if (av_stristart(p
, "Session:", &p
)) {
841 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
842 if (av_stristart(p
, ";timeout=", &p
) &&
843 (t
= strtol(p
, NULL
, 10)) > 0) {
846 } else if (av_stristart(p
, "Content-Length:", &p
)) {
847 reply
->content_length
= strtol(p
, NULL
, 10);
848 } else if (av_stristart(p
, "Transport:", &p
)) {
849 rtsp_parse_transport(reply
, p
);
850 } else if (av_stristart(p
, "CSeq:", &p
)) {
851 reply
->seq
= strtol(p
, NULL
, 10);
852 } else if (av_stristart(p
, "Range:", &p
)) {
853 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
854 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
855 p
+= strspn(p
, SPACE_CHARS
);
856 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
857 } else if (av_stristart(p
, "Server:", &p
)) {
858 p
+= strspn(p
, SPACE_CHARS
);
859 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
860 } else if (av_stristart(p
, "Notice:", &p
) ||
861 av_stristart(p
, "X-Notice:", &p
)) {
862 reply
->notice
= strtol(p
, NULL
, 10);
863 } else if (av_stristart(p
, "Location:", &p
)) {
864 p
+= strspn(p
, SPACE_CHARS
);
865 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
866 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && rt
) {
867 p
+= strspn(p
, SPACE_CHARS
);
868 ff_http_auth_handle_header(&rt
->auth_state
, "WWW-Authenticate", p
);
869 } else if (av_stristart(p
, "Authentication-Info:", &p
) && rt
) {
870 p
+= strspn(p
, SPACE_CHARS
);
871 ff_http_auth_handle_header(&rt
->auth_state
, "Authentication-Info", p
);
872 } else if (av_stristart(p
, "Content-Base:", &p
) && rt
) {
873 p
+= strspn(p
, SPACE_CHARS
);
874 if (method
&& !strcmp(method
, "DESCRIBE"))
875 av_strlcpy(rt
->control_uri
, p
, sizeof(rt
->control_uri
));
876 } else if (av_stristart(p
, "RTP-Info:", &p
) && rt
) {
877 p
+= strspn(p
, SPACE_CHARS
);
878 if (method
&& !strcmp(method
, "PLAY"))
879 rtsp_parse_rtp_info(rt
, p
);
880 } else if (av_stristart(p
, "Public:", &p
) && rt
) {
881 if (strstr(p
, "GET_PARAMETER") &&
882 method
&& !strcmp(method
, "OPTIONS"))
883 rt
->get_parameter_supported
= 1;
884 } else if (av_stristart(p
, "x-Accept-Dynamic-Rate:", &p
) && rt
) {
885 p
+= strspn(p
, SPACE_CHARS
);
886 rt
->accept_dynamic_rate
= atoi(p
);
887 } else if (av_stristart(p
, "Content-Type:", &p
)) {
888 p
+= strspn(p
, SPACE_CHARS
);
889 av_strlcpy(reply
->content_type
, p
, sizeof(reply
->content_type
));
893 /* skip a RTP/TCP interleaved packet */
894 void ff_rtsp_skip_packet(AVFormatContext
*s
)
896 RTSPState
*rt
= s
->priv_data
;
900 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, 3);
903 len
= AV_RB16(buf
+ 1);
905 av_dlog(s
, "skipping RTP packet len=%d\n", len
);
910 if (len1
> sizeof(buf
))
912 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, len1
);
919 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
920 unsigned char **content_ptr
,
921 int return_on_interleaved_data
, const char *method
)
923 RTSPState
*rt
= s
->priv_data
;
924 char buf
[4096], buf1
[1024], *q
;
927 int ret
, content_length
, line_count
= 0, request
= 0;
928 unsigned char *content
= NULL
;
934 memset(reply
, 0, sizeof(*reply
));
936 /* parse reply (XXX: use buffers) */
937 rt
->last_reply
[0] = '\0';
941 ret
= ffurl_read_complete(rt
->rtsp_hd
, &ch
, 1);
942 av_dlog(s
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
948 /* XXX: only parse it if first char on line ? */
949 if (return_on_interleaved_data
) {
952 ff_rtsp_skip_packet(s
);
953 } else if (ch
!= '\r') {
954 if ((q
- buf
) < sizeof(buf
) - 1)
960 av_dlog(s
, "line='%s'\n", buf
);
962 /* test if last line */
966 if (line_count
== 0) {
968 get_word(buf1
, sizeof(buf1
), &p
);
969 if (!strncmp(buf1
, "RTSP/", 5)) {
970 get_word(buf1
, sizeof(buf1
), &p
);
971 reply
->status_code
= atoi(buf1
);
972 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
974 av_strlcpy(reply
->reason
, buf1
, sizeof(reply
->reason
)); // method
975 get_word(buf1
, sizeof(buf1
), &p
); // object
979 ff_rtsp_parse_line(reply
, p
, rt
, method
);
980 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
981 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
986 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0' && !request
)
987 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
989 content_length
= reply
->content_length
;
990 if (content_length
> 0) {
991 /* leave some room for a trailing '\0' (useful for simple parsing) */
992 content
= av_malloc(content_length
+ 1);
993 ffurl_read_complete(rt
->rtsp_hd
, content
, content_length
);
994 content
[content_length
] = '\0';
997 *content_ptr
= content
;
1003 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1004 const char* ptr
= buf
;
1006 if (!strcmp(reply
->reason
, "OPTIONS")) {
1007 snprintf(buf
, sizeof(buf
), "RTSP/1.0 200 OK\r\n");
1009 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", reply
->seq
);
1010 if (reply
->session_id
[0])
1011 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n",
1014 snprintf(buf
, sizeof(buf
), "RTSP/1.0 501 Not Implemented\r\n");
1016 av_strlcat(buf
, "\r\n", sizeof(buf
));
1018 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1019 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1022 ffurl_write(rt
->rtsp_hd_out
, ptr
, strlen(ptr
));
1024 rt
->last_cmd_time
= av_gettime();
1025 /* Even if the request from the server had data, it is not the data
1026 * that the caller wants or expects. The memory could also be leaked
1027 * if the actual following reply has content data. */
1029 av_freep(content_ptr
);
1030 /* If method is set, this is called from ff_rtsp_send_cmd,
1031 * where a reply to exactly this request is awaited. For
1032 * callers from within packet receiving, we just want to
1033 * return to the caller and go back to receiving packets. */
1039 if (rt
->seq
!= reply
->seq
) {
1040 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
1041 rt
->seq
, reply
->seq
);
1045 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
1046 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
1047 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
1048 rt
->state
= RTSP_STATE_IDLE
;
1049 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
1050 return AVERROR(EIO
); /* data or server error */
1051 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
1052 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
1053 return AVERROR(EPERM
);
1059 * Send a command to the RTSP server without waiting for the reply.
1061 * @param s RTSP (de)muxer context
1062 * @param method the method for the request
1063 * @param url the target url for the request
1064 * @param headers extra header lines to include in the request
1065 * @param send_content if non-null, the data to send as request body content
1066 * @param send_content_length the length of the send_content data, or 0 if
1067 * send_content is null
1069 * @return zero if success, nonzero otherwise
1071 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
1072 const char *method
, const char *url
,
1073 const char *headers
,
1074 const unsigned char *send_content
,
1075 int send_content_length
)
1077 RTSPState
*rt
= s
->priv_data
;
1078 char buf
[4096], *out_buf
;
1079 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1081 /* Add in RTSP headers */
1084 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
1086 av_strlcat(buf
, headers
, sizeof(buf
));
1087 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
1088 if (rt
->session_id
[0] != '\0' && (!headers
||
1089 !strstr(headers
, "\nIf-Match:"))) {
1090 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
1093 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
1094 rt
->auth
, url
, method
);
1096 av_strlcat(buf
, str
, sizeof(buf
));
1099 if (send_content_length
> 0 && send_content
)
1100 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
1101 av_strlcat(buf
, "\r\n", sizeof(buf
));
1103 /* base64 encode rtsp if tunneling */
1104 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1105 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1106 out_buf
= base64buf
;
1109 av_dlog(s
, "Sending:\n%s--\n", buf
);
1111 ffurl_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
1112 if (send_content_length
> 0 && send_content
) {
1113 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1114 av_log(s
, AV_LOG_ERROR
, "tunneling of RTSP requests "
1115 "with content data not supported\n");
1116 return AVERROR_PATCHWELCOME
;
1118 ffurl_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
1120 rt
->last_cmd_time
= av_gettime();
1125 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
1126 const char *url
, const char *headers
)
1128 return ff_rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
1131 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
1132 const char *headers
, RTSPMessageHeader
*reply
,
1133 unsigned char **content_ptr
)
1135 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
1136 content_ptr
, NULL
, 0);
1139 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
1140 const char *method
, const char *url
,
1142 RTSPMessageHeader
*reply
,
1143 unsigned char **content_ptr
,
1144 const unsigned char *send_content
,
1145 int send_content_length
)
1147 RTSPState
*rt
= s
->priv_data
;
1148 HTTPAuthType cur_auth_type
;
1149 int ret
, attempts
= 0;
1152 cur_auth_type
= rt
->auth_state
.auth_type
;
1153 if ((ret
= ff_rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
1155 send_content_length
)))
1158 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0, method
) ) < 0)
1162 if (reply
->status_code
== 401 &&
1163 (cur_auth_type
== HTTP_AUTH_NONE
|| rt
->auth_state
.stale
) &&
1164 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
&& attempts
< 2)
1167 if (reply
->status_code
> 400){
1168 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
1172 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
1178 int ff_rtsp_make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
1179 int lower_transport
, const char *real_challenge
)
1181 RTSPState
*rt
= s
->priv_data
;
1182 int rtx
= 0, j
, i
, err
, interleave
= 0, port_off
;
1183 RTSPStream
*rtsp_st
;
1184 RTSPMessageHeader reply1
, *reply
= &reply1
;
1186 const char *trans_pref
;
1188 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
1189 trans_pref
= "x-pn-tng";
1191 trans_pref
= "RTP/AVP";
1193 /* default timeout: 1 minute */
1196 /* for each stream, make the setup request */
1197 /* XXX: we assume the same server is used for the control of each
1200 /* Choose a random starting offset within the first half of the
1201 * port range, to allow for a number of ports to try even if the offset
1202 * happens to be at the end of the random range. */
1203 port_off
= av_get_random_seed() % ((rt
->rtp_port_max
- rt
->rtp_port_min
)/2);
1204 /* even random offset */
1205 port_off
-= port_off
& 0x01;
1207 for (j
= rt
->rtp_port_min
+ port_off
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1208 char transport
[2048];
1211 * WMS serves all UDP data over a single connection, the RTX, which
1212 * isn't necessarily the first in the SDP but has to be the first
1213 * to be set up, else the second/third SETUP will fail with a 461.
1215 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1216 rt
->server_type
== RTSP_SERVER_WMS
) {
1219 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1220 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1222 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1226 if (rtx
== rt
->nb_rtsp_streams
)
1227 return -1; /* no RTX found */
1228 rtsp_st
= rt
->rtsp_streams
[rtx
];
1230 rtsp_st
= rt
->rtsp_streams
[i
> rtx ? i
: i
- 1];
1232 rtsp_st
= rt
->rtsp_streams
[i
];
1235 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1238 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1239 port
= reply
->transports
[0].client_port_min
;
1243 /* first try in specified port range */
1244 while (j
<= rt
->rtp_port_max
) {
1245 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1246 "?localport=%d", j
);
1247 /* we will use two ports per rtp stream (rtp and rtcp) */
1249 if (!ffurl_open(&rtsp_st
->rtp_handle
, buf
, AVIO_FLAG_READ_WRITE
,
1250 &s
->interrupt_callback
, NULL
))
1254 av_log(s
, AV_LOG_ERROR
, "Unable to open an input RTP port\n");
1259 port
= ff_rtp_get_local_rtp_port(rtsp_st
->rtp_handle
);
1261 snprintf(transport
, sizeof(transport
) - 1,
1262 "%s/UDP;", trans_pref
);
1263 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1264 av_strlcat(transport
, "unicast;", sizeof(transport
));
1265 av_strlcatf(transport
, sizeof(transport
),
1266 "client_port=%d", port
);
1267 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1268 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1269 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1273 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1274 /* For WMS streams, the application streams are only used for
1275 * UDP. When trying to set it up for TCP streams, the server
1276 * will return an error. Therefore, we skip those streams. */
1277 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1278 (rtsp_st
->stream_index
< 0 ||
1279 s
->streams
[rtsp_st
->stream_index
]->codec
->codec_type
==
1282 snprintf(transport
, sizeof(transport
) - 1,
1283 "%s/TCP;", trans_pref
);
1284 if (rt
->transport
!= RTSP_TRANSPORT_RDT
)
1285 av_strlcat(transport
, "unicast;", sizeof(transport
));
1286 av_strlcatf(transport
, sizeof(transport
),
1287 "interleaved=%d-%d",
1288 interleave
, interleave
+ 1);
1292 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1293 snprintf(transport
, sizeof(transport
) - 1,
1294 "%s/UDP;multicast", trans_pref
);
1297 av_strlcat(transport
, ";mode=record", sizeof(transport
));
1298 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1299 rt
->server_type
== RTSP_SERVER_WMS
)
1300 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1301 snprintf(cmd
, sizeof(cmd
),
1302 "Transport: %s\r\n",
1304 if (rt
->accept_dynamic_rate
)
1305 av_strlcat(cmd
, "x-Dynamic-Rate: 0\r\n", sizeof(cmd
));
1306 if (i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
&& CONFIG_RTPDEC
) {
1307 char real_res
[41], real_csum
[9];
1308 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1310 av_strlcatf(cmd
, sizeof(cmd
),
1312 "RealChallenge2: %s, sd=%s\r\n",
1313 rt
->session_id
, real_res
, real_csum
);
1315 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1316 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1319 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1320 reply
->nb_transports
!= 1) {
1321 err
= AVERROR_INVALIDDATA
;
1325 /* XXX: same protocol for all streams is required */
1327 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1328 reply
->transports
[0].transport
!= rt
->transport
) {
1329 err
= AVERROR_INVALIDDATA
;
1333 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1334 rt
->transport
= reply
->transports
[0].transport
;
1337 /* Fail if the server responded with another lower transport mode
1338 * than what we requested. */
1339 if (reply
->transports
[0].lower_transport
!= lower_transport
) {
1340 av_log(s
, AV_LOG_ERROR
, "Nonmatching transport in server reply\n");
1341 err
= AVERROR_INVALIDDATA
;
1345 switch(reply
->transports
[0].lower_transport
) {
1346 case RTSP_LOWER_TRANSPORT_TCP
:
1347 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1348 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1351 case RTSP_LOWER_TRANSPORT_UDP
: {
1352 char url
[1024], options
[30] = "";
1354 if (rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
)
1355 av_strlcpy(options
, "?connect=1", sizeof(options
));
1356 /* Use source address if specified */
1357 if (reply
->transports
[0].source
[0]) {
1358 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1359 reply
->transports
[0].source
,
1360 reply
->transports
[0].server_port_min
, "%s", options
);
1362 ff_url_join(url
, sizeof(url
), "rtp", NULL
, host
,
1363 reply
->transports
[0].server_port_min
, "%s", options
);
1365 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1366 ff_rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1367 err
= AVERROR_INVALIDDATA
;
1370 /* Try to initialize the connection state in a
1371 * potential NAT router by sending dummy packets.
1372 * RTP/RTCP dummy packets are used for RDT, too.
1374 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) && s
->iformat
&&
1376 ff_rtp_send_punch_packets(rtsp_st
->rtp_handle
);
1379 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1380 char url
[1024], namebuf
[50], optbuf
[20] = "";
1381 struct sockaddr_storage addr
;
1384 if (reply
->transports
[0].destination
.ss_family
) {
1385 addr
= reply
->transports
[0].destination
;
1386 port
= reply
->transports
[0].port_min
;
1387 ttl
= reply
->transports
[0].ttl
;
1389 addr
= rtsp_st
->sdp_ip
;
1390 port
= rtsp_st
->sdp_port
;
1391 ttl
= rtsp_st
->sdp_ttl
;
1394 snprintf(optbuf
, sizeof(optbuf
), "?ttl=%d", ttl
);
1395 getnameinfo((struct sockaddr
*) &addr
, sizeof(addr
),
1396 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1397 ff_url_join(url
, sizeof(url
), "rtp", NULL
, namebuf
,
1398 port
, "%s", optbuf
);
1399 if (ffurl_open(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
,
1400 &s
->interrupt_callback
, NULL
) < 0) {
1401 err
= AVERROR_INVALIDDATA
;
1408 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
1412 if (rt
->nb_rtsp_streams
&& reply
->timeout
> 0)
1413 rt
->timeout
= reply
->timeout
;
1415 if (rt
->server_type
== RTSP_SERVER_REAL
)
1416 rt
->need_subscription
= 1;
1421 ff_rtsp_undo_setup(s
);
1425 void ff_rtsp_close_connections(AVFormatContext
*s
)
1427 RTSPState
*rt
= s
->priv_data
;
1428 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
) ffurl_close(rt
->rtsp_hd_out
);
1429 ffurl_close(rt
->rtsp_hd
);
1430 rt
->rtsp_hd
= rt
->rtsp_hd_out
= NULL
;
1433 int ff_rtsp_connect(AVFormatContext
*s
)
1435 RTSPState
*rt
= s
->priv_data
;
1436 char host
[1024], path
[1024], tcpname
[1024], cmd
[2048], auth
[128];
1437 int port
, err
, tcp_fd
;
1438 RTSPMessageHeader reply1
= {0}, *reply
= &reply1
;
1439 int lower_transport_mask
= 0;
1440 char real_challenge
[64] = "";
1441 struct sockaddr_storage peer
;
1442 socklen_t peer_len
= sizeof(peer
);
1444 if (rt
->rtp_port_max
< rt
->rtp_port_min
) {
1445 av_log(s
, AV_LOG_ERROR
, "Invalid UDP port range, max port %d less "
1446 "than min port %d\n", rt
->rtp_port_max
,
1448 return AVERROR(EINVAL
);
1451 if (!ff_network_init())
1452 return AVERROR(EIO
);
1454 if (s
->max_delay
< 0) /* Not set by the caller */
1455 s
->max_delay
= s
->iformat ? DEFAULT_REORDERING_DELAY
: 0;
1457 rt
->control_transport
= RTSP_MODE_PLAIN
;
1458 if (rt
->lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_HTTP
)) {
1459 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1460 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1462 /* Only pass through valid flags from here */
1463 rt
->lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1466 lower_transport_mask
= rt
->lower_transport_mask
;
1467 /* extract hostname and port */
1468 av_url_split(NULL
, 0, auth
, sizeof(auth
),
1469 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->filename
);
1471 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1474 port
= RTSP_DEFAULT_PORT
;
1476 if (!lower_transport_mask
)
1477 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1480 /* Only UDP or TCP - UDP multicast isn't supported. */
1481 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1482 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1483 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1484 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1485 "only UDP and TCP are supported for output.\n");
1486 err
= AVERROR(EINVAL
);
1491 /* Construct the URI used in request; this is similar to s->filename,
1492 * but with authentication credentials removed and RTSP specific options
1494 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), "rtsp", NULL
,
1495 host
, port
, "%s", path
);
1497 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1498 /* set up initial handshake for tunneling */
1499 char httpname
[1024];
1500 char sessioncookie
[17];
1503 ff_url_join(httpname
, sizeof(httpname
), "http", auth
, host
, port
, "%s", path
);
1504 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1505 av_get_random_seed(), av_get_random_seed());
1508 if (ffurl_alloc(&rt
->rtsp_hd
, httpname
, AVIO_FLAG_READ
,
1509 &s
->interrupt_callback
) < 0) {
1514 /* generate GET headers */
1515 snprintf(headers
, sizeof(headers
),
1516 "x-sessioncookie: %s\r\n"
1517 "Accept: application/x-rtsp-tunnelled\r\n"
1518 "Pragma: no-cache\r\n"
1519 "Cache-Control: no-cache\r\n",
1521 av_opt_set(rt
->rtsp_hd
->priv_data
, "headers", headers
, 0);
1523 /* complete the connection */
1524 if (ffurl_connect(rt
->rtsp_hd
, NULL
)) {
1530 if (ffurl_alloc(&rt
->rtsp_hd_out
, httpname
, AVIO_FLAG_WRITE
,
1531 &s
->interrupt_callback
) < 0 ) {
1536 /* generate POST headers */
1537 snprintf(headers
, sizeof(headers
),
1538 "x-sessioncookie: %s\r\n"
1539 "Content-Type: application/x-rtsp-tunnelled\r\n"
1540 "Pragma: no-cache\r\n"
1541 "Cache-Control: no-cache\r\n"
1542 "Content-Length: 32767\r\n"
1543 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1545 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "headers", headers
, 0);
1546 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "chunked_post", "0", 0);
1548 /* Initialize the authentication state for the POST session. The HTTP
1549 * protocol implementation doesn't properly handle multi-pass
1550 * authentication for POST requests, since it would require one of
1552 * - implementing Expect: 100-continue, which many HTTP servers
1553 * don't support anyway, even less the RTSP servers that do HTTP
1555 * - sending the whole POST data until getting a 401 reply specifying
1556 * what authentication method to use, then resending all that data
1557 * - waiting for potential 401 replies directly after sending the
1558 * POST header (waiting for some unspecified time)
1559 * Therefore, we copy the full auth state, which works for both basic
1560 * and digest. (For digest, we would have to synchronize the nonce
1561 * count variable between the two sessions, if we'd do more requests
1562 * with the original session, though.)
1564 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1566 /* complete the connection */
1567 if (ffurl_connect(rt
->rtsp_hd_out
, NULL
)) {
1572 /* open the tcp connection */
1573 ff_url_join(tcpname
, sizeof(tcpname
), "tcp", NULL
, host
, port
, NULL
);
1574 if (ffurl_open(&rt
->rtsp_hd
, tcpname
, AVIO_FLAG_READ_WRITE
,
1575 &s
->interrupt_callback
, NULL
) < 0) {
1579 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1583 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1584 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1585 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1586 NULL
, 0, NI_NUMERICHOST
);
1589 /* request options supported by the server; this also detects server
1591 for (rt
->server_type
= RTSP_SERVER_RTP
;;) {
1593 if (rt
->server_type
== RTSP_SERVER_REAL
)
1596 * The following entries are required for proper
1597 * streaming from a Realmedia server. They are
1598 * interdependent in some way although we currently
1599 * don't quite understand how. Values were copied
1600 * from mplayer SVN r23589.
1601 * ClientChallenge is a 16-byte ID in hex
1602 * CompanyID is a 16-byte ID in base64
1604 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1605 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1606 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1607 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1609 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
1610 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1611 err
= AVERROR_INVALIDDATA
;
1615 /* detect server type if not standard-compliant RTP */
1616 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
1617 rt
->server_type
= RTSP_SERVER_REAL
;
1619 } else if (!av_strncasecmp(reply
->server
, "WMServer/", 9)) {
1620 rt
->server_type
= RTSP_SERVER_WMS
;
1621 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
1622 strcpy(real_challenge
, reply
->real_challenge
);
1626 if (s
->iformat
&& CONFIG_RTSP_DEMUXER
)
1627 err
= ff_rtsp_setup_input_streams(s
, reply
);
1628 else if (CONFIG_RTSP_MUXER
)
1629 err
= ff_rtsp_setup_output_streams(s
, host
);
1634 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
1635 ~(lower_transport_mask
- 1)];
1637 err
= ff_rtsp_make_setup_request(s
, host
, port
, lower_transport
,
1638 rt
->server_type
== RTSP_SERVER_REAL ?
1639 real_challenge
: NULL
);
1642 lower_transport_mask
&= ~(1 << lower_transport
);
1643 if (lower_transport_mask
== 0 && err
== 1) {
1644 err
= AVERROR(EPROTONOSUPPORT
);
1649 rt
->lower_transport_mask
= lower_transport_mask
;
1650 av_strlcpy(rt
->real_challenge
, real_challenge
, sizeof(rt
->real_challenge
));
1651 rt
->state
= RTSP_STATE_IDLE
;
1652 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
1655 ff_rtsp_close_streams(s
);
1656 ff_rtsp_close_connections(s
);
1657 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
1658 av_strlcpy(s
->filename
, reply
->location
, sizeof(s
->filename
));
1659 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
1667 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1670 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1671 uint8_t *buf
, int buf_size
, int64_t wait_end
)
1673 RTSPState
*rt
= s
->priv_data
;
1674 RTSPStream
*rtsp_st
;
1675 int n
, i
, ret
, tcp_fd
, timeout_cnt
= 0;
1677 struct pollfd
*p
= rt
->p
;
1680 if (ff_check_interrupt(&s
->interrupt_callback
))
1681 return AVERROR_EXIT
;
1682 if (wait_end
&& wait_end
- av_gettime() < 0)
1683 return AVERROR(EAGAIN
);
1686 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1687 p
[max_p
].fd
= tcp_fd
;
1688 p
[max_p
++].events
= POLLIN
;
1692 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1693 rtsp_st
= rt
->rtsp_streams
[i
];
1694 if (rtsp_st
->rtp_handle
) {
1695 p
[max_p
].fd
= ffurl_get_file_handle(rtsp_st
->rtp_handle
);
1696 p
[max_p
++].events
= POLLIN
;
1697 p
[max_p
].fd
= ff_rtp_get_rtcp_file_handle(rtsp_st
->rtp_handle
);
1698 p
[max_p
++].events
= POLLIN
;
1701 n
= poll(p
, max_p
, POLL_TIMEOUT_MS
);
1703 int j
= 1 - (tcp_fd
== -1);
1705 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1706 rtsp_st
= rt
->rtsp_streams
[i
];
1707 if (rtsp_st
->rtp_handle
) {
1708 if (p
[j
].revents
& POLLIN
|| p
[j
+1].revents
& POLLIN
) {
1709 ret
= ffurl_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
1711 *prtsp_st
= rtsp_st
;
1718 #if CONFIG_RTSP_DEMUXER
1719 if (tcp_fd
!= -1 && p
[0].revents
& POLLIN
) {
1720 if (rt
->rtsp_flags
& RTSP_FLAG_LISTEN
) {
1721 if (rt
->state
== RTSP_STATE_STREAMING
) {
1722 if (!ff_rtsp_parse_streaming_commands(s
))
1725 av_log(s
, AV_LOG_WARNING
,
1726 "Unable to answer to TEARDOWN\n");
1730 RTSPMessageHeader reply
;
1731 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0, NULL
);
1734 /* XXX: parse message */
1735 if (rt
->state
!= RTSP_STATE_STREAMING
)
1740 } else if (n
== 0 && ++timeout_cnt
>= MAX_TIMEOUTS
) {
1741 return AVERROR(ETIMEDOUT
);
1742 } else if (n
< 0 && errno
!= EINTR
)
1743 return AVERROR(errno
);
1747 int ff_rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
1749 RTSPState
*rt
= s
->priv_data
;
1751 RTSPStream
*rtsp_st
, *first_queue_st
= NULL
;
1752 int64_t wait_end
= 0;
1754 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1757 /* get next frames from the same RTP packet */
1758 if (rt
->cur_transport_priv
) {
1759 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1760 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1762 ret
= ff_rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1764 rt
->cur_transport_priv
= NULL
;
1766 } else if (ret
== 1) {
1769 rt
->cur_transport_priv
= NULL
;
1772 if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
1774 int64_t first_queue_time
= 0;
1775 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1776 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
1780 queue_time
= ff_rtp_queued_packet_time(rtpctx
);
1781 if (queue_time
&& (queue_time
- first_queue_time
< 0 ||
1782 !first_queue_time
)) {
1783 first_queue_time
= queue_time
;
1784 first_queue_st
= rt
->rtsp_streams
[i
];
1787 if (first_queue_time
)
1788 wait_end
= first_queue_time
+ s
->max_delay
;
1791 /* read next RTP packet */
1794 rt
->recvbuf
= av_malloc(RECVBUF_SIZE
);
1796 return AVERROR(ENOMEM
);
1799 switch(rt
->lower_transport
) {
1801 #if CONFIG_RTSP_DEMUXER
1802 case RTSP_LOWER_TRANSPORT_TCP
:
1803 len
= ff_rtsp_tcp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
);
1806 case RTSP_LOWER_TRANSPORT_UDP
:
1807 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
1808 len
= udp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
, wait_end
);
1809 if (len
> 0 && rtsp_st
->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
1810 ff_rtp_check_and_send_back_rr(rtsp_st
->transport_priv
, len
);
1813 if (len
== AVERROR(EAGAIN
) && first_queue_st
&&
1814 rt
->transport
== RTSP_TRANSPORT_RTP
) {
1815 rtsp_st
= first_queue_st
;
1816 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, NULL
, 0);
1823 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1824 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1826 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1828 /* Either bad packet, or a RTCP packet. Check if the
1829 * first_rtcp_ntp_time field was initialized. */
1830 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1831 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
1832 /* first_rtcp_ntp_time has been initialized for this stream,
1833 * copy the same value to all other uninitialized streams,
1834 * in order to map their timestamp origin to the same ntp time
1837 AVStream
*st
= NULL
;
1838 if (rtsp_st
->stream_index
>= 0)
1839 st
= s
->streams
[rtsp_st
->stream_index
];
1840 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1841 RTPDemuxContext
*rtpctx2
= rt
->rtsp_streams
[i
]->transport_priv
;
1842 AVStream
*st2
= NULL
;
1843 if (rt
->rtsp_streams
[i
]->stream_index
>= 0)
1844 st2
= s
->streams
[rt
->rtsp_streams
[i
]->stream_index
];
1845 if (rtpctx2
&& st
&& st2
&&
1846 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) {
1847 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
1848 rtpctx2
->rtcp_ts_offset
= av_rescale_q(
1849 rtpctx
->rtcp_ts_offset
, st
->time_base
,
1854 if (ret
== -RTCP_BYE
) {
1857 av_log(s
, AV_LOG_DEBUG
, "Received BYE for stream %d (%d/%d)\n",
1858 rtsp_st
->stream_index
, rt
->nb_byes
, rt
->nb_rtsp_streams
);
1860 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1869 /* more packets may follow, so we save the RTP context */
1870 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
1874 #endif /* CONFIG_RTPDEC */
1876 #if CONFIG_SDP_DEMUXER
1877 static int sdp_probe(AVProbeData
*p1
)
1879 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
1881 /* we look for a line beginning "c=IN IP" */
1882 while (p
< p_end
&& *p
!= '\0') {
1883 if (p
+ sizeof("c=IN IP") - 1 < p_end
&&
1884 av_strstart(p
, "c=IN IP", NULL
))
1885 return AVPROBE_SCORE_MAX
/ 2;
1887 while (p
< p_end
- 1 && *p
!= '\n') p
++;
1896 static int sdp_read_header(AVFormatContext
*s
)
1898 RTSPState
*rt
= s
->priv_data
;
1899 RTSPStream
*rtsp_st
;
1904 if (!ff_network_init())
1905 return AVERROR(EIO
);
1907 if (s
->max_delay
< 0) /* Not set by the caller */
1908 s
->max_delay
= DEFAULT_REORDERING_DELAY
;
1910 /* read the whole sdp file */
1911 /* XXX: better loading */
1912 content
= av_malloc(SDP_MAX_SIZE
);
1913 size
= avio_read(s
->pb
, content
, SDP_MAX_SIZE
- 1);
1916 return AVERROR_INVALIDDATA
;
1918 content
[size
] ='\0';
1920 err
= ff_sdp_parse(s
, content
);
1924 /* open each RTP stream */
1925 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1927 rtsp_st
= rt
->rtsp_streams
[i
];
1929 getnameinfo((struct sockaddr
*) &rtsp_st
->sdp_ip
, sizeof(rtsp_st
->sdp_ip
),
1930 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1931 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1932 namebuf
, rtsp_st
->sdp_port
,
1933 "?localport=%d&ttl=%d&connect=%d", rtsp_st
->sdp_port
,
1935 rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC ?
1 : 0);
1936 if (ffurl_open(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
,
1937 &s
->interrupt_callback
, NULL
) < 0) {
1938 err
= AVERROR_INVALIDDATA
;
1941 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
1946 ff_rtsp_close_streams(s
);
1951 static int sdp_read_close(AVFormatContext
*s
)
1953 ff_rtsp_close_streams(s
);
1958 static const AVClass sdp_demuxer_class
= {
1959 .class_name
= "SDP demuxer",
1960 .item_name
= av_default_item_name
,
1961 .option
= sdp_options
,
1962 .version
= LIBAVUTIL_VERSION_INT
,
1965 AVInputFormat ff_sdp_demuxer
= {
1967 .long_name
= NULL_IF_CONFIG_SMALL("SDP"),
1968 .priv_data_size
= sizeof(RTSPState
),
1969 .read_probe
= sdp_probe
,
1970 .read_header
= sdp_read_header
,
1971 .read_packet
= ff_rtsp_fetch_packet
,
1972 .read_close
= sdp_read_close
,
1973 .priv_class
= &sdp_demuxer_class
,
1975 #endif /* CONFIG_SDP_DEMUXER */
1977 #if CONFIG_RTP_DEMUXER
1978 static int rtp_probe(AVProbeData
*p
)
1980 if (av_strstart(p
->filename
, "rtp:", NULL
))
1981 return AVPROBE_SCORE_MAX
;
1985 static int rtp_read_header(AVFormatContext
*s
)
1987 uint8_t recvbuf
[1500];
1988 char host
[500], sdp
[500];
1990 URLContext
* in
= NULL
;
1992 AVCodecContext codec
= { 0 };
1993 struct sockaddr_storage addr
;
1995 socklen_t addrlen
= sizeof(addr
);
1996 RTSPState
*rt
= s
->priv_data
;
1998 if (!ff_network_init())
1999 return AVERROR(EIO
);
2001 ret
= ffurl_open(&in
, s
->filename
, AVIO_FLAG_READ
,
2002 &s
->interrupt_callback
, NULL
);
2007 ret
= ffurl_read(in
, recvbuf
, sizeof(recvbuf
));
2008 if (ret
== AVERROR(EAGAIN
))
2013 av_log(s
, AV_LOG_WARNING
, "Received too short packet\n");
2017 if ((recvbuf
[0] & 0xc0) != 0x80) {
2018 av_log(s
, AV_LOG_WARNING
, "Unsupported RTP version packet "
2023 if (RTP_PT_IS_RTCP(recvbuf
[1]))
2026 payload_type
= recvbuf
[1] & 0x7f;
2029 getsockname(ffurl_get_file_handle(in
), (struct sockaddr
*) &addr
, &addrlen
);
2033 if (ff_rtp_get_codec_info(&codec
, payload_type
)) {
2034 av_log(s
, AV_LOG_ERROR
, "Unable to receive RTP payload type %d "
2035 "without an SDP file describing it\n",
2039 if (codec
.codec_type
!= AVMEDIA_TYPE_DATA
) {
2040 av_log(s
, AV_LOG_WARNING
, "Guessing on RTP content - if not received "
2041 "properly you need an SDP file "
2045 av_url_split(NULL
, 0, NULL
, 0, host
, sizeof(host
), &port
,
2046 NULL
, 0, s
->filename
);
2048 snprintf(sdp
, sizeof(sdp
),
2049 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2050 addr
.ss_family
== AF_INET ?
4 : 6, host
,
2051 codec
.codec_type
== AVMEDIA_TYPE_DATA ?
"application" :
2052 codec
.codec_type
== AVMEDIA_TYPE_VIDEO ?
"video" : "audio",
2053 port
, payload_type
);
2054 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
);
2056 ffio_init_context(&pb
, sdp
, strlen(sdp
), 0, NULL
, NULL
, NULL
, NULL
);
2059 /* sdp_read_header initializes this again */
2062 rt
->media_type_mask
= (1 << (AVMEDIA_TYPE_DATA
+1)) - 1;
2064 ret
= sdp_read_header(s
);
2075 static const AVClass rtp_demuxer_class
= {
2076 .class_name
= "RTP demuxer",
2077 .item_name
= av_default_item_name
,
2078 .option
= rtp_options
,
2079 .version
= LIBAVUTIL_VERSION_INT
,
2082 AVInputFormat ff_rtp_demuxer
= {
2084 .long_name
= NULL_IF_CONFIG_SMALL("RTP input"),
2085 .priv_data_size
= sizeof(RTSPState
),
2086 .read_probe
= rtp_probe
,
2087 .read_header
= rtp_read_header
,
2088 .read_packet
= ff_rtsp_fetch_packet
,
2089 .read_close
= sdp_read_close
,
2090 .flags
= AVFMT_NOFILE
,
2091 .priv_class
= &rtp_demuxer_class
,
2093 #endif /* CONFIG_RTP_DEMUXER */