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"
31 #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
60 #define OFFSET(x) offsetof(RTSPState, x)
61 #define DEC AV_OPT_FLAG_DECODING_PARAM
62 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 #define RTSP_FLAG_OPTS(name, longname) \
65 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
66 { "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 #define RTSP_MEDIATYPE_OPTS(name, longname) \
69 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
70 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
71 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
74 const AVOption ff_rtsp_options
[] = {
75 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause
), AV_OPT_TYPE_INT
, {0}, 0, 1, DEC
},
76 FF_RTP_FLAG_OPTS(RTSPState
, rtp_muxer_flags
),
77 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask
), AV_OPT_TYPE_FLAGS
, {0}, INT_MIN
, INT_MAX
, DEC
|ENC
, "rtsp_transport" }, \
78 { "udp", "UDP", 0, AV_OPT_TYPE_CONST
, {1 << RTSP_LOWER_TRANSPORT_UDP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
79 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST
, {1 << RTSP_LOWER_TRANSPORT_TCP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
80 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST
, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST
}, 0, 0, DEC
, "rtsp_transport" },
81 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST
, {(1 << RTSP_LOWER_TRANSPORT_HTTP
)}, 0, 0, DEC
, "rtsp_transport" },
82 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
83 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
87 static const AVOption sdp_options
[] = {
88 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
89 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
93 static const AVOption rtp_options
[] = {
94 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
98 static void get_word_until_chars(char *buf
, int buf_size
,
99 const char *sep
, const char **pp
)
105 p
+= strspn(p
, SPACE_CHARS
);
107 while (!strchr(sep
, *p
) && *p
!= '\0') {
108 if ((q
- buf
) < buf_size
- 1)
117 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
120 if (**pp
== '/') (*pp
)++;
121 get_word_until_chars(buf
, buf_size
, sep
, pp
);
124 static void get_word(char *buf
, int buf_size
, const char **pp
)
126 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
129 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
131 * Used for seeking in the rtp stream.
133 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
137 p
+= strspn(p
, SPACE_CHARS
);
138 if (!av_stristart(p
, "npt=", &p
))
141 *start
= AV_NOPTS_VALUE
;
142 *end
= AV_NOPTS_VALUE
;
144 get_word_sep(buf
, sizeof(buf
), "-", &p
);
145 av_parse_time(start
, buf
, 1);
148 get_word_sep(buf
, sizeof(buf
), "-", &p
);
149 av_parse_time(end
, buf
, 1);
151 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
152 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
155 static int get_sockaddr(const char *buf
, struct sockaddr_storage
*sock
)
157 struct addrinfo hints
, *ai
= NULL
;
158 memset(&hints
, 0, sizeof(hints
));
159 hints
.ai_flags
= AI_NUMERICHOST
;
160 if (getaddrinfo(buf
, NULL
, &hints
, &ai
))
162 memcpy(sock
, ai
->ai_addr
, FFMIN(sizeof(*sock
), ai
->ai_addrlen
));
168 static void init_rtp_handler(RTPDynamicProtocolHandler
*handler
,
169 RTSPStream
*rtsp_st
, AVCodecContext
*codec
)
173 codec
->codec_id
= handler
->codec_id
;
174 rtsp_st
->dynamic_handler
= handler
;
175 if (handler
->alloc
) {
176 rtsp_st
->dynamic_protocol_context
= handler
->alloc();
177 if (!rtsp_st
->dynamic_protocol_context
)
178 rtsp_st
->dynamic_handler
= NULL
;
182 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
183 static int sdp_parse_rtpmap(AVFormatContext
*s
,
184 AVStream
*st
, RTSPStream
*rtsp_st
,
185 int payload_type
, const char *p
)
187 AVCodecContext
*codec
= st
->codec
;
193 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
194 * see if we can handle this kind of payload.
195 * The space should normally not be there but some Real streams or
196 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
197 * have a trailing space. */
198 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
199 if (payload_type
>= RTP_PT_PRIVATE
) {
200 RTPDynamicProtocolHandler
*handler
=
201 ff_rtp_handler_find_by_name(buf
, codec
->codec_type
);
202 init_rtp_handler(handler
, rtsp_st
, codec
);
203 /* If no dynamic handler was found, check with the list of standard
204 * allocated types, if such a stream for some reason happens to
205 * use a private payload type. This isn't handled in rtpdec.c, since
206 * the format name from the rtpmap line never is passed into rtpdec. */
207 if (!rtsp_st
->dynamic_handler
)
208 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 /* search into AVRtpPayloadTypes[] */
213 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
216 c
= avcodec_find_decoder(codec
->codec_id
);
222 get_word_sep(buf
, sizeof(buf
), "/", &p
);
224 switch (codec
->codec_type
) {
225 case AVMEDIA_TYPE_AUDIO
:
226 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
227 codec
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
228 codec
->channels
= RTSP_DEFAULT_NB_AUDIO_CHANNELS
;
230 codec
->sample_rate
= i
;
231 av_set_pts_info(st
, 32, 1, codec
->sample_rate
);
232 get_word_sep(buf
, sizeof(buf
), "/", &p
);
236 // TODO: there is a bug here; if it is a mono stream, and
237 // less than 22000Hz, faad upconverts to stereo and twice
238 // the frequency. No problem, but the sample rate is being
239 // set here by the sdp line. Patch on its way. (rdm)
241 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
243 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
246 case AVMEDIA_TYPE_VIDEO
:
247 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
249 av_set_pts_info(st
, 32, 1, i
);
257 /* parse the attribute line from the fmtp a line of an sdp response. This
258 * is broken out as a function because it is used in rtp_h264.c, which is
260 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
261 char *value
, int value_size
)
263 *p
+= strspn(*p
, SPACE_CHARS
);
265 get_word_sep(attr
, attr_size
, "=", p
);
268 get_word_sep(value
, value_size
, ";", p
);
276 typedef struct SDPParseState
{
278 struct sockaddr_storage default_ip
;
280 int skip_media
; ///< set if an unknown m= line occurs
283 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
284 int letter
, const char *buf
)
286 RTSPState
*rt
= s
->priv_data
;
287 char buf1
[64], st_type
[64];
289 enum AVMediaType codec_type
;
293 struct sockaddr_storage sdp_ip
;
296 av_dlog(s
, "sdp: %c='%s'\n", letter
, buf
);
299 if (s1
->skip_media
&& letter
!= 'm')
303 get_word(buf1
, sizeof(buf1
), &p
);
304 if (strcmp(buf1
, "IN") != 0)
306 get_word(buf1
, sizeof(buf1
), &p
);
307 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6"))
309 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
310 if (get_sockaddr(buf1
, &sdp_ip
))
315 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
318 if (s
->nb_streams
== 0) {
319 s1
->default_ip
= sdp_ip
;
320 s1
->default_ttl
= ttl
;
322 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
323 rtsp_st
->sdp_ip
= sdp_ip
;
324 rtsp_st
->sdp_ttl
= ttl
;
328 av_dict_set(&s
->metadata
, "title", p
, 0);
331 if (s
->nb_streams
== 0) {
332 av_dict_set(&s
->metadata
, "comment", p
, 0);
339 codec_type
= AVMEDIA_TYPE_UNKNOWN
;
340 get_word(st_type
, sizeof(st_type
), &p
);
341 if (!strcmp(st_type
, "audio")) {
342 codec_type
= AVMEDIA_TYPE_AUDIO
;
343 } else if (!strcmp(st_type
, "video")) {
344 codec_type
= AVMEDIA_TYPE_VIDEO
;
345 } else if (!strcmp(st_type
, "application")) {
346 codec_type
= AVMEDIA_TYPE_DATA
;
348 if (codec_type
== AVMEDIA_TYPE_UNKNOWN
|| !(rt
->media_type_mask
& (1 << codec_type
))) {
352 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
355 rtsp_st
->stream_index
= -1;
356 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
358 rtsp_st
->sdp_ip
= s1
->default_ip
;
359 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
361 get_word(buf1
, sizeof(buf1
), &p
); /* port */
362 rtsp_st
->sdp_port
= atoi(buf1
);
364 get_word(buf1
, sizeof(buf1
), &p
); /* protocol (ignored) */
366 /* XXX: handle list of formats */
367 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
368 rtsp_st
->sdp_payload_type
= atoi(buf1
);
370 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
371 /* no corresponding stream */
373 st
= avformat_new_stream(s
, NULL
);
376 st
->id
= rt
->nb_rtsp_streams
- 1;
377 rtsp_st
->stream_index
= st
->index
;
378 st
->codec
->codec_type
= codec_type
;
379 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
380 RTPDynamicProtocolHandler
*handler
;
381 /* if standard payload type, we can find the codec right now */
382 ff_rtp_get_codec_info(st
->codec
, rtsp_st
->sdp_payload_type
);
383 if (st
->codec
->codec_type
== AVMEDIA_TYPE_AUDIO
&&
384 st
->codec
->sample_rate
> 0)
385 av_set_pts_info(st
, 32, 1, st
->codec
->sample_rate
);
386 /* Even static payload types may need a custom depacketizer */
387 handler
= ff_rtp_handler_find_by_id(
388 rtsp_st
->sdp_payload_type
, st
->codec
->codec_type
);
389 init_rtp_handler(handler
, rtsp_st
, st
->codec
);
392 /* put a default control url */
393 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
394 sizeof(rtsp_st
->control_url
));
397 if (av_strstart(p
, "control:", &p
)) {
398 if (s
->nb_streams
== 0) {
399 if (!strncmp(p
, "rtsp://", 7))
400 av_strlcpy(rt
->control_uri
, p
,
401 sizeof(rt
->control_uri
));
404 /* get the control url */
405 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
407 /* XXX: may need to add full url resolution */
408 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
410 if (proto
[0] == '\0') {
411 /* relative control URL */
412 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
413 av_strlcat(rtsp_st
->control_url
, "/",
414 sizeof(rtsp_st
->control_url
));
415 av_strlcat(rtsp_st
->control_url
, p
,
416 sizeof(rtsp_st
->control_url
));
418 av_strlcpy(rtsp_st
->control_url
, p
,
419 sizeof(rtsp_st
->control_url
));
421 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
422 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
423 get_word(buf1
, sizeof(buf1
), &p
);
424 payload_type
= atoi(buf1
);
425 st
= s
->streams
[s
->nb_streams
- 1];
426 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
427 sdp_parse_rtpmap(s
, st
, rtsp_st
, payload_type
, p
);
428 } else if (av_strstart(p
, "fmtp:", &p
) ||
429 av_strstart(p
, "framesize:", &p
)) {
430 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
431 // let dynamic protocol handlers have a stab at the line.
432 get_word(buf1
, sizeof(buf1
), &p
);
433 payload_type
= atoi(buf1
);
434 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
435 rtsp_st
= rt
->rtsp_streams
[i
];
436 if (rtsp_st
->sdp_payload_type
== payload_type
&&
437 rtsp_st
->dynamic_handler
&&
438 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
439 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, i
,
440 rtsp_st
->dynamic_protocol_context
, buf
);
442 } else if (av_strstart(p
, "range:", &p
)) {
445 // this is so that seeking on a streamed file can work.
446 rtsp_parse_range_npt(p
, &start
, &end
);
447 s
->start_time
= start
;
448 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
449 s
->duration
= (end
== AV_NOPTS_VALUE
) ?
450 AV_NOPTS_VALUE
: end
- start
;
451 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
453 rt
->transport
= RTSP_TRANSPORT_RDT
;
454 } else if (av_strstart(p
, "SampleRate:integer;", &p
) &&
456 st
= s
->streams
[s
->nb_streams
- 1];
457 st
->codec
->sample_rate
= atoi(p
);
459 if (rt
->server_type
== RTSP_SERVER_WMS
)
460 ff_wms_parse_sdp_a_line(s
, p
);
461 if (s
->nb_streams
> 0) {
462 if (rt
->server_type
== RTSP_SERVER_REAL
)
463 ff_real_parse_sdp_a_line(s
, s
->nb_streams
- 1, p
);
465 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
466 if (rtsp_st
->dynamic_handler
&&
467 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
468 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
470 rtsp_st
->dynamic_protocol_context
, buf
);
477 int ff_sdp_parse(AVFormatContext
*s
, const char *content
)
479 RTSPState
*rt
= s
->priv_data
;
482 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
483 * contain long SDP lines containing complete ASF Headers (several
484 * kB) or arrays of MDPR (RM stream descriptor) headers plus
485 * "rulebooks" describing their properties. Therefore, the SDP line
488 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
489 * in rtpdec_xiph.c. */
491 SDPParseState sdp_parse_state
, *s1
= &sdp_parse_state
;
493 memset(s1
, 0, sizeof(SDPParseState
));
496 p
+= strspn(p
, SPACE_CHARS
);
504 /* get the content */
506 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
507 if ((q
- buf
) < sizeof(buf
) - 1)
512 sdp_parse_line(s
, s1
, letter
, buf
);
514 while (*p
!= '\n' && *p
!= '\0')
519 rt
->p
= av_malloc(sizeof(struct pollfd
)*2*(rt
->nb_rtsp_streams
+1));
520 if (!rt
->p
) return AVERROR(ENOMEM
);
523 #endif /* CONFIG_RTPDEC */
525 void ff_rtsp_undo_setup(AVFormatContext
*s
)
527 RTSPState
*rt
= s
->priv_data
;
530 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
531 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
534 if (rtsp_st
->transport_priv
) {
536 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
537 av_write_trailer(rtpctx
);
538 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
540 avio_close_dyn_buf(rtpctx
->pb
, &ptr
);
543 avio_close(rtpctx
->pb
);
545 avformat_free_context(rtpctx
);
546 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
547 ff_rdt_parse_close(rtsp_st
->transport_priv
);
548 else if (CONFIG_RTPDEC
)
549 ff_rtp_parse_close(rtsp_st
->transport_priv
);
551 rtsp_st
->transport_priv
= NULL
;
552 if (rtsp_st
->rtp_handle
)
553 ffurl_close(rtsp_st
->rtp_handle
);
554 rtsp_st
->rtp_handle
= NULL
;
558 /* close and free RTSP streams */
559 void ff_rtsp_close_streams(AVFormatContext
*s
)
561 RTSPState
*rt
= s
->priv_data
;
565 ff_rtsp_undo_setup(s
);
566 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
567 rtsp_st
= rt
->rtsp_streams
[i
];
569 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
)
570 rtsp_st
->dynamic_handler
->free(
571 rtsp_st
->dynamic_protocol_context
);
575 av_free(rt
->rtsp_streams
);
577 av_close_input_stream (rt
->asf_ctx
);
581 av_free(rt
->recvbuf
);
584 static int rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
586 RTSPState
*rt
= s
->priv_data
;
589 /* open the RTP context */
590 if (rtsp_st
->stream_index
>= 0)
591 st
= s
->streams
[rtsp_st
->stream_index
];
593 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
595 if (s
->oformat
&& CONFIG_RTSP_MUXER
) {
596 rtsp_st
->transport_priv
= ff_rtp_chain_mux_open(s
, st
,
598 RTSP_TCP_MAX_PACKET_SIZE
);
599 /* Ownership of rtp_handle is passed to the rtp mux context */
600 rtsp_st
->rtp_handle
= NULL
;
601 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
602 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
603 rtsp_st
->dynamic_protocol_context
,
604 rtsp_st
->dynamic_handler
);
605 else if (CONFIG_RTPDEC
)
606 rtsp_st
->transport_priv
= ff_rtp_parse_open(s
, st
, rtsp_st
->rtp_handle
,
607 rtsp_st
->sdp_payload_type
,
608 (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
|| !s
->max_delay
)
609 ?
0 : RTP_REORDER_QUEUE_DEFAULT_SIZE
);
611 if (!rtsp_st
->transport_priv
) {
612 return AVERROR(ENOMEM
);
613 } else if (rt
->transport
!= RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
) {
614 if (rtsp_st
->dynamic_handler
) {
615 ff_rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
616 rtsp_st
->dynamic_protocol_context
,
617 rtsp_st
->dynamic_handler
);
624 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
625 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
631 p
+= strspn(p
, SPACE_CHARS
);
632 v
= strtol(p
, (char **)&p
, 10);
636 v
= strtol(p
, (char **)&p
, 10);
645 /* XXX: only one transport specification is parsed */
646 static void rtsp_parse_transport(RTSPMessageHeader
*reply
, const char *p
)
648 char transport_protocol
[16];
650 char lower_transport
[16];
652 RTSPTransportField
*th
;
655 reply
->nb_transports
= 0;
658 p
+= strspn(p
, SPACE_CHARS
);
662 th
= &reply
->transports
[reply
->nb_transports
];
664 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
666 if (!av_strcasecmp (transport_protocol
, "rtp")) {
667 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
668 lower_transport
[0] = '\0';
669 /* rtp/avp/<protocol> */
671 get_word_sep(lower_transport
, sizeof(lower_transport
),
674 th
->transport
= RTSP_TRANSPORT_RTP
;
675 } else if (!av_strcasecmp (transport_protocol
, "x-pn-tng") ||
676 !av_strcasecmp (transport_protocol
, "x-real-rdt")) {
677 /* x-pn-tng/<protocol> */
678 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
680 th
->transport
= RTSP_TRANSPORT_RDT
;
682 if (!av_strcasecmp(lower_transport
, "TCP"))
683 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
685 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
689 /* get each parameter */
690 while (*p
!= '\0' && *p
!= ',') {
691 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
692 if (!strcmp(parameter
, "port")) {
695 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
697 } else if (!strcmp(parameter
, "client_port")) {
700 rtsp_parse_range(&th
->client_port_min
,
701 &th
->client_port_max
, &p
);
703 } else if (!strcmp(parameter
, "server_port")) {
706 rtsp_parse_range(&th
->server_port_min
,
707 &th
->server_port_max
, &p
);
709 } else if (!strcmp(parameter
, "interleaved")) {
712 rtsp_parse_range(&th
->interleaved_min
,
713 &th
->interleaved_max
, &p
);
715 } else if (!strcmp(parameter
, "multicast")) {
716 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
717 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
718 } else if (!strcmp(parameter
, "ttl")) {
721 th
->ttl
= strtol(p
, (char **)&p
, 10);
723 } else if (!strcmp(parameter
, "destination")) {
726 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
727 get_sockaddr(buf
, &th
->destination
);
729 } else if (!strcmp(parameter
, "source")) {
732 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
733 av_strlcpy(th
->source
, buf
, sizeof(th
->source
));
737 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
745 reply
->nb_transports
++;
749 static void handle_rtp_info(RTSPState
*rt
, const char *url
,
750 uint32_t seq
, uint32_t rtptime
)
753 if (!rtptime
|| !url
[0])
755 if (rt
->transport
!= RTSP_TRANSPORT_RTP
)
757 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
758 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
759 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
762 if (!strcmp(rtsp_st
->control_url
, url
)) {
763 rtpctx
->base_timestamp
= rtptime
;
769 static void rtsp_parse_rtp_info(RTSPState
*rt
, const char *p
)
772 char key
[20], value
[1024], url
[1024] = "";
773 uint32_t seq
= 0, rtptime
= 0;
776 p
+= strspn(p
, SPACE_CHARS
);
779 get_word_sep(key
, sizeof(key
), "=", &p
);
783 get_word_sep(value
, sizeof(value
), ";, ", &p
);
785 if (!strcmp(key
, "url"))
786 av_strlcpy(url
, value
, sizeof(url
));
787 else if (!strcmp(key
, "seq"))
788 seq
= strtoul(value
, NULL
, 10);
789 else if (!strcmp(key
, "rtptime"))
790 rtptime
= strtoul(value
, NULL
, 10);
792 handle_rtp_info(rt
, url
, seq
, rtptime
);
801 handle_rtp_info(rt
, url
, seq
, rtptime
);
804 void ff_rtsp_parse_line(RTSPMessageHeader
*reply
, const char *buf
,
805 RTSPState
*rt
, const char *method
)
809 /* NOTE: we do case independent match for broken servers */
811 if (av_stristart(p
, "Session:", &p
)) {
813 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
814 if (av_stristart(p
, ";timeout=", &p
) &&
815 (t
= strtol(p
, NULL
, 10)) > 0) {
818 } else if (av_stristart(p
, "Content-Length:", &p
)) {
819 reply
->content_length
= strtol(p
, NULL
, 10);
820 } else if (av_stristart(p
, "Transport:", &p
)) {
821 rtsp_parse_transport(reply
, p
);
822 } else if (av_stristart(p
, "CSeq:", &p
)) {
823 reply
->seq
= strtol(p
, NULL
, 10);
824 } else if (av_stristart(p
, "Range:", &p
)) {
825 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
826 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
827 p
+= strspn(p
, SPACE_CHARS
);
828 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
829 } else if (av_stristart(p
, "Server:", &p
)) {
830 p
+= strspn(p
, SPACE_CHARS
);
831 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
832 } else if (av_stristart(p
, "Notice:", &p
) ||
833 av_stristart(p
, "X-Notice:", &p
)) {
834 reply
->notice
= strtol(p
, NULL
, 10);
835 } else if (av_stristart(p
, "Location:", &p
)) {
836 p
+= strspn(p
, SPACE_CHARS
);
837 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
838 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && rt
) {
839 p
+= strspn(p
, SPACE_CHARS
);
840 ff_http_auth_handle_header(&rt
->auth_state
, "WWW-Authenticate", p
);
841 } else if (av_stristart(p
, "Authentication-Info:", &p
) && rt
) {
842 p
+= strspn(p
, SPACE_CHARS
);
843 ff_http_auth_handle_header(&rt
->auth_state
, "Authentication-Info", p
);
844 } else if (av_stristart(p
, "Content-Base:", &p
) && rt
) {
845 p
+= strspn(p
, SPACE_CHARS
);
846 if (method
&& !strcmp(method
, "DESCRIBE"))
847 av_strlcpy(rt
->control_uri
, p
, sizeof(rt
->control_uri
));
848 } else if (av_stristart(p
, "RTP-Info:", &p
) && rt
) {
849 p
+= strspn(p
, SPACE_CHARS
);
850 if (method
&& !strcmp(method
, "PLAY"))
851 rtsp_parse_rtp_info(rt
, p
);
852 } else if (av_stristart(p
, "Public:", &p
) && rt
) {
853 if (strstr(p
, "GET_PARAMETER") &&
854 method
&& !strcmp(method
, "OPTIONS"))
855 rt
->get_parameter_supported
= 1;
856 } else if (av_stristart(p
, "x-Accept-Dynamic-Rate:", &p
) && rt
) {
857 p
+= strspn(p
, SPACE_CHARS
);
858 rt
->accept_dynamic_rate
= atoi(p
);
862 /* skip a RTP/TCP interleaved packet */
863 void ff_rtsp_skip_packet(AVFormatContext
*s
)
865 RTSPState
*rt
= s
->priv_data
;
869 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, 3);
872 len
= AV_RB16(buf
+ 1);
874 av_dlog(s
, "skipping RTP packet len=%d\n", len
);
879 if (len1
> sizeof(buf
))
881 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, len1
);
888 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
889 unsigned char **content_ptr
,
890 int return_on_interleaved_data
, const char *method
)
892 RTSPState
*rt
= s
->priv_data
;
893 char buf
[4096], buf1
[1024], *q
;
896 int ret
, content_length
, line_count
= 0;
897 unsigned char *content
= NULL
;
899 memset(reply
, 0, sizeof(*reply
));
901 /* parse reply (XXX: use buffers) */
902 rt
->last_reply
[0] = '\0';
906 ret
= ffurl_read_complete(rt
->rtsp_hd
, &ch
, 1);
907 av_dlog(s
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
913 /* XXX: only parse it if first char on line ? */
914 if (return_on_interleaved_data
) {
917 ff_rtsp_skip_packet(s
);
918 } else if (ch
!= '\r') {
919 if ((q
- buf
) < sizeof(buf
) - 1)
925 av_dlog(s
, "line='%s'\n", buf
);
927 /* test if last line */
931 if (line_count
== 0) {
933 get_word(buf1
, sizeof(buf1
), &p
);
934 get_word(buf1
, sizeof(buf1
), &p
);
935 reply
->status_code
= atoi(buf1
);
936 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
938 ff_rtsp_parse_line(reply
, p
, rt
, method
);
939 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
940 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
945 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0')
946 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
948 content_length
= reply
->content_length
;
949 if (content_length
> 0) {
950 /* leave some room for a trailing '\0' (useful for simple parsing) */
951 content
= av_malloc(content_length
+ 1);
952 ffurl_read_complete(rt
->rtsp_hd
, content
, content_length
);
953 content
[content_length
] = '\0';
956 *content_ptr
= content
;
960 if (rt
->seq
!= reply
->seq
) {
961 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
962 rt
->seq
, reply
->seq
);
966 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
967 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
968 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
969 rt
->state
= RTSP_STATE_IDLE
;
970 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
971 return AVERROR(EIO
); /* data or server error */
972 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
973 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
974 return AVERROR(EPERM
);
980 * Send a command to the RTSP server without waiting for the reply.
982 * @param s RTSP (de)muxer context
983 * @param method the method for the request
984 * @param url the target url for the request
985 * @param headers extra header lines to include in the request
986 * @param send_content if non-null, the data to send as request body content
987 * @param send_content_length the length of the send_content data, or 0 if
988 * send_content is null
990 * @return zero if success, nonzero otherwise
992 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
993 const char *method
, const char *url
,
995 const unsigned char *send_content
,
996 int send_content_length
)
998 RTSPState
*rt
= s
->priv_data
;
999 char buf
[4096], *out_buf
;
1000 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1002 /* Add in RTSP headers */
1005 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
1007 av_strlcat(buf
, headers
, sizeof(buf
));
1008 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
1009 if (rt
->session_id
[0] != '\0' && (!headers
||
1010 !strstr(headers
, "\nIf-Match:"))) {
1011 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
1014 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
1015 rt
->auth
, url
, method
);
1017 av_strlcat(buf
, str
, sizeof(buf
));
1020 if (send_content_length
> 0 && send_content
)
1021 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
1022 av_strlcat(buf
, "\r\n", sizeof(buf
));
1024 /* base64 encode rtsp if tunneling */
1025 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1026 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1027 out_buf
= base64buf
;
1030 av_dlog(s
, "Sending:\n%s--\n", buf
);
1032 ffurl_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
1033 if (send_content_length
> 0 && send_content
) {
1034 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1035 av_log(s
, AV_LOG_ERROR
, "tunneling of RTSP requests "
1036 "with content data not supported\n");
1037 return AVERROR_PATCHWELCOME
;
1039 ffurl_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
1041 rt
->last_cmd_time
= av_gettime();
1046 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
1047 const char *url
, const char *headers
)
1049 return ff_rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
1052 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
1053 const char *headers
, RTSPMessageHeader
*reply
,
1054 unsigned char **content_ptr
)
1056 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
1057 content_ptr
, NULL
, 0);
1060 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
1061 const char *method
, const char *url
,
1063 RTSPMessageHeader
*reply
,
1064 unsigned char **content_ptr
,
1065 const unsigned char *send_content
,
1066 int send_content_length
)
1068 RTSPState
*rt
= s
->priv_data
;
1069 HTTPAuthType cur_auth_type
;
1073 cur_auth_type
= rt
->auth_state
.auth_type
;
1074 if ((ret
= ff_rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
1076 send_content_length
)))
1079 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0, method
) ) < 0)
1082 if (reply
->status_code
== 401 && cur_auth_type
== HTTP_AUTH_NONE
&&
1083 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
)
1086 if (reply
->status_code
> 400){
1087 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
1091 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
1097 int ff_rtsp_make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
1098 int lower_transport
, const char *real_challenge
)
1100 RTSPState
*rt
= s
->priv_data
;
1101 int rtx
, j
, i
, err
, interleave
= 0;
1102 RTSPStream
*rtsp_st
;
1103 RTSPMessageHeader reply1
, *reply
= &reply1
;
1105 const char *trans_pref
;
1107 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
1108 trans_pref
= "x-pn-tng";
1110 trans_pref
= "RTP/AVP";
1112 /* default timeout: 1 minute */
1115 /* for each stream, make the setup request */
1116 /* XXX: we assume the same server is used for the control of each
1119 for (j
= RTSP_RTP_PORT_MIN
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1120 char transport
[2048];
1123 * WMS serves all UDP data over a single connection, the RTX, which
1124 * isn't necessarily the first in the SDP but has to be the first
1125 * to be set up, else the second/third SETUP will fail with a 461.
1127 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1128 rt
->server_type
== RTSP_SERVER_WMS
) {
1131 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1132 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1134 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1138 if (rtx
== rt
->nb_rtsp_streams
)
1139 return -1; /* no RTX found */
1140 rtsp_st
= rt
->rtsp_streams
[rtx
];
1142 rtsp_st
= rt
->rtsp_streams
[i
> rtx ? i
: i
- 1];
1144 rtsp_st
= rt
->rtsp_streams
[i
];
1147 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1150 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1151 port
= reply
->transports
[0].client_port_min
;
1155 /* first try in specified port range */
1156 if (RTSP_RTP_PORT_MIN
!= 0) {
1157 while (j
<= RTSP_RTP_PORT_MAX
) {
1158 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1159 "?localport=%d", j
);
1160 /* we will use two ports per rtp stream (rtp and rtcp) */
1162 if (ffurl_open(&rtsp_st
->rtp_handle
, buf
, AVIO_FLAG_READ_WRITE
) == 0)
1167 av_log(s
, AV_LOG_ERROR
, "Unable to open an input RTP port\n");
1172 port
= ff_rtp_get_local_rtp_port(rtsp_st
->rtp_handle
);
1174 snprintf(transport
, sizeof(transport
) - 1,
1175 "%s/UDP;", trans_pref
);
1176 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1177 av_strlcat(transport
, "unicast;", sizeof(transport
));
1178 av_strlcatf(transport
, sizeof(transport
),
1179 "client_port=%d", port
);
1180 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1181 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1182 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1186 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1187 /* For WMS streams, the application streams are only used for
1188 * UDP. When trying to set it up for TCP streams, the server
1189 * will return an error. Therefore, we skip those streams. */
1190 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1191 s
->streams
[rtsp_st
->stream_index
]->codec
->codec_type
==
1194 snprintf(transport
, sizeof(transport
) - 1,
1195 "%s/TCP;", trans_pref
);
1196 if (rt
->transport
!= RTSP_TRANSPORT_RDT
)
1197 av_strlcat(transport
, "unicast;", sizeof(transport
));
1198 av_strlcatf(transport
, sizeof(transport
),
1199 "interleaved=%d-%d",
1200 interleave
, interleave
+ 1);
1204 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1205 snprintf(transport
, sizeof(transport
) - 1,
1206 "%s/UDP;multicast", trans_pref
);
1209 av_strlcat(transport
, ";mode=receive", sizeof(transport
));
1210 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1211 rt
->server_type
== RTSP_SERVER_WMS
)
1212 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1213 snprintf(cmd
, sizeof(cmd
),
1214 "Transport: %s\r\n",
1216 if (rt
->accept_dynamic_rate
)
1217 av_strlcat(cmd
, "x-Dynamic-Rate: 0\r\n", sizeof(cmd
));
1218 if (i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
&& CONFIG_RTPDEC
) {
1219 char real_res
[41], real_csum
[9];
1220 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1222 av_strlcatf(cmd
, sizeof(cmd
),
1224 "RealChallenge2: %s, sd=%s\r\n",
1225 rt
->session_id
, real_res
, real_csum
);
1227 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1228 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1231 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1232 reply
->nb_transports
!= 1) {
1233 err
= AVERROR_INVALIDDATA
;
1237 /* XXX: same protocol for all streams is required */
1239 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1240 reply
->transports
[0].transport
!= rt
->transport
) {
1241 err
= AVERROR_INVALIDDATA
;
1245 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1246 rt
->transport
= reply
->transports
[0].transport
;
1249 /* Fail if the server responded with another lower transport mode
1250 * than what we requested. */
1251 if (reply
->transports
[0].lower_transport
!= lower_transport
) {
1252 av_log(s
, AV_LOG_ERROR
, "Nonmatching transport in server reply\n");
1253 err
= AVERROR_INVALIDDATA
;
1257 switch(reply
->transports
[0].lower_transport
) {
1258 case RTSP_LOWER_TRANSPORT_TCP
:
1259 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1260 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1263 case RTSP_LOWER_TRANSPORT_UDP
: {
1264 char url
[1024], options
[30] = "";
1266 if (rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
)
1267 av_strlcpy(options
, "?connect=1", sizeof(options
));
1268 /* Use source address if specified */
1269 if (reply
->transports
[0].source
[0]) {
1270 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1271 reply
->transports
[0].source
,
1272 reply
->transports
[0].server_port_min
, "%s", options
);
1274 ff_url_join(url
, sizeof(url
), "rtp", NULL
, host
,
1275 reply
->transports
[0].server_port_min
, "%s", options
);
1277 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1278 ff_rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1279 err
= AVERROR_INVALIDDATA
;
1282 /* Try to initialize the connection state in a
1283 * potential NAT router by sending dummy packets.
1284 * RTP/RTCP dummy packets are used for RDT, too.
1286 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) && s
->iformat
&&
1288 ff_rtp_send_punch_packets(rtsp_st
->rtp_handle
);
1291 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1292 char url
[1024], namebuf
[50];
1293 struct sockaddr_storage addr
;
1296 if (reply
->transports
[0].destination
.ss_family
) {
1297 addr
= reply
->transports
[0].destination
;
1298 port
= reply
->transports
[0].port_min
;
1299 ttl
= reply
->transports
[0].ttl
;
1301 addr
= rtsp_st
->sdp_ip
;
1302 port
= rtsp_st
->sdp_port
;
1303 ttl
= rtsp_st
->sdp_ttl
;
1305 getnameinfo((struct sockaddr
*) &addr
, sizeof(addr
),
1306 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1307 ff_url_join(url
, sizeof(url
), "rtp", NULL
, namebuf
,
1308 port
, "?ttl=%d", ttl
);
1309 if (ffurl_open(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
) < 0) {
1310 err
= AVERROR_INVALIDDATA
;
1317 if ((err
= rtsp_open_transport_ctx(s
, rtsp_st
)))
1321 if (reply
->timeout
> 0)
1322 rt
->timeout
= reply
->timeout
;
1324 if (rt
->server_type
== RTSP_SERVER_REAL
)
1325 rt
->need_subscription
= 1;
1330 ff_rtsp_undo_setup(s
);
1334 void ff_rtsp_close_connections(AVFormatContext
*s
)
1336 RTSPState
*rt
= s
->priv_data
;
1337 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
) ffurl_close(rt
->rtsp_hd_out
);
1338 ffurl_close(rt
->rtsp_hd
);
1339 rt
->rtsp_hd
= rt
->rtsp_hd_out
= NULL
;
1342 int ff_rtsp_connect(AVFormatContext
*s
)
1344 RTSPState
*rt
= s
->priv_data
;
1345 char host
[1024], path
[1024], tcpname
[1024], cmd
[2048], auth
[128];
1346 char *option_list
, *option
, *filename
;
1347 int port
, err
, tcp_fd
;
1348 RTSPMessageHeader reply1
= {0}, *reply
= &reply1
;
1349 int lower_transport_mask
= 0;
1350 char real_challenge
[64] = "";
1351 struct sockaddr_storage peer
;
1352 socklen_t peer_len
= sizeof(peer
);
1354 if (!ff_network_init())
1355 return AVERROR(EIO
);
1357 rt
->control_transport
= RTSP_MODE_PLAIN
;
1358 if (rt
->lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_HTTP
)) {
1359 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1360 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1362 /* Only pass through valid flags from here */
1363 rt
->lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1366 lower_transport_mask
= rt
->lower_transport_mask
;
1367 /* extract hostname and port */
1368 av_url_split(NULL
, 0, auth
, sizeof(auth
),
1369 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->filename
);
1371 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1374 port
= RTSP_DEFAULT_PORT
;
1376 #if FF_API_RTSP_URL_OPTIONS
1377 /* search for options */
1378 option_list
= strrchr(path
, '?');
1380 /* Strip out the RTSP specific options, write out the rest of
1381 * the options back into the same string. */
1382 filename
= option_list
;
1383 while (option_list
) {
1385 /* move the option pointer */
1386 option
= ++option_list
;
1387 option_list
= strchr(option_list
, '&');
1391 /* handle the options */
1392 if (!strcmp(option
, "udp")) {
1393 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_UDP
);
1394 } else if (!strcmp(option
, "multicast")) {
1395 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST
);
1396 } else if (!strcmp(option
, "tcp")) {
1397 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_TCP
);
1398 } else if(!strcmp(option
, "http")) {
1399 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_TCP
);
1400 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1401 } else if (!strcmp(option
, "filter_src")) {
1402 rt
->rtsp_flags
|= RTSP_FLAG_FILTER_SRC
;
1404 /* Write options back into the buffer, using memmove instead
1405 * of strcpy since the strings may overlap. */
1406 int len
= strlen(option
);
1407 memmove(++filename
, option
, len
);
1409 if (option_list
) *filename
= '&';
1413 av_log(s
, AV_LOG_WARNING
, "Options passed via URL are "
1414 "deprecated, use -rtsp_transport "
1415 "and -rtsp_flags instead.\n");
1421 if (!lower_transport_mask
)
1422 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1425 /* Only UDP or TCP - UDP multicast isn't supported. */
1426 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1427 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1428 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1429 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1430 "only UDP and TCP are supported for output.\n");
1431 err
= AVERROR(EINVAL
);
1436 /* Construct the URI used in request; this is similar to s->filename,
1437 * but with authentication credentials removed and RTSP specific options
1439 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), "rtsp", NULL
,
1440 host
, port
, "%s", path
);
1442 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1443 /* set up initial handshake for tunneling */
1444 char httpname
[1024];
1445 char sessioncookie
[17];
1448 ff_url_join(httpname
, sizeof(httpname
), "http", auth
, host
, port
, "%s", path
);
1449 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1450 av_get_random_seed(), av_get_random_seed());
1453 if (ffurl_alloc(&rt
->rtsp_hd
, httpname
, AVIO_FLAG_READ
) < 0) {
1458 /* generate GET headers */
1459 snprintf(headers
, sizeof(headers
),
1460 "x-sessioncookie: %s\r\n"
1461 "Accept: application/x-rtsp-tunnelled\r\n"
1462 "Pragma: no-cache\r\n"
1463 "Cache-Control: no-cache\r\n",
1465 av_opt_set(rt
->rtsp_hd
->priv_data
, "headers", headers
, 0);
1467 /* complete the connection */
1468 if (ffurl_connect(rt
->rtsp_hd
)) {
1474 if (ffurl_alloc(&rt
->rtsp_hd_out
, httpname
, AVIO_FLAG_WRITE
) < 0 ) {
1479 /* generate POST headers */
1480 snprintf(headers
, sizeof(headers
),
1481 "x-sessioncookie: %s\r\n"
1482 "Content-Type: application/x-rtsp-tunnelled\r\n"
1483 "Pragma: no-cache\r\n"
1484 "Cache-Control: no-cache\r\n"
1485 "Content-Length: 32767\r\n"
1486 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1488 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "headers", headers
, 0);
1489 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "chunksize", "-1", 0);
1491 /* Initialize the authentication state for the POST session. The HTTP
1492 * protocol implementation doesn't properly handle multi-pass
1493 * authentication for POST requests, since it would require one of
1495 * - implementing Expect: 100-continue, which many HTTP servers
1496 * don't support anyway, even less the RTSP servers that do HTTP
1498 * - sending the whole POST data until getting a 401 reply specifying
1499 * what authentication method to use, then resending all that data
1500 * - waiting for potential 401 replies directly after sending the
1501 * POST header (waiting for some unspecified time)
1502 * Therefore, we copy the full auth state, which works for both basic
1503 * and digest. (For digest, we would have to synchronize the nonce
1504 * count variable between the two sessions, if we'd do more requests
1505 * with the original session, though.)
1507 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1509 /* complete the connection */
1510 if (ffurl_connect(rt
->rtsp_hd_out
)) {
1515 /* open the tcp connection */
1516 ff_url_join(tcpname
, sizeof(tcpname
), "tcp", NULL
, host
, port
, NULL
);
1517 if (ffurl_open(&rt
->rtsp_hd
, tcpname
, AVIO_FLAG_READ_WRITE
) < 0) {
1521 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1525 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1526 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1527 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1528 NULL
, 0, NI_NUMERICHOST
);
1531 /* request options supported by the server; this also detects server
1533 for (rt
->server_type
= RTSP_SERVER_RTP
;;) {
1535 if (rt
->server_type
== RTSP_SERVER_REAL
)
1538 * The following entries are required for proper
1539 * streaming from a Realmedia server. They are
1540 * interdependent in some way although we currently
1541 * don't quite understand how. Values were copied
1542 * from mplayer SVN r23589.
1543 * ClientChallenge is a 16-byte ID in hex
1544 * CompanyID is a 16-byte ID in base64
1546 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1547 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1548 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1549 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1551 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
1552 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1553 err
= AVERROR_INVALIDDATA
;
1557 /* detect server type if not standard-compliant RTP */
1558 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
1559 rt
->server_type
= RTSP_SERVER_REAL
;
1561 } else if (!av_strncasecmp(reply
->server
, "WMServer/", 9)) {
1562 rt
->server_type
= RTSP_SERVER_WMS
;
1563 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
1564 strcpy(real_challenge
, reply
->real_challenge
);
1568 if (s
->iformat
&& CONFIG_RTSP_DEMUXER
)
1569 err
= ff_rtsp_setup_input_streams(s
, reply
);
1570 else if (CONFIG_RTSP_MUXER
)
1571 err
= ff_rtsp_setup_output_streams(s
, host
);
1576 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
1577 ~(lower_transport_mask
- 1)];
1579 err
= ff_rtsp_make_setup_request(s
, host
, port
, lower_transport
,
1580 rt
->server_type
== RTSP_SERVER_REAL ?
1581 real_challenge
: NULL
);
1584 lower_transport_mask
&= ~(1 << lower_transport
);
1585 if (lower_transport_mask
== 0 && err
== 1) {
1586 err
= AVERROR(EPROTONOSUPPORT
);
1591 rt
->lower_transport_mask
= lower_transport_mask
;
1592 av_strlcpy(rt
->real_challenge
, real_challenge
, sizeof(rt
->real_challenge
));
1593 rt
->state
= RTSP_STATE_IDLE
;
1594 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
1597 ff_rtsp_close_streams(s
);
1598 ff_rtsp_close_connections(s
);
1599 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
1600 av_strlcpy(s
->filename
, reply
->location
, sizeof(s
->filename
));
1601 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
1609 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1612 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1613 uint8_t *buf
, int buf_size
, int64_t wait_end
)
1615 RTSPState
*rt
= s
->priv_data
;
1616 RTSPStream
*rtsp_st
;
1617 int n
, i
, ret
, tcp_fd
, timeout_cnt
= 0;
1619 struct pollfd
*p
= rt
->p
;
1622 if (url_interrupt_cb())
1623 return AVERROR_EXIT
;
1624 if (wait_end
&& wait_end
- av_gettime() < 0)
1625 return AVERROR(EAGAIN
);
1628 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1629 p
[max_p
].fd
= tcp_fd
;
1630 p
[max_p
++].events
= POLLIN
;
1634 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1635 rtsp_st
= rt
->rtsp_streams
[i
];
1636 if (rtsp_st
->rtp_handle
) {
1637 p
[max_p
].fd
= ffurl_get_file_handle(rtsp_st
->rtp_handle
);
1638 p
[max_p
++].events
= POLLIN
;
1639 p
[max_p
].fd
= ff_rtp_get_rtcp_file_handle(rtsp_st
->rtp_handle
);
1640 p
[max_p
++].events
= POLLIN
;
1643 n
= poll(p
, max_p
, POLL_TIMEOUT_MS
);
1645 int j
= 1 - (tcp_fd
== -1);
1647 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1648 rtsp_st
= rt
->rtsp_streams
[i
];
1649 if (rtsp_st
->rtp_handle
) {
1650 if (p
[j
].revents
& POLLIN
|| p
[j
+1].revents
& POLLIN
) {
1651 ret
= ffurl_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
1653 *prtsp_st
= rtsp_st
;
1660 #if CONFIG_RTSP_DEMUXER
1661 if (tcp_fd
!= -1 && p
[0].revents
& POLLIN
) {
1662 RTSPMessageHeader reply
;
1664 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0, NULL
);
1667 /* XXX: parse message */
1668 if (rt
->state
!= RTSP_STATE_STREAMING
)
1672 } else if (n
== 0 && ++timeout_cnt
>= MAX_TIMEOUTS
) {
1673 return AVERROR(ETIMEDOUT
);
1674 } else if (n
< 0 && errno
!= EINTR
)
1675 return AVERROR(errno
);
1679 int ff_rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
1681 RTSPState
*rt
= s
->priv_data
;
1683 RTSPStream
*rtsp_st
, *first_queue_st
= NULL
;
1684 int64_t wait_end
= 0;
1686 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1689 /* get next frames from the same RTP packet */
1690 if (rt
->cur_transport_priv
) {
1691 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1692 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1694 ret
= ff_rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1696 rt
->cur_transport_priv
= NULL
;
1698 } else if (ret
== 1) {
1701 rt
->cur_transport_priv
= NULL
;
1704 if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
1706 int64_t first_queue_time
= 0;
1707 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1708 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
1712 queue_time
= ff_rtp_queued_packet_time(rtpctx
);
1713 if (queue_time
&& (queue_time
- first_queue_time
< 0 ||
1714 !first_queue_time
)) {
1715 first_queue_time
= queue_time
;
1716 first_queue_st
= rt
->rtsp_streams
[i
];
1719 if (first_queue_time
)
1720 wait_end
= first_queue_time
+ s
->max_delay
;
1723 /* read next RTP packet */
1726 rt
->recvbuf
= av_malloc(RECVBUF_SIZE
);
1728 return AVERROR(ENOMEM
);
1731 switch(rt
->lower_transport
) {
1733 #if CONFIG_RTSP_DEMUXER
1734 case RTSP_LOWER_TRANSPORT_TCP
:
1735 len
= ff_rtsp_tcp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
);
1738 case RTSP_LOWER_TRANSPORT_UDP
:
1739 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
1740 len
= udp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
, wait_end
);
1741 if (len
> 0 && rtsp_st
->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
1742 ff_rtp_check_and_send_back_rr(rtsp_st
->transport_priv
, len
);
1745 if (len
== AVERROR(EAGAIN
) && first_queue_st
&&
1746 rt
->transport
== RTSP_TRANSPORT_RTP
) {
1747 rtsp_st
= first_queue_st
;
1748 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, NULL
, 0);
1755 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1756 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1758 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1760 /* Either bad packet, or a RTCP packet. Check if the
1761 * first_rtcp_ntp_time field was initialized. */
1762 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1763 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
1764 /* first_rtcp_ntp_time has been initialized for this stream,
1765 * copy the same value to all other uninitialized streams,
1766 * in order to map their timestamp origin to the same ntp time
1769 AVStream
*st
= NULL
;
1770 if (rtsp_st
->stream_index
>= 0)
1771 st
= s
->streams
[rtsp_st
->stream_index
];
1772 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1773 RTPDemuxContext
*rtpctx2
= rt
->rtsp_streams
[i
]->transport_priv
;
1774 AVStream
*st2
= NULL
;
1775 if (rt
->rtsp_streams
[i
]->stream_index
>= 0)
1776 st2
= s
->streams
[rt
->rtsp_streams
[i
]->stream_index
];
1777 if (rtpctx2
&& st
&& st2
&&
1778 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) {
1779 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
1780 rtpctx2
->rtcp_ts_offset
= av_rescale_q(
1781 rtpctx
->rtcp_ts_offset
, st
->time_base
,
1786 if (ret
== -RTCP_BYE
) {
1789 av_log(s
, AV_LOG_DEBUG
, "Received BYE for stream %d (%d/%d)\n",
1790 rtsp_st
->stream_index
, rt
->nb_byes
, rt
->nb_rtsp_streams
);
1792 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1801 /* more packets may follow, so we save the RTP context */
1802 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
1806 #endif /* CONFIG_RTPDEC */
1808 #if CONFIG_SDP_DEMUXER
1809 static int sdp_probe(AVProbeData
*p1
)
1811 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
1813 /* we look for a line beginning "c=IN IP" */
1814 while (p
< p_end
&& *p
!= '\0') {
1815 if (p
+ sizeof("c=IN IP") - 1 < p_end
&&
1816 av_strstart(p
, "c=IN IP", NULL
))
1817 return AVPROBE_SCORE_MAX
/ 2;
1819 while (p
< p_end
- 1 && *p
!= '\n') p
++;
1828 static int sdp_read_header(AVFormatContext
*s
, AVFormatParameters
*ap
)
1830 RTSPState
*rt
= s
->priv_data
;
1831 RTSPStream
*rtsp_st
;
1836 if (!ff_network_init())
1837 return AVERROR(EIO
);
1839 /* read the whole sdp file */
1840 /* XXX: better loading */
1841 content
= av_malloc(SDP_MAX_SIZE
);
1842 size
= avio_read(s
->pb
, content
, SDP_MAX_SIZE
- 1);
1845 return AVERROR_INVALIDDATA
;
1847 content
[size
] ='\0';
1849 err
= ff_sdp_parse(s
, content
);
1853 /* open each RTP stream */
1854 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1856 rtsp_st
= rt
->rtsp_streams
[i
];
1858 getnameinfo((struct sockaddr
*) &rtsp_st
->sdp_ip
, sizeof(rtsp_st
->sdp_ip
),
1859 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1860 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1861 namebuf
, rtsp_st
->sdp_port
,
1862 "?localport=%d&ttl=%d&connect=%d", rtsp_st
->sdp_port
,
1864 rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC ?
1 : 0);
1865 if (ffurl_open(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
) < 0) {
1866 err
= AVERROR_INVALIDDATA
;
1869 if ((err
= rtsp_open_transport_ctx(s
, rtsp_st
)))
1874 ff_rtsp_close_streams(s
);
1879 static int sdp_read_close(AVFormatContext
*s
)
1881 ff_rtsp_close_streams(s
);
1886 static const AVClass sdp_demuxer_class
= {
1887 .class_name
= "SDP demuxer",
1888 .item_name
= av_default_item_name
,
1889 .option
= sdp_options
,
1890 .version
= LIBAVUTIL_VERSION_INT
,
1893 AVInputFormat ff_sdp_demuxer
= {
1895 .long_name
= NULL_IF_CONFIG_SMALL("SDP"),
1896 .priv_data_size
= sizeof(RTSPState
),
1897 .read_probe
= sdp_probe
,
1898 .read_header
= sdp_read_header
,
1899 .read_packet
= ff_rtsp_fetch_packet
,
1900 .read_close
= sdp_read_close
,
1901 .priv_class
= &sdp_demuxer_class
1903 #endif /* CONFIG_SDP_DEMUXER */
1905 #if CONFIG_RTP_DEMUXER
1906 static int rtp_probe(AVProbeData
*p
)
1908 if (av_strstart(p
->filename
, "rtp:", NULL
))
1909 return AVPROBE_SCORE_MAX
;
1913 static int rtp_read_header(AVFormatContext
*s
,
1914 AVFormatParameters
*ap
)
1916 uint8_t recvbuf
[1500];
1917 char host
[500], sdp
[500];
1919 URLContext
* in
= NULL
;
1921 AVCodecContext codec
;
1922 struct sockaddr_storage addr
;
1924 socklen_t addrlen
= sizeof(addr
);
1926 if (!ff_network_init())
1927 return AVERROR(EIO
);
1929 ret
= ffurl_open(&in
, s
->filename
, AVIO_FLAG_READ
);
1934 ret
= ffurl_read(in
, recvbuf
, sizeof(recvbuf
));
1935 if (ret
== AVERROR(EAGAIN
))
1940 av_log(s
, AV_LOG_WARNING
, "Received too short packet\n");
1944 if ((recvbuf
[0] & 0xc0) != 0x80) {
1945 av_log(s
, AV_LOG_WARNING
, "Unsupported RTP version packet "
1950 payload_type
= recvbuf
[1] & 0x7f;
1953 getsockname(ffurl_get_file_handle(in
), (struct sockaddr
*) &addr
, &addrlen
);
1957 memset(&codec
, 0, sizeof(codec
));
1958 if (ff_rtp_get_codec_info(&codec
, payload_type
)) {
1959 av_log(s
, AV_LOG_ERROR
, "Unable to receive RTP payload type %d "
1960 "without an SDP file describing it\n",
1964 if (codec
.codec_type
!= AVMEDIA_TYPE_DATA
) {
1965 av_log(s
, AV_LOG_WARNING
, "Guessing on RTP content - if not received "
1966 "properly you need an SDP file "
1970 av_url_split(NULL
, 0, NULL
, 0, host
, sizeof(host
), &port
,
1971 NULL
, 0, s
->filename
);
1973 snprintf(sdp
, sizeof(sdp
),
1974 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1975 addr
.ss_family
== AF_INET ?
4 : 6, host
,
1976 codec
.codec_type
== AVMEDIA_TYPE_DATA ?
"application" :
1977 codec
.codec_type
== AVMEDIA_TYPE_VIDEO ?
"video" : "audio",
1978 port
, payload_type
);
1979 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
);
1981 ffio_init_context(&pb
, sdp
, strlen(sdp
), 0, NULL
, NULL
, NULL
, NULL
);
1984 /* sdp_read_header initializes this again */
1987 ret
= sdp_read_header(s
, ap
);
1998 static const AVClass rtp_demuxer_class
= {
1999 .class_name
= "RTP demuxer",
2000 .item_name
= av_default_item_name
,
2001 .option
= rtp_options
,
2002 .version
= LIBAVUTIL_VERSION_INT
,
2005 AVInputFormat ff_rtp_demuxer
= {
2007 .long_name
= NULL_IF_CONFIG_SMALL("RTP input format"),
2008 .priv_data_size
= sizeof(RTSPState
),
2009 .read_probe
= rtp_probe
,
2010 .read_header
= rtp_read_header
,
2011 .read_packet
= ff_rtsp_fetch_packet
,
2012 .read_close
= sdp_read_close
,
2013 .flags
= AVFMT_NOFILE
,
2014 .priv_class
= &rtp_demuxer_class
2016 #endif /* CONFIG_RTP_DEMUXER */