3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/bprint.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/parseutils.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/dict.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/time.h"
34 #include "avio_internal.h"
41 #include "os_support.h"
48 #include "rtpdec_formats.h"
49 #include "rtpenc_chain.h"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
76 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
78 #define COMMON_OPTS() \
79 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
80 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
81 { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
84 const AVOption ff_rtsp_options[] = {
85 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause
), AV_OPT_TYPE_BOOL
, {.i64
= 0}, 0, 1, DEC
},
86 FF_RTP_FLAG_OPTS(RTSPState
, rtp_muxer_flags
),
87 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask
), AV_OPT_TYPE_FLAGS
, {.i64
= 0}, INT_MIN
, INT_MAX
, DEC
|ENC
, "rtsp_transport" }, \
88 { "udp", "UDP", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_UDP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
89 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_TCP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
90 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST
}, 0, 0, DEC
, "rtsp_transport" },
91 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST
, {.i64
= (1 << RTSP_LOWER_TRANSPORT_HTTP
)}, 0, 0, DEC
, "rtsp_transport" },
92 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST
, {.i64
= (1 << RTSP_LOWER_TRANSPORT_HTTPS
)}, 0, 0, DEC
, "rtsp_transport" },
93 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
94 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_LISTEN
}, 0, 0, DEC
, "rtsp_flags" },
95 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_PREFER_TCP
}, 0, 0, DEC
|ENC
, "rtsp_flags" },
96 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
97 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min
), AV_OPT_TYPE_INT
, {.i64
= RTSP_RTP_PORT_MIN
}, 0, 65535, DEC
|ENC
},
98 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max
), AV_OPT_TYPE_INT
, {.i64
= RTSP_RTP_PORT_MAX
}, 0, 65535, DEC
|ENC
},
99 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout
), AV_OPT_TYPE_INT
, {.i64
= -1}, INT_MIN
, INT_MAX
, DEC
},
100 #if FF_API_OLD_RTSP_OPTIONS
101 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout
), AV_OPT_TYPE_INT
, {.i64
= -1}, INT_MIN
, INT_MAX
, DEC
},
102 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout
), AV_OPT_TYPE_INT
, {.i64
= 0}, INT_MIN
, INT_MAX
, DEC
},
104 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout
), AV_OPT_TYPE_INT
, {.i64
= 0}, INT_MIN
, INT_MAX
, DEC
},
107 { "user_agent", "override User-Agent header", OFFSET(user_agent
), AV_OPT_TYPE_STRING
, {.str
= LIBAVFORMAT_IDENT
}, 0, 0, DEC
},
108 #if FF_API_OLD_RTSP_OPTIONS
109 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent
), AV_OPT_TYPE_STRING
, {.str
= LIBAVFORMAT_IDENT
}, 0, 0, DEC
},
114 static const AVOption sdp_options
[] = {
115 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
116 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_CUSTOM_IO
}, 0, 0, DEC
, "rtsp_flags" },
117 { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_RTCP_TO_SOURCE
}, 0, 0, DEC
, "rtsp_flags" },
118 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
123 static const AVOption rtp_options
[] = {
124 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
130 static AVDictionary
*map_to_opts(RTSPState
*rt
)
132 AVDictionary
*opts
= NULL
;
135 snprintf(buf
, sizeof(buf
), "%d", rt
->buffer_size
);
136 av_dict_set(&opts
, "buffer_size", buf
, 0);
137 snprintf(buf
, sizeof(buf
), "%d", rt
->pkt_size
);
138 av_dict_set(&opts
, "pkt_size", buf
, 0);
143 static void get_word_until_chars(char *buf
, int buf_size
,
144 const char *sep
, const char **pp
)
150 p
+= strspn(p
, SPACE_CHARS
);
152 while (!strchr(sep
, *p
) && *p
!= '\0') {
153 if ((q
- buf
) < buf_size
- 1)
162 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
165 if (**pp
== '/') (*pp
)++;
166 get_word_until_chars(buf
, buf_size
, sep
, pp
);
169 static void get_word(char *buf
, int buf_size
, const char **pp
)
171 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
174 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
176 * Used for seeking in the rtp stream.
178 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
182 p
+= strspn(p
, SPACE_CHARS
);
183 if (!av_stristart(p
, "npt=", &p
))
186 *start
= AV_NOPTS_VALUE
;
187 *end
= AV_NOPTS_VALUE
;
189 get_word_sep(buf
, sizeof(buf
), "-", &p
);
190 if (av_parse_time(start
, buf
, 1) < 0)
194 get_word_sep(buf
, sizeof(buf
), "-", &p
);
195 if (av_parse_time(end
, buf
, 1) < 0)
196 av_log(NULL
, AV_LOG_DEBUG
, "Failed to parse interval end specification '%s'\n", buf
);
200 static int get_sockaddr(AVFormatContext
*s
,
201 const char *buf
, struct sockaddr_storage
*sock
)
203 struct addrinfo hints
= { 0 }, *ai
= NULL
;
206 hints
.ai_flags
= AI_NUMERICHOST
;
207 if ((ret
= getaddrinfo(buf
, NULL
, &hints
, &ai
))) {
208 av_log(s
, AV_LOG_ERROR
, "getaddrinfo(%s): %s\n",
213 memcpy(sock
, ai
->ai_addr
, FFMIN(sizeof(*sock
), ai
->ai_addrlen
));
219 static void init_rtp_handler(const RTPDynamicProtocolHandler
*handler
,
220 RTSPStream
*rtsp_st
, AVStream
*st
)
222 AVCodecParameters
*par
= st
? st
->codecpar
: NULL
;
226 par
->codec_id
= handler
->codec_id
;
227 rtsp_st
->dynamic_handler
= handler
;
229 st
->need_parsing
= handler
->need_parsing
;
230 if (handler
->priv_data_size
) {
231 rtsp_st
->dynamic_protocol_context
= av_mallocz(handler
->priv_data_size
);
232 if (!rtsp_st
->dynamic_protocol_context
)
233 rtsp_st
->dynamic_handler
= NULL
;
237 static void finalize_rtp_handler_init(AVFormatContext
*s
, RTSPStream
*rtsp_st
,
240 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_handler
->init
) {
241 int ret
= rtsp_st
->dynamic_handler
->init(s
, st
? st
->index
: -1,
242 rtsp_st
->dynamic_protocol_context
);
244 if (rtsp_st
->dynamic_protocol_context
) {
245 if (rtsp_st
->dynamic_handler
->close
)
246 rtsp_st
->dynamic_handler
->close(
247 rtsp_st
->dynamic_protocol_context
);
248 av_free(rtsp_st
->dynamic_protocol_context
);
250 rtsp_st
->dynamic_protocol_context
= NULL
;
251 rtsp_st
->dynamic_handler
= NULL
;
256 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
257 static int sdp_parse_rtpmap(AVFormatContext
*s
,
258 AVStream
*st
, RTSPStream
*rtsp_st
,
259 int payload_type
, const char *p
)
261 AVCodecParameters
*par
= st
->codecpar
;
264 const AVCodecDescriptor
*desc
;
267 /* See if we can handle this kind of payload.
268 * The space should normally not be there but some Real streams or
269 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
270 * have a trailing space. */
271 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
272 if (payload_type
< RTP_PT_PRIVATE
) {
273 /* We are in a standard case
274 * (from http://www.iana.org/assignments/rtp-parameters). */
275 par
->codec_id
= ff_rtp_codec_id(buf
, par
->codec_type
);
278 if (par
->codec_id
== AV_CODEC_ID_NONE
) {
279 const RTPDynamicProtocolHandler
*handler
=
280 ff_rtp_handler_find_by_name(buf
, par
->codec_type
);
281 init_rtp_handler(handler
, rtsp_st
, st
);
282 /* If no dynamic handler was found, check with the list of standard
283 * allocated types, if such a stream for some reason happens to
284 * use a private payload type. This isn't handled in rtpdec.c, since
285 * the format name from the rtpmap line never is passed into rtpdec. */
286 if (!rtsp_st
->dynamic_handler
)
287 par
->codec_id
= ff_rtp_codec_id(buf
, par
->codec_type
);
290 desc
= avcodec_descriptor_get(par
->codec_id
);
291 if (desc
&& desc
->name
)
296 get_word_sep(buf
, sizeof(buf
), "/", &p
);
298 switch (par
->codec_type
) {
299 case AVMEDIA_TYPE_AUDIO
:
300 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
301 par
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
302 par
->channels
= RTSP_DEFAULT_NB_AUDIO_CHANNELS
;
304 par
->sample_rate
= i
;
305 avpriv_set_pts_info(st
, 32, 1, par
->sample_rate
);
306 get_word_sep(buf
, sizeof(buf
), "/", &p
);
311 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
313 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
316 case AVMEDIA_TYPE_VIDEO
:
317 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
319 avpriv_set_pts_info(st
, 32, 1, i
);
324 finalize_rtp_handler_init(s
, rtsp_st
, st
);
328 /* parse the attribute line from the fmtp a line of an sdp response. This
329 * is broken out as a function because it is used in rtp_h264.c, which is
331 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
332 char *value
, int value_size
)
334 *p
+= strspn(*p
, SPACE_CHARS
);
336 get_word_sep(attr
, attr_size
, "=", p
);
339 get_word_sep(value
, value_size
, ";", p
);
347 typedef struct SDPParseState
{
349 struct sockaddr_storage default_ip
;
351 int skip_media
; ///< set if an unknown m= line occurs
352 int nb_default_include_source_addrs
; /**< Number of source-specific multicast include source IP address (from SDP content) */
353 struct RTSPSource
**default_include_source_addrs
; /**< Source-specific multicast include source IP address (from SDP content) */
354 int nb_default_exclude_source_addrs
; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
355 struct RTSPSource
**default_exclude_source_addrs
; /**< Source-specific multicast exclude source IP address (from SDP content) */
358 char delayed_fmtp
[2048];
361 static void copy_default_source_addrs(struct RTSPSource
**addrs
, int count
,
362 struct RTSPSource
***dest
, int *dest_count
)
364 RTSPSource
*rtsp_src
, *rtsp_src2
;
366 for (i
= 0; i
< count
; i
++) {
368 rtsp_src2
= av_malloc(sizeof(*rtsp_src2
));
371 memcpy(rtsp_src2
, rtsp_src
, sizeof(*rtsp_src
));
372 dynarray_add(dest
, dest_count
, rtsp_src2
);
376 static void parse_fmtp(AVFormatContext
*s
, RTSPState
*rt
,
377 int payload_type
, const char *line
)
381 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
382 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
383 if (rtsp_st
->sdp_payload_type
== payload_type
&&
384 rtsp_st
->dynamic_handler
&&
385 rtsp_st
->dynamic_handler
->parse_sdp_a_line
) {
386 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, rtsp_st
->stream_index
,
387 rtsp_st
->dynamic_protocol_context
, line
);
392 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
393 int letter
, const char *buf
)
395 RTSPState
*rt
= s
->priv_data
;
396 char buf1
[64], st_type
[64];
398 enum AVMediaType codec_type
;
402 RTSPSource
*rtsp_src
;
403 struct sockaddr_storage sdp_ip
;
406 av_log(s
, AV_LOG_TRACE
, "sdp: %c='%s'\n", letter
, buf
);
409 if (s1
->skip_media
&& letter
!= 'm')
413 get_word(buf1
, sizeof(buf1
), &p
);
414 if (strcmp(buf1
, "IN") != 0)
416 get_word(buf1
, sizeof(buf1
), &p
);
417 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6"))
419 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
420 if (get_sockaddr(s
, buf1
, &sdp_ip
))
425 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
428 if (s
->nb_streams
== 0) {
429 s1
->default_ip
= sdp_ip
;
430 s1
->default_ttl
= ttl
;
432 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
433 rtsp_st
->sdp_ip
= sdp_ip
;
434 rtsp_st
->sdp_ttl
= ttl
;
438 av_dict_set(&s
->metadata
, "title", p
, 0);
441 if (s
->nb_streams
== 0) {
442 av_dict_set(&s
->metadata
, "comment", p
, 0);
451 codec_type
= AVMEDIA_TYPE_UNKNOWN
;
452 get_word(st_type
, sizeof(st_type
), &p
);
453 if (!strcmp(st_type
, "audio")) {
454 codec_type
= AVMEDIA_TYPE_AUDIO
;
455 } else if (!strcmp(st_type
, "video")) {
456 codec_type
= AVMEDIA_TYPE_VIDEO
;
457 } else if (!strcmp(st_type
, "application")) {
458 codec_type
= AVMEDIA_TYPE_DATA
;
459 } else if (!strcmp(st_type
, "text")) {
460 codec_type
= AVMEDIA_TYPE_SUBTITLE
;
462 if (codec_type
== AVMEDIA_TYPE_UNKNOWN
||
463 !(rt
->media_type_mask
& (1 << codec_type
)) ||
464 rt
->nb_rtsp_streams
>= s
->max_streams
469 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
472 rtsp_st
->stream_index
= -1;
473 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
475 rtsp_st
->sdp_ip
= s1
->default_ip
;
476 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
478 copy_default_source_addrs(s1
->default_include_source_addrs
,
479 s1
->nb_default_include_source_addrs
,
480 &rtsp_st
->include_source_addrs
,
481 &rtsp_st
->nb_include_source_addrs
);
482 copy_default_source_addrs(s1
->default_exclude_source_addrs
,
483 s1
->nb_default_exclude_source_addrs
,
484 &rtsp_st
->exclude_source_addrs
,
485 &rtsp_st
->nb_exclude_source_addrs
);
487 get_word(buf1
, sizeof(buf1
), &p
); /* port */
488 rtsp_st
->sdp_port
= atoi(buf1
);
490 get_word(buf1
, sizeof(buf1
), &p
); /* protocol */
491 if (!strcmp(buf1
, "udp"))
492 rt
->transport
= RTSP_TRANSPORT_RAW
;
493 else if (strstr(buf1
, "/AVPF") || strstr(buf1
, "/SAVPF"))
494 rtsp_st
->feedback
= 1;
496 /* XXX: handle list of formats */
497 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
498 rtsp_st
->sdp_payload_type
= atoi(buf1
);
500 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
501 /* no corresponding stream */
502 if (rt
->transport
== RTSP_TRANSPORT_RAW
) {
503 if (CONFIG_RTPDEC
&& !rt
->ts
)
504 rt
->ts
= avpriv_mpegts_parse_open(s
);
506 const RTPDynamicProtocolHandler
*handler
;
507 handler
= ff_rtp_handler_find_by_id(
508 rtsp_st
->sdp_payload_type
, AVMEDIA_TYPE_DATA
);
509 init_rtp_handler(handler
, rtsp_st
, NULL
);
510 finalize_rtp_handler_init(s
, rtsp_st
, NULL
);
512 } else if (rt
->server_type
== RTSP_SERVER_WMS
&&
513 codec_type
== AVMEDIA_TYPE_DATA
) {
514 /* RTX stream, a stream that carries all the other actual
515 * audio/video streams. Don't expose this to the callers. */
517 st
= avformat_new_stream(s
, NULL
);
520 st
->id
= rt
->nb_rtsp_streams
- 1;
521 rtsp_st
->stream_index
= st
->index
;
522 st
->codecpar
->codec_type
= codec_type
;
523 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
524 const RTPDynamicProtocolHandler
*handler
;
525 /* if standard payload type, we can find the codec right now */
526 ff_rtp_get_codec_info(st
->codecpar
, rtsp_st
->sdp_payload_type
);
527 if (st
->codecpar
->codec_type
== AVMEDIA_TYPE_AUDIO
&&
528 st
->codecpar
->sample_rate
> 0)
529 avpriv_set_pts_info(st
, 32, 1, st
->codecpar
->sample_rate
);
530 /* Even static payload types may need a custom depacketizer */
531 handler
= ff_rtp_handler_find_by_id(
532 rtsp_st
->sdp_payload_type
, st
->codecpar
->codec_type
);
533 init_rtp_handler(handler
, rtsp_st
, st
);
534 finalize_rtp_handler_init(s
, rtsp_st
, st
);
536 if (rt
->default_lang
[0])
537 av_dict_set(&st
->metadata
, "language", rt
->default_lang
, 0);
539 /* put a default control url */
540 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
541 sizeof(rtsp_st
->control_url
));
544 if (av_strstart(p
, "control:", &p
)) {
545 if (s
->nb_streams
== 0) {
546 if (!strncmp(p
, "rtsp://", 7))
547 av_strlcpy(rt
->control_uri
, p
,
548 sizeof(rt
->control_uri
));
551 /* get the control url */
552 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
554 /* XXX: may need to add full url resolution */
555 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
557 if (proto
[0] == '\0') {
558 /* relative control URL */
559 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
560 av_strlcat(rtsp_st
->control_url
, "/",
561 sizeof(rtsp_st
->control_url
));
562 av_strlcat(rtsp_st
->control_url
, p
,
563 sizeof(rtsp_st
->control_url
));
565 av_strlcpy(rtsp_st
->control_url
, p
,
566 sizeof(rtsp_st
->control_url
));
568 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
569 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
570 get_word(buf1
, sizeof(buf1
), &p
);
571 payload_type
= atoi(buf1
);
572 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
573 if (rtsp_st
->stream_index
>= 0) {
574 st
= s
->streams
[rtsp_st
->stream_index
];
575 sdp_parse_rtpmap(s
, st
, rtsp_st
, payload_type
, p
);
579 parse_fmtp(s
, rt
, payload_type
, s1
->delayed_fmtp
);
581 } else if (av_strstart(p
, "fmtp:", &p
) ||
582 av_strstart(p
, "framesize:", &p
)) {
583 // let dynamic protocol handlers have a stab at the line.
584 get_word(buf1
, sizeof(buf1
), &p
);
585 payload_type
= atoi(buf1
);
586 if (s1
->seen_rtpmap
) {
587 parse_fmtp(s
, rt
, payload_type
, buf
);
590 av_strlcpy(s1
->delayed_fmtp
, buf
, sizeof(s1
->delayed_fmtp
));
592 } else if (av_strstart(p
, "ssrc:", &p
) && s
->nb_streams
> 0) {
593 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
594 get_word(buf1
, sizeof(buf1
), &p
);
595 rtsp_st
->ssrc
= strtoll(buf1
, NULL
, 10);
596 } else if (av_strstart(p
, "range:", &p
)) {
599 // this is so that seeking on a streamed file can work.
600 rtsp_parse_range_npt(p
, &start
, &end
);
601 s
->start_time
= start
;
602 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
603 s
->duration
= (end
== AV_NOPTS_VALUE
) ?
604 AV_NOPTS_VALUE
: end
- start
;
605 } else if (av_strstart(p
, "lang:", &p
)) {
606 if (s
->nb_streams
> 0) {
607 get_word(buf1
, sizeof(buf1
), &p
);
608 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
609 if (rtsp_st
->stream_index
>= 0) {
610 st
= s
->streams
[rtsp_st
->stream_index
];
611 av_dict_set(&st
->metadata
, "language", buf1
, 0);
614 get_word(rt
->default_lang
, sizeof(rt
->default_lang
), &p
);
615 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
617 rt
->transport
= RTSP_TRANSPORT_RDT
;
618 } else if (av_strstart(p
, "SampleRate:integer;", &p
) &&
620 st
= s
->streams
[s
->nb_streams
- 1];
621 st
->codecpar
->sample_rate
= atoi(p
);
622 } else if (av_strstart(p
, "crypto:", &p
) && s
->nb_streams
> 0) {
624 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
625 get_word(buf1
, sizeof(buf1
), &p
); // ignore tag
626 get_word(rtsp_st
->crypto_suite
, sizeof(rtsp_st
->crypto_suite
), &p
);
627 p
+= strspn(p
, SPACE_CHARS
);
628 if (av_strstart(p
, "inline:", &p
))
629 get_word(rtsp_st
->crypto_params
, sizeof(rtsp_st
->crypto_params
), &p
);
630 } else if (av_strstart(p
, "source-filter:", &p
)) {
632 get_word(buf1
, sizeof(buf1
), &p
);
633 if (strcmp(buf1
, "incl") && strcmp(buf1
, "excl"))
635 exclude
= !strcmp(buf1
, "excl");
637 get_word(buf1
, sizeof(buf1
), &p
);
638 if (strcmp(buf1
, "IN") != 0)
640 get_word(buf1
, sizeof(buf1
), &p
);
641 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6") && strcmp(buf1
, "*"))
643 // not checking that the destination address actually matches or is wildcard
644 get_word(buf1
, sizeof(buf1
), &p
);
647 rtsp_src
= av_mallocz(sizeof(*rtsp_src
));
650 get_word(rtsp_src
->addr
, sizeof(rtsp_src
->addr
), &p
);
652 if (s
->nb_streams
== 0) {
653 dynarray_add(&s1
->default_exclude_source_addrs
, &s1
->nb_default_exclude_source_addrs
, rtsp_src
);
655 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
656 dynarray_add(&rtsp_st
->exclude_source_addrs
, &rtsp_st
->nb_exclude_source_addrs
, rtsp_src
);
659 if (s
->nb_streams
== 0) {
660 dynarray_add(&s1
->default_include_source_addrs
, &s1
->nb_default_include_source_addrs
, rtsp_src
);
662 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
663 dynarray_add(&rtsp_st
->include_source_addrs
, &rtsp_st
->nb_include_source_addrs
, rtsp_src
);
668 if (rt
->server_type
== RTSP_SERVER_WMS
)
669 ff_wms_parse_sdp_a_line(s
, p
);
670 if (s
->nb_streams
> 0) {
671 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
673 if (rt
->server_type
== RTSP_SERVER_REAL
)
674 ff_real_parse_sdp_a_line(s
, rtsp_st
->stream_index
, p
);
676 if (rtsp_st
->dynamic_handler
&&
677 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
678 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
679 rtsp_st
->stream_index
,
680 rtsp_st
->dynamic_protocol_context
, buf
);
687 int ff_sdp_parse(AVFormatContext
*s
, const char *content
)
691 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
692 * contain long SDP lines containing complete ASF Headers (several
693 * kB) or arrays of MDPR (RM stream descriptor) headers plus
694 * "rulebooks" describing their properties. Therefore, the SDP line
697 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
698 * in rtpdec_xiph.c. */
700 SDPParseState sdp_parse_state
= { { 0 } }, *s1
= &sdp_parse_state
;
704 p
+= strspn(p
, SPACE_CHARS
);
712 /* get the content */
714 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
715 if ((q
- buf
) < sizeof(buf
) - 1)
720 sdp_parse_line(s
, s1
, letter
, buf
);
722 while (*p
!= '\n' && *p
!= '\0')
728 for (i
= 0; i
< s1
->nb_default_include_source_addrs
; i
++)
729 av_freep(&s1
->default_include_source_addrs
[i
]);
730 av_freep(&s1
->default_include_source_addrs
);
731 for (i
= 0; i
< s1
->nb_default_exclude_source_addrs
; i
++)
732 av_freep(&s1
->default_exclude_source_addrs
[i
]);
733 av_freep(&s1
->default_exclude_source_addrs
);
737 #endif /* CONFIG_RTPDEC */
739 void ff_rtsp_undo_setup(AVFormatContext
*s
, int send_packets
)
741 RTSPState
*rt
= s
->priv_data
;
744 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
745 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
748 if (rtsp_st
->transport_priv
) {
750 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
751 av_write_trailer(rtpctx
);
752 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
753 if (CONFIG_RTSP_MUXER
&& rtpctx
->pb
&& send_packets
)
754 ff_rtsp_tcp_write_packet(s
, rtsp_st
);
755 ffio_free_dyn_buf(&rtpctx
->pb
);
757 avio_closep(&rtpctx
->pb
);
759 avformat_free_context(rtpctx
);
760 } else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RDT
)
761 ff_rdt_parse_close(rtsp_st
->transport_priv
);
762 else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
763 ff_rtp_parse_close(rtsp_st
->transport_priv
);
765 rtsp_st
->transport_priv
= NULL
;
766 ffurl_closep(&rtsp_st
->rtp_handle
);
770 /* close and free RTSP streams */
771 void ff_rtsp_close_streams(AVFormatContext
*s
)
773 RTSPState
*rt
= s
->priv_data
;
777 ff_rtsp_undo_setup(s
, 0);
778 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
779 rtsp_st
= rt
->rtsp_streams
[i
];
781 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
) {
782 if (rtsp_st
->dynamic_handler
->close
)
783 rtsp_st
->dynamic_handler
->close(
784 rtsp_st
->dynamic_protocol_context
);
785 av_free(rtsp_st
->dynamic_protocol_context
);
787 for (j
= 0; j
< rtsp_st
->nb_include_source_addrs
; j
++)
788 av_freep(&rtsp_st
->include_source_addrs
[j
]);
789 av_freep(&rtsp_st
->include_source_addrs
);
790 for (j
= 0; j
< rtsp_st
->nb_exclude_source_addrs
; j
++)
791 av_freep(&rtsp_st
->exclude_source_addrs
[j
]);
792 av_freep(&rtsp_st
->exclude_source_addrs
);
797 av_freep(&rt
->rtsp_streams
);
799 avformat_close_input(&rt
->asf_ctx
);
801 if (CONFIG_RTPDEC
&& rt
->ts
)
802 avpriv_mpegts_parse_close(rt
->ts
);
804 av_freep(&rt
->recvbuf
);
807 int ff_rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
809 RTSPState
*rt
= s
->priv_data
;
811 int reordering_queue_size
= rt
->reordering_queue_size
;
812 if (reordering_queue_size
< 0) {
813 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
|| !s
->max_delay
)
814 reordering_queue_size
= 0;
816 reordering_queue_size
= RTP_REORDER_QUEUE_DEFAULT_SIZE
;
819 /* open the RTP context */
820 if (rtsp_st
->stream_index
>= 0)
821 st
= s
->streams
[rtsp_st
->stream_index
];
823 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
825 if (CONFIG_RTSP_MUXER
&& s
->oformat
&& st
) {
826 int ret
= ff_rtp_chain_mux_open((AVFormatContext
**)&rtsp_st
->transport_priv
,
827 s
, st
, rtsp_st
->rtp_handle
,
828 RTSP_TCP_MAX_PACKET_SIZE
,
829 rtsp_st
->stream_index
);
830 /* Ownership of rtp_handle is passed to the rtp mux context */
831 rtsp_st
->rtp_handle
= NULL
;
834 st
->time_base
= ((AVFormatContext
*)rtsp_st
->transport_priv
)->streams
[0]->time_base
;
835 } else if (rt
->transport
== RTSP_TRANSPORT_RAW
) {
836 return 0; // Don't need to open any parser here
837 } else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RDT
&& st
)
838 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
839 rtsp_st
->dynamic_protocol_context
,
840 rtsp_st
->dynamic_handler
);
841 else if (CONFIG_RTPDEC
)
842 rtsp_st
->transport_priv
= ff_rtp_parse_open(s
, st
,
843 rtsp_st
->sdp_payload_type
,
844 reordering_queue_size
);
846 if (!rtsp_st
->transport_priv
) {
847 return AVERROR(ENOMEM
);
848 } else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RTP
&&
850 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
851 rtpctx
->ssrc
= rtsp_st
->ssrc
;
852 if (rtsp_st
->dynamic_handler
) {
853 ff_rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
854 rtsp_st
->dynamic_protocol_context
,
855 rtsp_st
->dynamic_handler
);
857 if (rtsp_st
->crypto_suite
[0])
858 ff_rtp_parse_set_crypto(rtsp_st
->transport_priv
,
859 rtsp_st
->crypto_suite
,
860 rtsp_st
->crypto_params
);
866 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
867 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
874 q
+= strspn(q
, SPACE_CHARS
);
875 v
= strtol(q
, &p
, 10);
879 v
= strtol(p
, &p
, 10);
888 /* XXX: only one transport specification is parsed */
889 static void rtsp_parse_transport(AVFormatContext
*s
,
890 RTSPMessageHeader
*reply
, const char *p
)
892 char transport_protocol
[16];
894 char lower_transport
[16];
896 RTSPTransportField
*th
;
899 reply
->nb_transports
= 0;
902 p
+= strspn(p
, SPACE_CHARS
);
906 th
= &reply
->transports
[reply
->nb_transports
];
908 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
910 if (!av_strcasecmp (transport_protocol
, "rtp")) {
911 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
912 lower_transport
[0] = '\0';
913 /* rtp/avp/<protocol> */
915 get_word_sep(lower_transport
, sizeof(lower_transport
),
918 th
->transport
= RTSP_TRANSPORT_RTP
;
919 } else if (!av_strcasecmp (transport_protocol
, "x-pn-tng") ||
920 !av_strcasecmp (transport_protocol
, "x-real-rdt")) {
921 /* x-pn-tng/<protocol> */
922 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
924 th
->transport
= RTSP_TRANSPORT_RDT
;
925 } else if (!av_strcasecmp(transport_protocol
, "raw")) {
926 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
927 lower_transport
[0] = '\0';
928 /* raw/raw/<protocol> */
930 get_word_sep(lower_transport
, sizeof(lower_transport
),
933 th
->transport
= RTSP_TRANSPORT_RAW
;
937 if (!av_strcasecmp(lower_transport
, "TCP"))
938 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
940 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
944 /* get each parameter */
945 while (*p
!= '\0' && *p
!= ',') {
946 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
947 if (!strcmp(parameter
, "port")) {
950 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
952 } else if (!strcmp(parameter
, "client_port")) {
955 rtsp_parse_range(&th
->client_port_min
,
956 &th
->client_port_max
, &p
);
958 } else if (!strcmp(parameter
, "server_port")) {
961 rtsp_parse_range(&th
->server_port_min
,
962 &th
->server_port_max
, &p
);
964 } else if (!strcmp(parameter
, "interleaved")) {
967 rtsp_parse_range(&th
->interleaved_min
,
968 &th
->interleaved_max
, &p
);
970 } else if (!strcmp(parameter
, "multicast")) {
971 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
972 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
973 } else if (!strcmp(parameter
, "ttl")) {
977 th
->ttl
= strtol(p
, &end
, 10);
980 } else if (!strcmp(parameter
, "destination")) {
983 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
984 get_sockaddr(s
, buf
, &th
->destination
);
986 } else if (!strcmp(parameter
, "source")) {
989 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
990 av_strlcpy(th
->source
, buf
, sizeof(th
->source
));
992 } else if (!strcmp(parameter
, "mode")) {
995 get_word_sep(buf
, sizeof(buf
), ";, ", &p
);
996 if (!strcmp(buf
, "record") ||
997 !strcmp(buf
, "receive"))
1002 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
1010 reply
->nb_transports
++;
1011 if (reply
->nb_transports
>= RTSP_MAX_TRANSPORTS
)
1016 static void handle_rtp_info(RTSPState
*rt
, const char *url
,
1017 uint32_t seq
, uint32_t rtptime
)
1020 if (!rtptime
|| !url
[0])
1022 if (rt
->transport
!= RTSP_TRANSPORT_RTP
)
1024 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1025 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
1026 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1029 if (!strcmp(rtsp_st
->control_url
, url
)) {
1030 rtpctx
->base_timestamp
= rtptime
;
1036 static void rtsp_parse_rtp_info(RTSPState
*rt
, const char *p
)
1039 char key
[20], value
[1024], url
[1024] = "";
1040 uint32_t seq
= 0, rtptime
= 0;
1043 p
+= strspn(p
, SPACE_CHARS
);
1046 get_word_sep(key
, sizeof(key
), "=", &p
);
1050 get_word_sep(value
, sizeof(value
), ";, ", &p
);
1052 if (!strcmp(key
, "url"))
1053 av_strlcpy(url
, value
, sizeof(url
));
1054 else if (!strcmp(key
, "seq"))
1055 seq
= strtoul(value
, NULL
, 10);
1056 else if (!strcmp(key
, "rtptime"))
1057 rtptime
= strtoul(value
, NULL
, 10);
1059 handle_rtp_info(rt
, url
, seq
, rtptime
);
1068 handle_rtp_info(rt
, url
, seq
, rtptime
);
1071 void ff_rtsp_parse_line(AVFormatContext
*s
,
1072 RTSPMessageHeader
*reply
, const char *buf
,
1073 RTSPState
*rt
, const char *method
)
1077 /* NOTE: we do case independent match for broken servers */
1079 if (av_stristart(p
, "Session:", &p
)) {
1081 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
1082 if (av_stristart(p
, ";timeout=", &p
) &&
1083 (t
= strtol(p
, NULL
, 10)) > 0) {
1086 } else if (av_stristart(p
, "Content-Length:", &p
)) {
1087 reply
->content_length
= strtol(p
, NULL
, 10);
1088 } else if (av_stristart(p
, "Transport:", &p
)) {
1089 rtsp_parse_transport(s
, reply
, p
);
1090 } else if (av_stristart(p
, "CSeq:", &p
)) {
1091 reply
->seq
= strtol(p
, NULL
, 10);
1092 } else if (av_stristart(p
, "Range:", &p
)) {
1093 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
1094 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
1095 p
+= strspn(p
, SPACE_CHARS
);
1096 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
1097 } else if (av_stristart(p
, "Server:", &p
)) {
1098 p
+= strspn(p
, SPACE_CHARS
);
1099 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
1100 } else if (av_stristart(p
, "Notice:", &p
) ||
1101 av_stristart(p
, "X-Notice:", &p
)) {
1102 reply
->notice
= strtol(p
, NULL
, 10);
1103 } else if (av_stristart(p
, "Location:", &p
)) {
1104 p
+= strspn(p
, SPACE_CHARS
);
1105 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
1106 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && rt
) {
1107 p
+= strspn(p
, SPACE_CHARS
);
1108 ff_http_auth_handle_header(&rt
->auth_state
, "WWW-Authenticate", p
);
1109 } else if (av_stristart(p
, "Authentication-Info:", &p
) && rt
) {
1110 p
+= strspn(p
, SPACE_CHARS
);
1111 ff_http_auth_handle_header(&rt
->auth_state
, "Authentication-Info", p
);
1112 } else if (av_stristart(p
, "Content-Base:", &p
) && rt
) {
1113 p
+= strspn(p
, SPACE_CHARS
);
1114 if (method
&& !strcmp(method
, "DESCRIBE"))
1115 av_strlcpy(rt
->control_uri
, p
, sizeof(rt
->control_uri
));
1116 } else if (av_stristart(p
, "RTP-Info:", &p
) && rt
) {
1117 p
+= strspn(p
, SPACE_CHARS
);
1118 if (method
&& !strcmp(method
, "PLAY"))
1119 rtsp_parse_rtp_info(rt
, p
);
1120 } else if (av_stristart(p
, "Public:", &p
) && rt
) {
1121 if (strstr(p
, "GET_PARAMETER") &&
1122 method
&& !strcmp(method
, "OPTIONS"))
1123 rt
->get_parameter_supported
= 1;
1124 } else if (av_stristart(p
, "x-Accept-Dynamic-Rate:", &p
) && rt
) {
1125 p
+= strspn(p
, SPACE_CHARS
);
1126 rt
->accept_dynamic_rate
= atoi(p
);
1127 } else if (av_stristart(p
, "Content-Type:", &p
)) {
1128 p
+= strspn(p
, SPACE_CHARS
);
1129 av_strlcpy(reply
->content_type
, p
, sizeof(reply
->content_type
));
1133 /* skip a RTP/TCP interleaved packet */
1134 void ff_rtsp_skip_packet(AVFormatContext
*s
)
1136 RTSPState
*rt
= s
->priv_data
;
1140 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, 3);
1143 len
= AV_RB16(buf
+ 1);
1145 av_log(s
, AV_LOG_TRACE
, "skipping RTP packet len=%d\n", len
);
1150 if (len1
> sizeof(buf
))
1152 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, len1
);
1159 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
1160 unsigned char **content_ptr
,
1161 int return_on_interleaved_data
, const char *method
)
1163 RTSPState
*rt
= s
->priv_data
;
1164 char buf
[4096], buf1
[1024], *q
;
1167 int ret
, content_length
, line_count
= 0, request
= 0;
1168 unsigned char *content
= NULL
;
1174 memset(reply
, 0, sizeof(*reply
));
1176 /* parse reply (XXX: use buffers) */
1177 rt
->last_reply
[0] = '\0';
1181 ret
= ffurl_read_complete(rt
->rtsp_hd
, &ch
, 1);
1182 av_log(s
, AV_LOG_TRACE
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
1187 if (ch
== '$' && q
== buf
) {
1188 if (return_on_interleaved_data
) {
1191 ff_rtsp_skip_packet(s
);
1192 } else if (ch
!= '\r') {
1193 if ((q
- buf
) < sizeof(buf
) - 1)
1199 av_log(s
, AV_LOG_TRACE
, "line='%s'\n", buf
);
1201 /* test if last line */
1205 if (line_count
== 0) {
1206 /* get reply code */
1207 get_word(buf1
, sizeof(buf1
), &p
);
1208 if (!strncmp(buf1
, "RTSP/", 5)) {
1209 get_word(buf1
, sizeof(buf1
), &p
);
1210 reply
->status_code
= atoi(buf1
);
1211 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
1213 av_strlcpy(reply
->reason
, buf1
, sizeof(reply
->reason
)); // method
1214 get_word(buf1
, sizeof(buf1
), &p
); // object
1218 ff_rtsp_parse_line(s
, reply
, p
, rt
, method
);
1219 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
1220 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
1225 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0' && !request
)
1226 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
1228 content_length
= reply
->content_length
;
1229 if (content_length
> 0) {
1230 /* leave some room for a trailing '\0' (useful for simple parsing) */
1231 content
= av_malloc(content_length
+ 1);
1233 return AVERROR(ENOMEM
);
1234 ffurl_read_complete(rt
->rtsp_hd
, content
, content_length
);
1235 content
[content_length
] = '\0';
1238 *content_ptr
= content
;
1244 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1245 const char* ptr
= buf
;
1247 if (!strcmp(reply
->reason
, "OPTIONS")) {
1248 snprintf(buf
, sizeof(buf
), "RTSP/1.0 200 OK\r\n");
1250 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", reply
->seq
);
1251 if (reply
->session_id
[0])
1252 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n",
1255 snprintf(buf
, sizeof(buf
), "RTSP/1.0 501 Not Implemented\r\n");
1257 av_strlcat(buf
, "\r\n", sizeof(buf
));
1259 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1260 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1263 ffurl_write(rt
->rtsp_hd_out
, ptr
, strlen(ptr
));
1265 rt
->last_cmd_time
= av_gettime_relative();
1266 /* Even if the request from the server had data, it is not the data
1267 * that the caller wants or expects. The memory could also be leaked
1268 * if the actual following reply has content data. */
1270 av_freep(content_ptr
);
1271 /* If method is set, this is called from ff_rtsp_send_cmd,
1272 * where a reply to exactly this request is awaited. For
1273 * callers from within packet receiving, we just want to
1274 * return to the caller and go back to receiving packets. */
1280 if (rt
->seq
!= reply
->seq
) {
1281 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
1282 rt
->seq
, reply
->seq
);
1286 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
1287 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
1288 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
1289 rt
->state
= RTSP_STATE_IDLE
;
1290 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
1291 return AVERROR(EIO
); /* data or server error */
1292 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
1293 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
1294 return AVERROR(EPERM
);
1300 * Send a command to the RTSP server without waiting for the reply.
1302 * @param s RTSP (de)muxer context
1303 * @param method the method for the request
1304 * @param url the target url for the request
1305 * @param headers extra header lines to include in the request
1306 * @param send_content if non-null, the data to send as request body content
1307 * @param send_content_length the length of the send_content data, or 0 if
1308 * send_content is null
1310 * @return zero if success, nonzero otherwise
1312 static int rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
1313 const char *method
, const char *url
,
1314 const char *headers
,
1315 const unsigned char *send_content
,
1316 int send_content_length
)
1318 RTSPState
*rt
= s
->priv_data
;
1319 char buf
[4096], *out_buf
;
1320 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1322 if (!rt
->rtsp_hd_out
)
1323 return AVERROR(ENOTCONN
);
1325 /* Add in RTSP headers */
1328 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
1330 av_strlcat(buf
, headers
, sizeof(buf
));
1331 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
1332 av_strlcatf(buf
, sizeof(buf
), "User-Agent: %s\r\n", rt
->user_agent
);
1333 if (rt
->session_id
[0] != '\0' && (!headers
||
1334 !strstr(headers
, "\nIf-Match:"))) {
1335 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
1338 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
1339 rt
->auth
, url
, method
);
1341 av_strlcat(buf
, str
, sizeof(buf
));
1344 if (send_content_length
> 0 && send_content
)
1345 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
1346 av_strlcat(buf
, "\r\n", sizeof(buf
));
1348 /* base64 encode rtsp if tunneling */
1349 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1350 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1351 out_buf
= base64buf
;
1354 av_log(s
, AV_LOG_TRACE
, "Sending:\n%s--\n", buf
);
1356 ffurl_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
1357 if (send_content_length
> 0 && send_content
) {
1358 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1359 avpriv_report_missing_feature(s
, "Tunneling of RTSP requests with content data");
1360 return AVERROR_PATCHWELCOME
;
1362 ffurl_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
1364 rt
->last_cmd_time
= av_gettime_relative();
1369 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
1370 const char *url
, const char *headers
)
1372 return rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
1375 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
1376 const char *headers
, RTSPMessageHeader
*reply
,
1377 unsigned char **content_ptr
)
1379 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
1380 content_ptr
, NULL
, 0);
1383 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
1384 const char *method
, const char *url
,
1386 RTSPMessageHeader
*reply
,
1387 unsigned char **content_ptr
,
1388 const unsigned char *send_content
,
1389 int send_content_length
)
1391 RTSPState
*rt
= s
->priv_data
;
1392 HTTPAuthType cur_auth_type
;
1393 int ret
, attempts
= 0;
1396 cur_auth_type
= rt
->auth_state
.auth_type
;
1397 if ((ret
= rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
1399 send_content_length
)) < 0)
1402 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0, method
) ) < 0)
1406 if (reply
->status_code
== 401 &&
1407 (cur_auth_type
== HTTP_AUTH_NONE
|| rt
->auth_state
.stale
) &&
1408 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
&& attempts
< 2)
1411 if (reply
->status_code
> 400){
1412 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
1416 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
1422 int ff_rtsp_make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
1423 int lower_transport
, const char *real_challenge
)
1425 RTSPState
*rt
= s
->priv_data
;
1426 int rtx
= 0, j
, i
, err
, interleave
= 0, port_off
;
1427 RTSPStream
*rtsp_st
;
1428 RTSPMessageHeader reply1
, *reply
= &reply1
;
1430 const char *trans_pref
;
1432 memset(&reply1
, 0, sizeof(reply1
));
1434 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
1435 trans_pref
= "x-pn-tng";
1436 else if (rt
->transport
== RTSP_TRANSPORT_RAW
)
1437 trans_pref
= "RAW/RAW";
1439 trans_pref
= "RTP/AVP";
1441 /* default timeout: 1 minute */
1444 /* Choose a random starting offset within the first half of the
1445 * port range, to allow for a number of ports to try even if the offset
1446 * happens to be at the end of the random range. */
1447 port_off
= av_get_random_seed() % ((rt
->rtp_port_max
- rt
->rtp_port_min
)/2);
1448 /* even random offset */
1449 port_off
-= port_off
& 0x01;
1451 for (j
= rt
->rtp_port_min
+ port_off
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1452 char transport
[2048];
1455 * WMS serves all UDP data over a single connection, the RTX, which
1456 * isn't necessarily the first in the SDP but has to be the first
1457 * to be set up, else the second/third SETUP will fail with a 461.
1459 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1460 rt
->server_type
== RTSP_SERVER_WMS
) {
1463 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1464 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1466 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1470 if (rtx
== rt
->nb_rtsp_streams
)
1471 return -1; /* no RTX found */
1472 rtsp_st
= rt
->rtsp_streams
[rtx
];
1474 rtsp_st
= rt
->rtsp_streams
[i
> rtx
? i
: i
- 1];
1476 rtsp_st
= rt
->rtsp_streams
[i
];
1479 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1482 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1483 port
= reply
->transports
[0].client_port_min
;
1487 /* first try in specified port range */
1488 while (j
<= rt
->rtp_port_max
) {
1489 AVDictionary
*opts
= map_to_opts(rt
);
1491 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1492 "?localport=%d", j
);
1493 /* we will use two ports per rtp stream (rtp and rtcp) */
1495 err
= ffurl_open_whitelist(&rtsp_st
->rtp_handle
, buf
, AVIO_FLAG_READ_WRITE
,
1496 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
1498 av_dict_free(&opts
);
1503 av_log(s
, AV_LOG_ERROR
, "Unable to open an input RTP port\n");
1508 port
= ff_rtp_get_local_rtp_port(rtsp_st
->rtp_handle
);
1510 snprintf(transport
, sizeof(transport
) - 1,
1511 "%s/UDP;", trans_pref
);
1512 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1513 av_strlcat(transport
, "unicast;", sizeof(transport
));
1514 av_strlcatf(transport
, sizeof(transport
),
1515 "client_port=%d", port
);
1516 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1517 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1518 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1522 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1523 /* For WMS streams, the application streams are only used for
1524 * UDP. When trying to set it up for TCP streams, the server
1525 * will return an error. Therefore, we skip those streams. */
1526 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1527 (rtsp_st
->stream_index
< 0 ||
1528 s
->streams
[rtsp_st
->stream_index
]->codecpar
->codec_type
==
1531 snprintf(transport
, sizeof(transport
) - 1,
1532 "%s/TCP;", trans_pref
);
1533 if (rt
->transport
!= RTSP_TRANSPORT_RDT
)
1534 av_strlcat(transport
, "unicast;", sizeof(transport
));
1535 av_strlcatf(transport
, sizeof(transport
),
1536 "interleaved=%d-%d",
1537 interleave
, interleave
+ 1);
1541 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1542 snprintf(transport
, sizeof(transport
) - 1,
1543 "%s/UDP;multicast", trans_pref
);
1545 err
= AVERROR(EINVAL
);
1546 goto fail
; // transport would be uninitialized
1550 av_strlcat(transport
, ";mode=record", sizeof(transport
));
1551 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1552 rt
->server_type
== RTSP_SERVER_WMS
)
1553 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1554 snprintf(cmd
, sizeof(cmd
),
1555 "Transport: %s\r\n",
1557 if (rt
->accept_dynamic_rate
)
1558 av_strlcat(cmd
, "x-Dynamic-Rate: 0\r\n", sizeof(cmd
));
1559 if (CONFIG_RTPDEC
&& i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
) {
1560 char real_res
[41], real_csum
[9];
1561 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1563 av_strlcatf(cmd
, sizeof(cmd
),
1565 "RealChallenge2: %s, sd=%s\r\n",
1566 rt
->session_id
, real_res
, real_csum
);
1568 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1569 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1572 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1573 reply
->nb_transports
!= 1) {
1574 err
= ff_rtsp_averror(reply
->status_code
, AVERROR_INVALIDDATA
);
1578 /* XXX: same protocol for all streams is required */
1580 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1581 reply
->transports
[0].transport
!= rt
->transport
) {
1582 err
= AVERROR_INVALIDDATA
;
1586 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1587 rt
->transport
= reply
->transports
[0].transport
;
1590 /* Fail if the server responded with another lower transport mode
1591 * than what we requested. */
1592 if (reply
->transports
[0].lower_transport
!= lower_transport
) {
1593 av_log(s
, AV_LOG_ERROR
, "Nonmatching transport in server reply\n");
1594 err
= AVERROR_INVALIDDATA
;
1598 switch(reply
->transports
[0].lower_transport
) {
1599 case RTSP_LOWER_TRANSPORT_TCP
:
1600 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1601 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1604 case RTSP_LOWER_TRANSPORT_UDP
: {
1605 char url
[1024], options
[30] = "";
1606 const char *peer
= host
;
1608 if (rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
)
1609 av_strlcpy(options
, "?connect=1", sizeof(options
));
1610 /* Use source address if specified */
1611 if (reply
->transports
[0].source
[0])
1612 peer
= reply
->transports
[0].source
;
1613 ff_url_join(url
, sizeof(url
), "rtp", NULL
, peer
,
1614 reply
->transports
[0].server_port_min
, "%s", options
);
1615 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1616 ff_rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1617 err
= AVERROR_INVALIDDATA
;
1622 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1623 char url
[1024], namebuf
[50], optbuf
[20] = "";
1624 struct sockaddr_storage addr
;
1626 AVDictionary
*opts
= map_to_opts(rt
);
1628 if (reply
->transports
[0].destination
.ss_family
) {
1629 addr
= reply
->transports
[0].destination
;
1630 port
= reply
->transports
[0].port_min
;
1631 ttl
= reply
->transports
[0].ttl
;
1633 addr
= rtsp_st
->sdp_ip
;
1634 port
= rtsp_st
->sdp_port
;
1635 ttl
= rtsp_st
->sdp_ttl
;
1638 snprintf(optbuf
, sizeof(optbuf
), "?ttl=%d", ttl
);
1639 getnameinfo((struct sockaddr
*) &addr
, sizeof(addr
),
1640 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1641 ff_url_join(url
, sizeof(url
), "rtp", NULL
, namebuf
,
1642 port
, "%s", optbuf
);
1643 err
= ffurl_open_whitelist(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
,
1644 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
1645 av_dict_free(&opts
);
1648 err
= AVERROR_INVALIDDATA
;
1655 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
1659 if (rt
->nb_rtsp_streams
&& reply
->timeout
> 0)
1660 rt
->timeout
= reply
->timeout
;
1662 if (rt
->server_type
== RTSP_SERVER_REAL
)
1663 rt
->need_subscription
= 1;
1668 ff_rtsp_undo_setup(s
, 0);
1672 void ff_rtsp_close_connections(AVFormatContext
*s
)
1674 RTSPState
*rt
= s
->priv_data
;
1675 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
)
1676 ffurl_closep(&rt
->rtsp_hd_out
);
1677 rt
->rtsp_hd_out
= NULL
;
1678 ffurl_closep(&rt
->rtsp_hd
);
1681 int ff_rtsp_connect(AVFormatContext
*s
)
1683 RTSPState
*rt
= s
->priv_data
;
1684 char proto
[128], host
[1024], path
[1024];
1685 char tcpname
[1024], cmd
[2048], auth
[128];
1686 const char *lower_rtsp_proto
= "tcp";
1687 int port
, err
, tcp_fd
;
1688 RTSPMessageHeader reply1
, *reply
= &reply1
;
1689 int lower_transport_mask
= 0;
1690 int default_port
= RTSP_DEFAULT_PORT
;
1691 int https_tunnel
= 0;
1692 char real_challenge
[64] = "";
1693 struct sockaddr_storage peer
;
1694 socklen_t peer_len
= sizeof(peer
);
1696 if (rt
->rtp_port_max
< rt
->rtp_port_min
) {
1697 av_log(s
, AV_LOG_ERROR
, "Invalid UDP port range, max port %d less "
1698 "than min port %d\n", rt
->rtp_port_max
,
1700 return AVERROR(EINVAL
);
1703 if (!ff_network_init())
1704 return AVERROR(EIO
);
1706 if (s
->max_delay
< 0) /* Not set by the caller */
1707 s
->max_delay
= s
->iformat
? DEFAULT_REORDERING_DELAY
: 0;
1709 rt
->control_transport
= RTSP_MODE_PLAIN
;
1710 if (rt
->lower_transport_mask
& ((1 << RTSP_LOWER_TRANSPORT_HTTP
) |
1711 (1 << RTSP_LOWER_TRANSPORT_HTTPS
))) {
1712 https_tunnel
= !!(rt
->lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_HTTPS
));
1713 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1714 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1716 /* Only pass through valid flags from here */
1717 rt
->lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1720 memset(&reply1
, 0, sizeof(reply1
));
1721 /* extract hostname and port */
1722 av_url_split(proto
, sizeof(proto
), auth
, sizeof(auth
),
1723 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->url
);
1725 if (!strcmp(proto
, "rtsps")) {
1726 lower_rtsp_proto
= "tls";
1727 default_port
= RTSPS_DEFAULT_PORT
;
1728 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1732 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1735 port
= default_port
;
1737 lower_transport_mask
= rt
->lower_transport_mask
;
1739 if (!lower_transport_mask
)
1740 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1743 /* Only UDP or TCP - UDP multicast isn't supported. */
1744 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1745 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1746 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1747 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1748 "only UDP and TCP are supported for output.\n");
1749 err
= AVERROR(EINVAL
);
1754 /* Construct the URI used in request; this is similar to s->url,
1755 * but with authentication credentials removed and RTSP specific options
1757 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), proto
, NULL
,
1758 host
, port
, "%s", path
);
1760 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1761 /* set up initial handshake for tunneling */
1762 char httpname
[1024];
1763 char sessioncookie
[17];
1765 AVDictionary
*options
= NULL
;
1767 av_dict_set_int(&options
, "timeout", rt
->stimeout
, 0);
1769 ff_url_join(httpname
, sizeof(httpname
), https_tunnel
? "https" : "http", auth
, host
, port
, "%s", path
);
1770 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1771 av_get_random_seed(), av_get_random_seed());
1774 if (ffurl_alloc(&rt
->rtsp_hd
, httpname
, AVIO_FLAG_READ
,
1775 &s
->interrupt_callback
) < 0) {
1780 /* generate GET headers */
1781 snprintf(headers
, sizeof(headers
),
1782 "x-sessioncookie: %s\r\n"
1783 "Accept: application/x-rtsp-tunnelled\r\n"
1784 "Pragma: no-cache\r\n"
1785 "Cache-Control: no-cache\r\n",
1787 av_opt_set(rt
->rtsp_hd
->priv_data
, "headers", headers
, 0);
1789 if (!rt
->rtsp_hd
->protocol_whitelist
&& s
->protocol_whitelist
) {
1790 rt
->rtsp_hd
->protocol_whitelist
= av_strdup(s
->protocol_whitelist
);
1791 if (!rt
->rtsp_hd
->protocol_whitelist
) {
1792 err
= AVERROR(ENOMEM
);
1797 /* complete the connection */
1798 if (ffurl_connect(rt
->rtsp_hd
, &options
)) {
1799 av_dict_free(&options
);
1805 if (ffurl_alloc(&rt
->rtsp_hd_out
, httpname
, AVIO_FLAG_WRITE
,
1806 &s
->interrupt_callback
) < 0 ) {
1811 /* generate POST headers */
1812 snprintf(headers
, sizeof(headers
),
1813 "x-sessioncookie: %s\r\n"
1814 "Content-Type: application/x-rtsp-tunnelled\r\n"
1815 "Pragma: no-cache\r\n"
1816 "Cache-Control: no-cache\r\n"
1817 "Content-Length: 32767\r\n"
1818 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1820 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "headers", headers
, 0);
1821 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "chunked_post", "0", 0);
1822 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "send_expect_100", "0", 0);
1824 /* Initialize the authentication state for the POST session. The HTTP
1825 * protocol implementation doesn't properly handle multi-pass
1826 * authentication for POST requests, since it would require one of
1828 * - implementing Expect: 100-continue, which many HTTP servers
1829 * don't support anyway, even less the RTSP servers that do HTTP
1831 * - sending the whole POST data until getting a 401 reply specifying
1832 * what authentication method to use, then resending all that data
1833 * - waiting for potential 401 replies directly after sending the
1834 * POST header (waiting for some unspecified time)
1835 * Therefore, we copy the full auth state, which works for both basic
1836 * and digest. (For digest, we would have to synchronize the nonce
1837 * count variable between the two sessions, if we'd do more requests
1838 * with the original session, though.)
1840 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1842 /* complete the connection */
1843 if (ffurl_connect(rt
->rtsp_hd_out
, &options
)) {
1844 av_dict_free(&options
);
1848 av_dict_free(&options
);
1851 /* open the tcp connection */
1852 ff_url_join(tcpname
, sizeof(tcpname
), lower_rtsp_proto
, NULL
,
1854 "?timeout=%d", rt
->stimeout
);
1855 if ((ret
= ffurl_open_whitelist(&rt
->rtsp_hd
, tcpname
, AVIO_FLAG_READ_WRITE
,
1856 &s
->interrupt_callback
, NULL
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
)) < 0) {
1860 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1864 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1869 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1870 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1871 NULL
, 0, NI_NUMERICHOST
);
1874 /* request options supported by the server; this also detects server
1876 for (rt
->server_type
= RTSP_SERVER_RTP
;;) {
1878 if (rt
->server_type
== RTSP_SERVER_REAL
)
1881 * The following entries are required for proper
1882 * streaming from a Realmedia server. They are
1883 * interdependent in some way although we currently
1884 * don't quite understand how. Values were copied
1885 * from mplayer SVN r23589.
1886 * ClientChallenge is a 16-byte ID in hex
1887 * CompanyID is a 16-byte ID in base64
1889 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1890 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1891 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1892 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1894 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
1895 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1896 err
= ff_rtsp_averror(reply
->status_code
, AVERROR_INVALIDDATA
);
1900 /* detect server type if not standard-compliant RTP */
1901 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
1902 rt
->server_type
= RTSP_SERVER_REAL
;
1904 } else if (!av_strncasecmp(reply
->server
, "WMServer/", 9)) {
1905 rt
->server_type
= RTSP_SERVER_WMS
;
1906 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
1907 strcpy(real_challenge
, reply
->real_challenge
);
1911 if (CONFIG_RTSP_DEMUXER
&& s
->iformat
)
1912 err
= ff_rtsp_setup_input_streams(s
, reply
);
1913 else if (CONFIG_RTSP_MUXER
)
1914 err
= ff_rtsp_setup_output_streams(s
, host
);
1921 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
1922 ~(lower_transport_mask
- 1)];
1924 if ((lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_TCP
))
1925 && (rt
->rtsp_flags
& RTSP_FLAG_PREFER_TCP
))
1926 lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
1928 err
= ff_rtsp_make_setup_request(s
, host
, port
, lower_transport
,
1929 rt
->server_type
== RTSP_SERVER_REAL
?
1930 real_challenge
: NULL
);
1933 lower_transport_mask
&= ~(1 << lower_transport
);
1934 if (lower_transport_mask
== 0 && err
== 1) {
1935 err
= AVERROR(EPROTONOSUPPORT
);
1940 rt
->lower_transport_mask
= lower_transport_mask
;
1941 av_strlcpy(rt
->real_challenge
, real_challenge
, sizeof(rt
->real_challenge
));
1942 rt
->state
= RTSP_STATE_IDLE
;
1943 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
1946 ff_rtsp_close_streams(s
);
1947 ff_rtsp_close_connections(s
);
1948 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
1949 char *new_url
= av_strdup(reply
->location
);
1951 err
= AVERROR(ENOMEM
);
1954 ff_format_set_url(s
, new_url
);
1955 rt
->session_id
[0] = '\0';
1956 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
1965 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1968 static int parse_rtsp_message(AVFormatContext
*s
)
1970 RTSPState
*rt
= s
->priv_data
;
1973 if (rt
->rtsp_flags
& RTSP_FLAG_LISTEN
) {
1974 if (rt
->state
== RTSP_STATE_STREAMING
) {
1975 if (!ff_rtsp_parse_streaming_commands(s
))
1978 av_log(s
, AV_LOG_WARNING
,
1979 "Unable to answer to TEARDOWN\n");
1983 RTSPMessageHeader reply
;
1984 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0, NULL
);
1987 /* XXX: parse message */
1988 if (rt
->state
!= RTSP_STATE_STREAMING
)
1995 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1996 uint8_t *buf
, int buf_size
, int64_t wait_end
)
1998 RTSPState
*rt
= s
->priv_data
;
1999 RTSPStream
*rtsp_st
;
2000 int n
, i
, ret
, timeout_cnt
= 0;
2001 struct pollfd
*p
= rt
->p
;
2002 int *fds
= NULL
, fdsnum
, fdsidx
;
2005 p
= rt
->p
= av_malloc_array(2 * (rt
->nb_rtsp_streams
+ 1), sizeof(struct pollfd
));
2007 return AVERROR(ENOMEM
);
2010 p
[rt
->max_p
].fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
2011 p
[rt
->max_p
++].events
= POLLIN
;
2013 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2014 rtsp_st
= rt
->rtsp_streams
[i
];
2015 if (rtsp_st
->rtp_handle
) {
2016 if (ret
= ffurl_get_multi_file_handle(rtsp_st
->rtp_handle
,
2018 av_log(s
, AV_LOG_ERROR
, "Unable to recover rtp ports\n");
2022 av_log(s
, AV_LOG_ERROR
,
2023 "Number of fds %d not supported\n", fdsnum
);
2024 return AVERROR_INVALIDDATA
;
2026 for (fdsidx
= 0; fdsidx
< fdsnum
; fdsidx
++) {
2027 p
[rt
->max_p
].fd
= fds
[fdsidx
];
2028 p
[rt
->max_p
++].events
= POLLIN
;
2036 if (ff_check_interrupt(&s
->interrupt_callback
))
2037 return AVERROR_EXIT
;
2038 if (wait_end
&& wait_end
- av_gettime_relative() < 0)
2039 return AVERROR(EAGAIN
);
2040 n
= poll(p
, rt
->max_p
, POLL_TIMEOUT_MS
);
2042 int j
= rt
->rtsp_hd
? 1 : 0;
2044 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2045 rtsp_st
= rt
->rtsp_streams
[i
];
2046 if (rtsp_st
->rtp_handle
) {
2047 if (p
[j
].revents
& POLLIN
|| p
[j
+1].revents
& POLLIN
) {
2048 ret
= ffurl_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
2050 *prtsp_st
= rtsp_st
;
2057 #if CONFIG_RTSP_DEMUXER
2058 if (rt
->rtsp_hd
&& p
[0].revents
& POLLIN
) {
2059 if ((ret
= parse_rtsp_message(s
)) < 0) {
2064 } else if (n
== 0 && ++timeout_cnt
>= MAX_TIMEOUTS
) {
2065 return AVERROR(ETIMEDOUT
);
2066 } else if (n
< 0 && errno
!= EINTR
)
2067 return AVERROR(errno
);
2071 static int pick_stream(AVFormatContext
*s
, RTSPStream
**rtsp_st
,
2072 const uint8_t *buf
, int len
)
2074 RTSPState
*rt
= s
->priv_data
;
2078 if (rt
->nb_rtsp_streams
== 1) {
2079 *rtsp_st
= rt
->rtsp_streams
[0];
2082 if (len
>= 8 && rt
->transport
== RTSP_TRANSPORT_RTP
) {
2083 if (RTP_PT_IS_RTCP(rt
->recvbuf
[1])) {
2085 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2086 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
2089 if (rtpctx
->ssrc
== AV_RB32(&buf
[4])) {
2090 *rtsp_st
= rt
->rtsp_streams
[i
];
2097 av_log(s
, AV_LOG_WARNING
,
2098 "Unable to pick stream for packet - SSRC not known for "
2100 return AVERROR(EAGAIN
);
2103 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2104 if ((buf
[1] & 0x7f) == rt
->rtsp_streams
[i
]->sdp_payload_type
) {
2105 *rtsp_st
= rt
->rtsp_streams
[i
];
2111 av_log(s
, AV_LOG_WARNING
, "Unable to pick stream for packet\n");
2112 return AVERROR(EAGAIN
);
2115 static int read_packet(AVFormatContext
*s
,
2116 RTSPStream
**rtsp_st
, RTSPStream
*first_queue_st
,
2119 RTSPState
*rt
= s
->priv_data
;
2122 switch(rt
->lower_transport
) {
2124 #if CONFIG_RTSP_DEMUXER
2125 case RTSP_LOWER_TRANSPORT_TCP
:
2126 len
= ff_rtsp_tcp_read_packet(s
, rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
);
2129 case RTSP_LOWER_TRANSPORT_UDP
:
2130 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
2131 len
= udp_read_packet(s
, rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
, wait_end
);
2132 if (len
> 0 && (*rtsp_st
)->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
2133 ff_rtp_check_and_send_back_rr((*rtsp_st
)->transport_priv
, (*rtsp_st
)->rtp_handle
, NULL
, len
);
2135 case RTSP_LOWER_TRANSPORT_CUSTOM
:
2136 if (first_queue_st
&& rt
->transport
== RTSP_TRANSPORT_RTP
&&
2137 wait_end
&& wait_end
< av_gettime_relative())
2138 len
= AVERROR(EAGAIN
);
2140 len
= avio_read_partial(s
->pb
, rt
->recvbuf
, RECVBUF_SIZE
);
2141 len
= pick_stream(s
, rtsp_st
, rt
->recvbuf
, len
);
2142 if (len
> 0 && (*rtsp_st
)->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
2143 ff_rtp_check_and_send_back_rr((*rtsp_st
)->transport_priv
, NULL
, s
->pb
, len
);
2153 int ff_rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
2155 RTSPState
*rt
= s
->priv_data
;
2157 RTSPStream
*rtsp_st
, *first_queue_st
= NULL
;
2158 int64_t wait_end
= 0;
2160 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
2163 /* get next frames from the same RTP packet */
2164 if (rt
->cur_transport_priv
) {
2165 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
2166 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
2167 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
2168 ret
= ff_rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
2169 } else if (CONFIG_RTPDEC
&& rt
->ts
) {
2170 ret
= avpriv_mpegts_parse_packet(rt
->ts
, pkt
, rt
->recvbuf
+ rt
->recvbuf_pos
, rt
->recvbuf_len
- rt
->recvbuf_pos
);
2172 rt
->recvbuf_pos
+= ret
;
2173 ret
= rt
->recvbuf_pos
< rt
->recvbuf_len
;
2178 rt
->cur_transport_priv
= NULL
;
2180 } else if (ret
== 1) {
2183 rt
->cur_transport_priv
= NULL
;
2187 if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
2189 int64_t first_queue_time
= 0;
2190 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2191 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
2195 queue_time
= ff_rtp_queued_packet_time(rtpctx
);
2196 if (queue_time
&& (queue_time
- first_queue_time
< 0 ||
2197 !first_queue_time
)) {
2198 first_queue_time
= queue_time
;
2199 first_queue_st
= rt
->rtsp_streams
[i
];
2202 if (first_queue_time
) {
2203 wait_end
= first_queue_time
+ s
->max_delay
;
2206 first_queue_st
= NULL
;
2210 /* read next RTP packet */
2212 rt
->recvbuf
= av_malloc(RECVBUF_SIZE
);
2214 return AVERROR(ENOMEM
);
2217 len
= read_packet(s
, &rtsp_st
, first_queue_st
, wait_end
);
2218 if (len
== AVERROR(EAGAIN
) && first_queue_st
&&
2219 rt
->transport
== RTSP_TRANSPORT_RTP
) {
2220 av_log(s
, AV_LOG_WARNING
,
2221 "max delay reached. need to consume packet\n");
2222 rtsp_st
= first_queue_st
;
2223 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, NULL
, 0);
2229 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
2230 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
2231 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
2232 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
2233 if (rtsp_st
->feedback
) {
2234 AVIOContext
*pb
= NULL
;
2235 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_CUSTOM
)
2237 ff_rtp_send_rtcp_feedback(rtsp_st
->transport_priv
, rtsp_st
->rtp_handle
, pb
);
2240 /* Either bad packet, or a RTCP packet. Check if the
2241 * first_rtcp_ntp_time field was initialized. */
2242 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
2243 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
2244 /* first_rtcp_ntp_time has been initialized for this stream,
2245 * copy the same value to all other uninitialized streams,
2246 * in order to map their timestamp origin to the same ntp time
2249 AVStream
*st
= NULL
;
2250 if (rtsp_st
->stream_index
>= 0)
2251 st
= s
->streams
[rtsp_st
->stream_index
];
2252 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2253 RTPDemuxContext
*rtpctx2
= rt
->rtsp_streams
[i
]->transport_priv
;
2254 AVStream
*st2
= NULL
;
2255 if (rt
->rtsp_streams
[i
]->stream_index
>= 0)
2256 st2
= s
->streams
[rt
->rtsp_streams
[i
]->stream_index
];
2257 if (rtpctx2
&& st
&& st2
&&
2258 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) {
2259 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
2260 rtpctx2
->rtcp_ts_offset
= av_rescale_q(
2261 rtpctx
->rtcp_ts_offset
, st
->time_base
,
2265 // Make real NTP start time available in AVFormatContext
2266 if (s
->start_time_realtime
== AV_NOPTS_VALUE
) {
2267 s
->start_time_realtime
= av_rescale (rtpctx
->first_rtcp_ntp_time
- (NTP_OFFSET
<< 32), 1000000, 1LL << 32);
2269 s
->start_time_realtime
-=
2270 av_rescale (rtpctx
->rtcp_ts_offset
,
2271 (uint64_t) rtpctx
->st
->time_base
.num
* 1000000,
2272 rtpctx
->st
->time_base
.den
);
2276 if (ret
== -RTCP_BYE
) {
2279 av_log(s
, AV_LOG_DEBUG
, "Received BYE for stream %d (%d/%d)\n",
2280 rtsp_st
->stream_index
, rt
->nb_byes
, rt
->nb_rtsp_streams
);
2282 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
2286 } else if (CONFIG_RTPDEC
&& rt
->ts
) {
2287 ret
= avpriv_mpegts_parse_packet(rt
->ts
, pkt
, rt
->recvbuf
, len
);
2290 rt
->recvbuf_len
= len
;
2291 rt
->recvbuf_pos
= ret
;
2292 rt
->cur_transport_priv
= rt
->ts
;
2299 return AVERROR_INVALIDDATA
;
2305 /* more packets may follow, so we save the RTP context */
2306 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
2310 #endif /* CONFIG_RTPDEC */
2312 #if CONFIG_SDP_DEMUXER
2313 static int sdp_probe(const AVProbeData
*p1
)
2315 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
2317 /* we look for a line beginning "c=IN IP" */
2318 while (p
< p_end
&& *p
!= '\0') {
2319 if (sizeof("c=IN IP") - 1 < p_end
- p
&&
2320 av_strstart(p
, "c=IN IP", NULL
))
2321 return AVPROBE_SCORE_EXTENSION
;
2323 while (p
< p_end
- 1 && *p
!= '\n') p
++;
2332 static void append_source_addrs(char *buf
, int size
, const char *name
,
2333 int count
, struct RTSPSource
**addrs
)
2338 av_strlcatf(buf
, size
, "&%s=%s", name
, addrs
[0]->addr
);
2339 for (i
= 1; i
< count
; i
++)
2340 av_strlcatf(buf
, size
, ",%s", addrs
[i
]->addr
);
2343 static int sdp_read_header(AVFormatContext
*s
)
2345 RTSPState
*rt
= s
->priv_data
;
2346 RTSPStream
*rtsp_st
;
2351 if (!ff_network_init())
2352 return AVERROR(EIO
);
2354 if (s
->max_delay
< 0) /* Not set by the caller */
2355 s
->max_delay
= DEFAULT_REORDERING_DELAY
;
2356 if (rt
->rtsp_flags
& RTSP_FLAG_CUSTOM_IO
)
2357 rt
->lower_transport
= RTSP_LOWER_TRANSPORT_CUSTOM
;
2359 /* read the whole sdp file */
2360 /* XXX: better loading */
2361 content
= av_malloc(SDP_MAX_SIZE
);
2363 return AVERROR(ENOMEM
);
2364 size
= avio_read(s
->pb
, content
, SDP_MAX_SIZE
- 1);
2367 return AVERROR_INVALIDDATA
;
2369 content
[size
] ='\0';
2371 err
= ff_sdp_parse(s
, content
);
2375 /* open each RTP stream */
2376 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2378 rtsp_st
= rt
->rtsp_streams
[i
];
2380 if (!(rt
->rtsp_flags
& RTSP_FLAG_CUSTOM_IO
)) {
2381 AVDictionary
*opts
= map_to_opts(rt
);
2383 err
= getnameinfo((struct sockaddr
*) &rtsp_st
->sdp_ip
,
2384 sizeof(rtsp_st
->sdp_ip
),
2385 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
2387 av_log(s
, AV_LOG_ERROR
, "getnameinfo: %s\n", gai_strerror(err
));
2389 av_dict_free(&opts
);
2392 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
2393 namebuf
, rtsp_st
->sdp_port
,
2394 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2395 rtsp_st
->sdp_port
, rtsp_st
->sdp_ttl
,
2396 rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
? 1 : 0,
2397 rt
->rtsp_flags
& RTSP_FLAG_RTCP_TO_SOURCE
? 1 : 0);
2399 append_source_addrs(url
, sizeof(url
), "sources",
2400 rtsp_st
->nb_include_source_addrs
,
2401 rtsp_st
->include_source_addrs
);
2402 append_source_addrs(url
, sizeof(url
), "block",
2403 rtsp_st
->nb_exclude_source_addrs
,
2404 rtsp_st
->exclude_source_addrs
);
2405 err
= ffurl_open_whitelist(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ
,
2406 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
2408 av_dict_free(&opts
);
2411 err
= AVERROR_INVALIDDATA
;
2415 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
2420 ff_rtsp_close_streams(s
);
2425 static int sdp_read_close(AVFormatContext
*s
)
2427 ff_rtsp_close_streams(s
);
2432 static const AVClass sdp_demuxer_class
= {
2433 .class_name
= "SDP demuxer",
2434 .item_name
= av_default_item_name
,
2435 .option
= sdp_options
,
2436 .version
= LIBAVUTIL_VERSION_INT
,
2439 AVInputFormat ff_sdp_demuxer
= {
2441 .long_name
= NULL_IF_CONFIG_SMALL("SDP"),
2442 .priv_data_size
= sizeof(RTSPState
),
2443 .read_probe
= sdp_probe
,
2444 .read_header
= sdp_read_header
,
2445 .read_packet
= ff_rtsp_fetch_packet
,
2446 .read_close
= sdp_read_close
,
2447 .priv_class
= &sdp_demuxer_class
,
2449 #endif /* CONFIG_SDP_DEMUXER */
2451 #if CONFIG_RTP_DEMUXER
2452 static int rtp_probe(const AVProbeData
*p
)
2454 if (av_strstart(p
->filename
, "rtp:", NULL
))
2455 return AVPROBE_SCORE_MAX
;
2459 static int rtp_read_header(AVFormatContext
*s
)
2461 uint8_t recvbuf
[RTP_MAX_PACKET_LENGTH
];
2462 char host
[500], filters_buf
[1000];
2464 URLContext
* in
= NULL
;
2466 AVCodecParameters
*par
= NULL
;
2467 struct sockaddr_storage addr
;
2469 socklen_t addrlen
= sizeof(addr
);
2470 RTSPState
*rt
= s
->priv_data
;
2474 if (!ff_network_init())
2475 return AVERROR(EIO
);
2477 ret
= ffurl_open_whitelist(&in
, s
->url
, AVIO_FLAG_READ
,
2478 &s
->interrupt_callback
, NULL
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
2483 ret
= ffurl_read(in
, recvbuf
, sizeof(recvbuf
));
2484 if (ret
== AVERROR(EAGAIN
))
2489 av_log(s
, AV_LOG_WARNING
, "Received too short packet\n");
2493 if ((recvbuf
[0] & 0xc0) != 0x80) {
2494 av_log(s
, AV_LOG_WARNING
, "Unsupported RTP version packet "
2499 if (RTP_PT_IS_RTCP(recvbuf
[1]))
2502 payload_type
= recvbuf
[1] & 0x7f;
2505 getsockname(ffurl_get_file_handle(in
), (struct sockaddr
*) &addr
, &addrlen
);
2508 par
= avcodec_parameters_alloc();
2510 ret
= AVERROR(ENOMEM
);
2514 if (ff_rtp_get_codec_info(par
, payload_type
)) {
2515 av_log(s
, AV_LOG_ERROR
, "Unable to receive RTP payload type %d "
2516 "without an SDP file describing it\n",
2520 if (par
->codec_type
!= AVMEDIA_TYPE_DATA
) {
2521 av_log(s
, AV_LOG_WARNING
, "Guessing on RTP content - if not received "
2522 "properly you need an SDP file "
2526 av_url_split(NULL
, 0, NULL
, 0, host
, sizeof(host
), &port
,
2529 av_bprint_init(&sdp
, 0, AV_BPRINT_SIZE_UNLIMITED
);
2530 av_bprintf(&sdp
, "v=0\r\nc=IN IP%d %s\r\n",
2531 addr
.ss_family
== AF_INET
? 4 : 6, host
);
2533 p
= strchr(s
->url
, '?');
2535 static const char filters
[][2][8] = { { "sources", "incl" },
2536 { "block", "excl" } };
2539 for (i
= 0; i
< FF_ARRAY_ELEMS(filters
); i
++) {
2540 if (av_find_info_tag(filters_buf
, sizeof(filters_buf
), filters
[i
][0], p
)) {
2542 while ((q
= strchr(q
, ',')) != NULL
)
2544 av_bprintf(&sdp
, "a=source-filter:%s IN IP%d %s %s\r\n",
2546 addr
.ss_family
== AF_INET
? 4 : 6, host
,
2552 av_bprintf(&sdp
, "m=%s %d RTP/AVP %d\r\n",
2553 par
->codec_type
== AVMEDIA_TYPE_DATA
? "application" :
2554 par
->codec_type
== AVMEDIA_TYPE_VIDEO
? "video" : "audio",
2555 port
, payload_type
);
2556 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
.str
);
2557 if (!av_bprint_is_complete(&sdp
))
2559 avcodec_parameters_free(&par
);
2561 ffio_init_context(&pb
, sdp
.str
, sdp
.len
, 0, NULL
, NULL
, NULL
, NULL
);
2564 /* sdp_read_header initializes this again */
2567 rt
->media_type_mask
= (1 << (AVMEDIA_TYPE_SUBTITLE
+1)) - 1;
2569 ret
= sdp_read_header(s
);
2571 av_bprint_finalize(&sdp
, NULL
);
2575 ret
= AVERROR(ENOMEM
);
2576 av_log(s
, AV_LOG_ERROR
, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2577 av_bprint_finalize(&sdp
, NULL
);
2579 avcodec_parameters_free(&par
);
2585 static const AVClass rtp_demuxer_class
= {
2586 .class_name
= "RTP demuxer",
2587 .item_name
= av_default_item_name
,
2588 .option
= rtp_options
,
2589 .version
= LIBAVUTIL_VERSION_INT
,
2592 AVInputFormat ff_rtp_demuxer
= {
2594 .long_name
= NULL_IF_CONFIG_SMALL("RTP input"),
2595 .priv_data_size
= sizeof(RTSPState
),
2596 .read_probe
= rtp_probe
,
2597 .read_header
= rtp_read_header
,
2598 .read_packet
= ff_rtsp_fetch_packet
,
2599 .read_close
= sdp_read_close
,
2600 .flags
= AVFMT_NOFILE
,
2601 .priv_class
= &rtp_demuxer_class
,
2603 #endif /* CONFIG_RTP_DEMUXER */