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 "config_components.h"
24 #include "libavutil/avassert.h"
25 #include "libavutil/base64.h"
26 #include "libavutil/bprint.h"
27 #include "libavutil/avstring.h"
28 #include "libavutil/intreadwrite.h"
29 #include "libavutil/mathematics.h"
30 #include "libavutil/mem.h"
31 #include "libavutil/parseutils.h"
32 #include "libavutil/random_seed.h"
33 #include "libavutil/dict.h"
34 #include "libavutil/opt.h"
35 #include "libavutil/time.h"
36 #include "libavcodec/codec_desc.h"
38 #include "avio_internal.h"
46 #include "os_support.h"
53 #include "rtpdec_formats.h"
54 #include "rtpenc_chain.h"
61 /* Default timeout values for read packet in seconds */
62 #define READ_PACKET_TIMEOUT_S 10
63 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
64 #define DEFAULT_REORDERING_DELAY 100000
66 #define OFFSET(x) offsetof(RTSPState, x)
67 #define DEC AV_OPT_FLAG_DECODING_PARAM
68 #define ENC AV_OPT_FLAG_ENCODING_PARAM
70 #define RTSP_FLAG_OPTS(name, longname) \
71 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, .unit = "rtsp_flags" }, \
72 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, .unit = "rtsp_flags" }
74 #define RTSP_MEDIATYPE_OPTS(name, longname) \
75 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, .unit = "allowed_media_types" }, \
76 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, .unit = "allowed_media_types" }, \
77 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, .unit = "allowed_media_types" }, \
78 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, .unit = "allowed_media_types" }, \
79 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, .unit = "allowed_media_types" }
81 #define COMMON_OPTS() \
82 { "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 }, \
83 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
84 { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = 1472 }, -1, INT_MAX, ENC } \
87 const AVOption ff_rtsp_options[] = {
88 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause
), AV_OPT_TYPE_BOOL
, {.i64
= 0}, 0, 1, DEC
},
89 FF_RTP_FLAG_OPTS(RTSPState
, rtp_muxer_flags
),
90 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask
), AV_OPT_TYPE_FLAGS
, {.i64
= 0}, INT_MIN
, INT_MAX
, DEC
|ENC
, .unit
= "rtsp_transport" }, \
91 { "udp", "UDP", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_UDP
}, 0, 0, DEC
|ENC
, .unit
= "rtsp_transport" }, \
92 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_TCP
}, 0, 0, DEC
|ENC
, .unit
= "rtsp_transport" }, \
93 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST
}, 0, 0, DEC
, .unit
= "rtsp_transport" },
94 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST
, {.i64
= (1 << RTSP_LOWER_TRANSPORT_HTTP
)}, 0, 0, DEC
, .unit
= "rtsp_transport" },
95 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST
, {.i64
= (1 << RTSP_LOWER_TRANSPORT_HTTPS
)}, 0, 0, DEC
, .unit
= "rtsp_transport" },
96 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
97 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_LISTEN
}, 0, 0, DEC
, .unit
= "rtsp_flags" },
98 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_PREFER_TCP
}, 0, 0, DEC
|ENC
, .unit
= "rtsp_flags" },
99 { "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_SATIP_RAW
}, 0, 0, DEC
, .unit
= "rtsp_flags" },
100 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
101 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min
), AV_OPT_TYPE_INT
, {.i64
= RTSP_RTP_PORT_MIN
}, 0, 65535, DEC
|ENC
},
102 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max
), AV_OPT_TYPE_INT
, {.i64
= RTSP_RTP_PORT_MAX
}, 0, 65535, DEC
|ENC
},
103 { "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
},
104 { "timeout", "set timeout (in microseconds) of socket I/O operations", OFFSET(stimeout
), AV_OPT_TYPE_INT64
, {.i64
= 0}, INT_MIN
, INT64_MAX
, DEC
},
106 { "user_agent", "override User-Agent header", OFFSET(user_agent
), AV_OPT_TYPE_STRING
, {.str
= LIBAVFORMAT_IDENT
}, 0, 0, DEC
},
109 FF_TLS_CLIENT_OPTIONS(RTSPState
, tls_opts
),
113 static const AVOption sdp_options
[] = {
114 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
115 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_CUSTOM_IO
}, 0, 0, DEC
, .unit
= "rtsp_flags" },
116 { "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
, .unit
= "rtsp_flags" },
117 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(stimeout
), AV_OPT_TYPE_DURATION
, {.i64
= READ_PACKET_TIMEOUT_S
*1000000}, INT_MIN
, INT64_MAX
, DEC
},
118 { "localaddr", "local address", OFFSET(localaddr
),AV_OPT_TYPE_STRING
, {.str
= NULL
}, 0, 0, DEC
}, \
119 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
124 static const AVOption rtp_options
[] = {
125 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
126 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(stimeout
), AV_OPT_TYPE_DURATION
, {.i64
= READ_PACKET_TIMEOUT_S
*1000000}, INT_MIN
, INT64_MAX
, DEC
},
127 { "localaddr", "local address", OFFSET(localaddr
),AV_OPT_TYPE_STRING
, {.str
= NULL
}, 0, 0, DEC
}, \
128 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
134 static AVDictionary
*map_to_opts(RTSPState
*rt
)
136 AVDictionary
*opts
= NULL
;
138 av_dict_set_int(&opts
, "buffer_size", rt
->buffer_size
, 0);
139 av_dict_set_int(&opts
, "pkt_size", rt
->pkt_size
, 0);
140 if (rt
->localaddr
&& rt
->localaddr
[0])
141 av_dict_set(&opts
, "localaddr", rt
->localaddr
, 0);
154 * Add the TLS options of the given RTSPState to the dict
156 static int copy_tls_opts_dict(RTSPState
*rt
, AVDictionary
**dict
)
158 ERR_RET(av_dict_set_int(dict
, "tls_verify", rt
->tls_opts
.verify
, 0));
159 ERR_RET(av_dict_set(dict
, "ca_file", rt
->tls_opts
.ca_file
, 0));
160 ERR_RET(av_dict_set(dict
, "cert_file", rt
->tls_opts
.cert_file
, 0));
161 ERR_RET(av_dict_set(dict
, "key_file", rt
->tls_opts
.key_file
, 0));
162 ERR_RET(av_dict_set(dict
, "verifyhost", rt
->tls_opts
.host
, 0));
169 static void get_word_until_chars(char *buf
, int buf_size
,
170 const char *sep
, const char **pp
)
176 p
+= strspn(p
, SPACE_CHARS
);
178 while (!strchr(sep
, *p
) && *p
!= '\0') {
179 if ((q
- buf
) < buf_size
- 1)
188 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
191 if (**pp
== '/') (*pp
)++;
192 get_word_until_chars(buf
, buf_size
, sep
, pp
);
195 static void get_word(char *buf
, int buf_size
, const char **pp
)
197 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
200 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
202 * Used for seeking in the rtp stream.
204 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
208 p
+= strspn(p
, SPACE_CHARS
);
209 if (!av_stristart(p
, "npt=", &p
))
212 *start
= AV_NOPTS_VALUE
;
213 *end
= AV_NOPTS_VALUE
;
215 get_word_sep(buf
, sizeof(buf
), "-", &p
);
216 if (av_parse_time(start
, buf
, 1) < 0)
220 get_word_sep(buf
, sizeof(buf
), "-", &p
);
221 if (av_parse_time(end
, buf
, 1) < 0)
222 av_log(NULL
, AV_LOG_DEBUG
, "Failed to parse interval end specification '%s'\n", buf
);
226 static int get_sockaddr(AVFormatContext
*s
,
227 const char *buf
, struct sockaddr_storage
*sock
)
229 struct addrinfo hints
= { 0 }, *ai
= NULL
;
232 hints
.ai_flags
= AI_NUMERICHOST
;
233 if ((ret
= getaddrinfo(buf
, NULL
, &hints
, &ai
))) {
234 av_log(s
, AV_LOG_ERROR
, "getaddrinfo(%s): %s\n",
239 memcpy(sock
, ai
->ai_addr
, FFMIN(sizeof(*sock
), ai
->ai_addrlen
));
245 static void init_rtp_handler(const RTPDynamicProtocolHandler
*handler
,
246 RTSPStream
*rtsp_st
, AVStream
*st
)
248 AVCodecParameters
*par
= st
? st
->codecpar
: NULL
;
252 par
->codec_id
= handler
->codec_id
;
253 rtsp_st
->dynamic_handler
= handler
;
255 ffstream(st
)->need_parsing
= handler
->need_parsing
;
256 if (handler
->priv_data_size
) {
257 rtsp_st
->dynamic_protocol_context
= av_mallocz(handler
->priv_data_size
);
258 if (!rtsp_st
->dynamic_protocol_context
)
259 rtsp_st
->dynamic_handler
= NULL
;
263 static void finalize_rtp_handler_init(AVFormatContext
*s
, RTSPStream
*rtsp_st
,
266 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_handler
->init
) {
267 int ret
= rtsp_st
->dynamic_handler
->init(s
, st
? st
->index
: -1,
268 rtsp_st
->dynamic_protocol_context
);
270 if (rtsp_st
->dynamic_protocol_context
) {
271 if (rtsp_st
->dynamic_handler
->close
)
272 rtsp_st
->dynamic_handler
->close(
273 rtsp_st
->dynamic_protocol_context
);
274 av_free(rtsp_st
->dynamic_protocol_context
);
276 rtsp_st
->dynamic_protocol_context
= NULL
;
277 rtsp_st
->dynamic_handler
= NULL
;
282 #if CONFIG_RTSP_DEMUXER
283 static int init_satip_stream(AVFormatContext
*s
)
285 RTSPState
*rt
= s
->priv_data
;
286 RTSPStream
*rtsp_st
= av_mallocz(sizeof(RTSPStream
));
288 return AVERROR(ENOMEM
);
289 dynarray_add(&rt
->rtsp_streams
,
290 &rt
->nb_rtsp_streams
, rtsp_st
);
292 rtsp_st
->sdp_payload_type
= 33; // MP2T
293 av_strlcpy(rtsp_st
->control_url
,
294 rt
->control_uri
, sizeof(rtsp_st
->control_url
));
296 if (rt
->rtsp_flags
& RTSP_FLAG_SATIP_RAW
) {
297 AVStream
*st
= avformat_new_stream(s
, NULL
);
299 return AVERROR(ENOMEM
);
300 st
->id
= rt
->nb_rtsp_streams
- 1;
301 rtsp_st
->stream_index
= st
->index
;
302 st
->codecpar
->codec_type
= AVMEDIA_TYPE_DATA
;
303 st
->codecpar
->codec_id
= AV_CODEC_ID_MPEG2TS
;
305 rtsp_st
->stream_index
= -1;
306 init_rtp_handler(&ff_mpegts_dynamic_handler
, rtsp_st
, NULL
);
307 finalize_rtp_handler_init(s
, rtsp_st
, NULL
);
313 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
314 static int sdp_parse_rtpmap(AVFormatContext
*s
,
315 AVStream
*st
, RTSPStream
*rtsp_st
,
316 int payload_type
, const char *p
)
318 AVCodecParameters
*par
= st
->codecpar
;
321 const AVCodecDescriptor
*desc
;
324 /* See if we can handle this kind of payload.
325 * The space should normally not be there but some Real streams or
326 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
327 * have a trailing space. */
328 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
329 if (payload_type
< RTP_PT_PRIVATE
) {
330 /* We are in a standard case
331 * (from http://www.iana.org/assignments/rtp-parameters). */
332 par
->codec_id
= ff_rtp_codec_id(buf
, par
->codec_type
);
335 if (par
->codec_id
== AV_CODEC_ID_NONE
) {
336 const RTPDynamicProtocolHandler
*handler
=
337 ff_rtp_handler_find_by_name(buf
, par
->codec_type
);
338 init_rtp_handler(handler
, rtsp_st
, st
);
339 /* If no dynamic handler was found, check with the list of standard
340 * allocated types, if such a stream for some reason happens to
341 * use a private payload type. This isn't handled in rtpdec.c, since
342 * the format name from the rtpmap line never is passed into rtpdec. */
343 if (!rtsp_st
->dynamic_handler
)
344 par
->codec_id
= ff_rtp_codec_id(buf
, par
->codec_type
);
347 desc
= avcodec_descriptor_get(par
->codec_id
);
348 if (desc
&& desc
->name
)
353 get_word_sep(buf
, sizeof(buf
), "/", &p
);
355 switch (par
->codec_type
) {
356 case AVMEDIA_TYPE_AUDIO
:
357 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
358 par
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
359 par
->ch_layout
= (AVChannelLayout
)AV_CHANNEL_LAYOUT_MONO
;
361 par
->sample_rate
= i
;
362 avpriv_set_pts_info(st
, 32, 1, par
->sample_rate
);
363 get_word_sep(buf
, sizeof(buf
), "/", &p
);
366 av_channel_layout_default(&par
->ch_layout
, i
);
368 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
370 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
371 par
->ch_layout
.nb_channels
);
373 case AVMEDIA_TYPE_VIDEO
:
374 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
376 avpriv_set_pts_info(st
, 32, 1, i
);
381 finalize_rtp_handler_init(s
, rtsp_st
, st
);
385 /* parse the attribute line from the fmtp a line of an sdp response. This
386 * is broken out as a function because it is used in rtp_h264.c, which is
388 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
389 char *value
, int value_size
)
391 *p
+= strspn(*p
, SPACE_CHARS
);
393 get_word_sep(attr
, attr_size
, "=", p
);
396 get_word_sep(value
, value_size
, ";", p
);
404 typedef struct SDPParseState
{
406 struct sockaddr_storage default_ip
;
408 int skip_media
; ///< set if an unknown m= line occurs
409 int nb_default_include_source_addrs
; /**< Number of source-specific multicast include source IP address (from SDP content) */
410 struct RTSPSource
**default_include_source_addrs
; /**< Source-specific multicast include source IP address (from SDP content) */
411 int nb_default_exclude_source_addrs
; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
412 struct RTSPSource
**default_exclude_source_addrs
; /**< Source-specific multicast exclude source IP address (from SDP content) */
415 char delayed_fmtp
[2048];
418 static void copy_default_source_addrs(struct RTSPSource
**addrs
, int count
,
419 struct RTSPSource
***dest
, int *dest_count
)
421 RTSPSource
*rtsp_src
, *rtsp_src2
;
423 for (i
= 0; i
< count
; i
++) {
425 rtsp_src2
= av_memdup(rtsp_src
, sizeof(*rtsp_src
));
428 dynarray_add(dest
, dest_count
, rtsp_src2
);
432 static void parse_fmtp(AVFormatContext
*s
, RTSPState
*rt
,
433 int payload_type
, const char *line
)
437 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
438 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
439 if (rtsp_st
->sdp_payload_type
== payload_type
&&
440 rtsp_st
->dynamic_handler
&&
441 rtsp_st
->dynamic_handler
->parse_sdp_a_line
) {
442 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, rtsp_st
->stream_index
,
443 rtsp_st
->dynamic_protocol_context
, line
);
448 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
449 int letter
, const char *buf
)
451 RTSPState
*rt
= s
->priv_data
;
452 char buf1
[64], st_type
[64];
454 enum AVMediaType codec_type
;
458 RTSPSource
*rtsp_src
;
459 struct sockaddr_storage sdp_ip
;
462 av_log(s
, AV_LOG_TRACE
, "sdp: %c='%s'\n", letter
, buf
);
465 if (s1
->skip_media
&& letter
!= 'm')
469 get_word(buf1
, sizeof(buf1
), &p
);
470 if (strcmp(buf1
, "IN") != 0)
472 get_word(buf1
, sizeof(buf1
), &p
);
473 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6"))
475 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
476 if (get_sockaddr(s
, buf1
, &sdp_ip
))
481 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
484 if (s
->nb_streams
== 0) {
485 s1
->default_ip
= sdp_ip
;
486 s1
->default_ttl
= ttl
;
488 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
489 rtsp_st
->sdp_ip
= sdp_ip
;
490 rtsp_st
->sdp_ttl
= ttl
;
494 av_dict_set(&s
->metadata
, "title", p
, 0);
497 if (s
->nb_streams
== 0) {
498 av_dict_set(&s
->metadata
, "comment", p
, 0);
507 codec_type
= AVMEDIA_TYPE_UNKNOWN
;
508 get_word(st_type
, sizeof(st_type
), &p
);
509 if (!strcmp(st_type
, "audio")) {
510 codec_type
= AVMEDIA_TYPE_AUDIO
;
511 } else if (!strcmp(st_type
, "video")) {
512 codec_type
= AVMEDIA_TYPE_VIDEO
;
513 } else if (!strcmp(st_type
, "application")) {
514 codec_type
= AVMEDIA_TYPE_DATA
;
515 } else if (!strcmp(st_type
, "text")) {
516 codec_type
= AVMEDIA_TYPE_SUBTITLE
;
518 if (codec_type
== AVMEDIA_TYPE_UNKNOWN
||
519 !(rt
->media_type_mask
& (1 << codec_type
)) ||
520 rt
->nb_rtsp_streams
>= s
->max_streams
525 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
528 rtsp_st
->stream_index
= -1;
529 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
531 rtsp_st
->sdp_ip
= s1
->default_ip
;
532 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
534 copy_default_source_addrs(s1
->default_include_source_addrs
,
535 s1
->nb_default_include_source_addrs
,
536 &rtsp_st
->include_source_addrs
,
537 &rtsp_st
->nb_include_source_addrs
);
538 copy_default_source_addrs(s1
->default_exclude_source_addrs
,
539 s1
->nb_default_exclude_source_addrs
,
540 &rtsp_st
->exclude_source_addrs
,
541 &rtsp_st
->nb_exclude_source_addrs
);
543 get_word(buf1
, sizeof(buf1
), &p
); /* port */
544 rtsp_st
->sdp_port
= atoi(buf1
);
546 get_word(buf1
, sizeof(buf1
), &p
); /* protocol */
547 if (!strcmp(buf1
, "udp"))
548 rt
->transport
= RTSP_TRANSPORT_RAW
;
549 else if (strstr(buf1
, "/AVPF") || strstr(buf1
, "/SAVPF"))
550 rtsp_st
->feedback
= 1;
552 /* XXX: handle list of formats */
553 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
554 rtsp_st
->sdp_payload_type
= atoi(buf1
);
556 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
557 /* no corresponding stream */
558 if (rt
->transport
== RTSP_TRANSPORT_RAW
) {
559 if (CONFIG_RTPDEC
&& !rt
->ts
)
560 rt
->ts
= avpriv_mpegts_parse_open(s
);
562 const RTPDynamicProtocolHandler
*handler
;
563 handler
= ff_rtp_handler_find_by_id(
564 rtsp_st
->sdp_payload_type
, AVMEDIA_TYPE_DATA
);
565 init_rtp_handler(handler
, rtsp_st
, NULL
);
566 finalize_rtp_handler_init(s
, rtsp_st
, NULL
);
568 } else if (rt
->server_type
== RTSP_SERVER_WMS
&&
569 codec_type
== AVMEDIA_TYPE_DATA
) {
570 /* RTX stream, a stream that carries all the other actual
571 * audio/video streams. Don't expose this to the callers. */
573 st
= avformat_new_stream(s
, NULL
);
576 st
->id
= rt
->nb_rtsp_streams
- 1;
577 rtsp_st
->stream_index
= st
->index
;
578 st
->codecpar
->codec_type
= codec_type
;
579 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
580 const RTPDynamicProtocolHandler
*handler
;
581 /* if standard payload type, we can find the codec right now */
582 ff_rtp_get_codec_info(st
->codecpar
, rtsp_st
->sdp_payload_type
);
583 if (st
->codecpar
->codec_type
== AVMEDIA_TYPE_AUDIO
&&
584 st
->codecpar
->sample_rate
> 0)
585 avpriv_set_pts_info(st
, 32, 1, st
->codecpar
->sample_rate
);
586 /* Even static payload types may need a custom depacketizer */
587 handler
= ff_rtp_handler_find_by_id(
588 rtsp_st
->sdp_payload_type
, st
->codecpar
->codec_type
);
589 init_rtp_handler(handler
, rtsp_st
, st
);
590 finalize_rtp_handler_init(s
, rtsp_st
, st
);
592 if (rt
->default_lang
[0])
593 av_dict_set(&st
->metadata
, "language", rt
->default_lang
, 0);
595 /* put a default control url */
596 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
597 sizeof(rtsp_st
->control_url
));
600 if (av_strstart(p
, "control:", &p
)) {
601 if (rt
->nb_rtsp_streams
== 0) {
602 if (!strncmp(p
, "rtsp://", 7))
603 av_strlcpy(rt
->control_uri
, p
,
604 sizeof(rt
->control_uri
));
607 /* get the control url */
608 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
610 /* XXX: may need to add full url resolution */
611 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
613 if (proto
[0] == '\0') {
614 /* relative control URL */
615 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
616 av_strlcat(rtsp_st
->control_url
, "/",
617 sizeof(rtsp_st
->control_url
));
618 av_strlcat(rtsp_st
->control_url
, p
,
619 sizeof(rtsp_st
->control_url
));
621 av_strlcpy(rtsp_st
->control_url
, p
,
622 sizeof(rtsp_st
->control_url
));
624 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
625 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
626 get_word(buf1
, sizeof(buf1
), &p
);
627 payload_type
= atoi(buf1
);
628 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
629 if (rtsp_st
->stream_index
>= 0) {
630 st
= s
->streams
[rtsp_st
->stream_index
];
631 sdp_parse_rtpmap(s
, st
, rtsp_st
, payload_type
, p
);
635 parse_fmtp(s
, rt
, payload_type
, s1
->delayed_fmtp
);
637 } else if (av_strstart(p
, "fmtp:", &p
) ||
638 av_strstart(p
, "framesize:", &p
)) {
639 // let dynamic protocol handlers have a stab at the line.
640 get_word(buf1
, sizeof(buf1
), &p
);
641 payload_type
= atoi(buf1
);
642 if (s1
->seen_rtpmap
) {
643 parse_fmtp(s
, rt
, payload_type
, buf
);
646 av_strlcpy(s1
->delayed_fmtp
, buf
, sizeof(s1
->delayed_fmtp
));
648 } else if (av_strstart(p
, "framerate:", &p
) && s
->nb_streams
> 0) {
651 if (av_sscanf(p
, "%lf%c", &framerate
, &(char){0}) == 1) {
652 st
= s
->streams
[s
->nb_streams
- 1];
653 st
->avg_frame_rate
= av_d2q(framerate
, INT_MAX
);
655 } else if (av_strstart(p
, "ssrc:", &p
) && s
->nb_streams
> 0) {
656 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
657 get_word(buf1
, sizeof(buf1
), &p
);
658 rtsp_st
->ssrc
= strtoll(buf1
, NULL
, 10);
659 } else if (av_strstart(p
, "range:", &p
)) {
662 // this is so that seeking on a streamed file can work.
663 rtsp_parse_range_npt(p
, &start
, &end
);
664 s
->start_time
= start
;
665 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
666 if (end
!= AV_NOPTS_VALUE
)
667 s
->duration
= end
- start
;
668 } else if (av_strstart(p
, "lang:", &p
)) {
669 if (s
->nb_streams
> 0) {
670 get_word(buf1
, sizeof(buf1
), &p
);
671 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
672 if (rtsp_st
->stream_index
>= 0) {
673 st
= s
->streams
[rtsp_st
->stream_index
];
674 av_dict_set(&st
->metadata
, "language", buf1
, 0);
677 get_word(rt
->default_lang
, sizeof(rt
->default_lang
), &p
);
678 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
680 rt
->transport
= RTSP_TRANSPORT_RDT
;
681 } else if (av_strstart(p
, "SampleRate:integer;", &p
) &&
683 st
= s
->streams
[s
->nb_streams
- 1];
684 st
->codecpar
->sample_rate
= atoi(p
);
685 } else if (av_strstart(p
, "crypto:", &p
) && s
->nb_streams
> 0) {
687 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
688 get_word(buf1
, sizeof(buf1
), &p
); // ignore tag
689 get_word(rtsp_st
->crypto_suite
, sizeof(rtsp_st
->crypto_suite
), &p
);
690 p
+= strspn(p
, SPACE_CHARS
);
691 if (av_strstart(p
, "inline:", &p
))
692 get_word(rtsp_st
->crypto_params
, sizeof(rtsp_st
->crypto_params
), &p
);
693 } else if (av_strstart(p
, "source-filter:", &p
)) {
695 get_word(buf1
, sizeof(buf1
), &p
);
696 if (strcmp(buf1
, "incl") && strcmp(buf1
, "excl"))
698 exclude
= !strcmp(buf1
, "excl");
700 get_word(buf1
, sizeof(buf1
), &p
);
701 if (strcmp(buf1
, "IN") != 0)
703 get_word(buf1
, sizeof(buf1
), &p
);
704 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6") && strcmp(buf1
, "*"))
706 // not checking that the destination address actually matches or is wildcard
707 get_word(buf1
, sizeof(buf1
), &p
);
710 rtsp_src
= av_mallocz(sizeof(*rtsp_src
));
713 get_word(rtsp_src
->addr
, sizeof(rtsp_src
->addr
), &p
);
715 if (s
->nb_streams
== 0) {
716 dynarray_add(&s1
->default_exclude_source_addrs
, &s1
->nb_default_exclude_source_addrs
, rtsp_src
);
718 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
719 dynarray_add(&rtsp_st
->exclude_source_addrs
, &rtsp_st
->nb_exclude_source_addrs
, rtsp_src
);
722 if (s
->nb_streams
== 0) {
723 dynarray_add(&s1
->default_include_source_addrs
, &s1
->nb_default_include_source_addrs
, rtsp_src
);
725 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
726 dynarray_add(&rtsp_st
->include_source_addrs
, &rtsp_st
->nb_include_source_addrs
, rtsp_src
);
731 if (rt
->server_type
== RTSP_SERVER_WMS
)
732 ff_wms_parse_sdp_a_line(s
, p
);
733 if (s
->nb_streams
> 0) {
734 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
736 if (rt
->server_type
== RTSP_SERVER_REAL
)
737 ff_real_parse_sdp_a_line(s
, rtsp_st
->stream_index
, p
);
739 if (rtsp_st
->dynamic_handler
&&
740 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
741 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
742 rtsp_st
->stream_index
,
743 rtsp_st
->dynamic_protocol_context
, buf
);
750 int ff_sdp_parse(AVFormatContext
*s
, const char *content
)
754 char buf
[SDP_MAX_SIZE
], *q
;
755 SDPParseState sdp_parse_state
= { { 0 } }, *s1
= &sdp_parse_state
;
757 s
->duration
= AV_NOPTS_VALUE
;
761 p
+= strspn(p
, SPACE_CHARS
);
769 /* get the content */
771 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
772 if ((q
- buf
) < sizeof(buf
) - 1)
777 sdp_parse_line(s
, s1
, letter
, buf
);
779 while (*p
!= '\n' && *p
!= '\0')
785 for (i
= 0; i
< s1
->nb_default_include_source_addrs
; i
++)
786 av_freep(&s1
->default_include_source_addrs
[i
]);
787 av_freep(&s1
->default_include_source_addrs
);
788 for (i
= 0; i
< s1
->nb_default_exclude_source_addrs
; i
++)
789 av_freep(&s1
->default_exclude_source_addrs
[i
]);
790 av_freep(&s1
->default_exclude_source_addrs
);
792 if (s
->duration
== AV_NOPTS_VALUE
)
793 s
->ctx_flags
|= AVFMTCTX_UNSEEKABLE
;
797 #endif /* CONFIG_RTPDEC */
799 void ff_rtsp_undo_setup(AVFormatContext
*s
, int send_packets
)
801 RTSPState
*rt
= s
->priv_data
;
804 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
805 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
808 if (rtsp_st
->transport_priv
) {
810 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
811 av_write_trailer(rtpctx
);
812 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
813 if (CONFIG_RTSP_MUXER
&& rtpctx
->pb
&& send_packets
)
814 ff_rtsp_tcp_write_packet(s
, rtsp_st
);
815 ffio_free_dyn_buf(&rtpctx
->pb
);
817 avio_closep(&rtpctx
->pb
);
819 avformat_free_context(rtpctx
);
820 } else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RDT
)
821 ff_rdt_parse_close(rtsp_st
->transport_priv
);
822 else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
823 ff_rtp_parse_close(rtsp_st
->transport_priv
);
825 rtsp_st
->transport_priv
= NULL
;
826 ffurl_closep(&rtsp_st
->rtp_handle
);
830 /* close and free RTSP streams */
831 void ff_rtsp_close_streams(AVFormatContext
*s
)
833 RTSPState
*rt
= s
->priv_data
;
837 ff_rtsp_undo_setup(s
, 0);
838 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
839 rtsp_st
= rt
->rtsp_streams
[i
];
841 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
) {
842 if (rtsp_st
->dynamic_handler
->close
)
843 rtsp_st
->dynamic_handler
->close(
844 rtsp_st
->dynamic_protocol_context
);
845 av_free(rtsp_st
->dynamic_protocol_context
);
847 for (j
= 0; j
< rtsp_st
->nb_include_source_addrs
; j
++)
848 av_freep(&rtsp_st
->include_source_addrs
[j
]);
849 av_freep(&rtsp_st
->include_source_addrs
);
850 for (j
= 0; j
< rtsp_st
->nb_exclude_source_addrs
; j
++)
851 av_freep(&rtsp_st
->exclude_source_addrs
[j
]);
852 av_freep(&rtsp_st
->exclude_source_addrs
);
857 av_freep(&rt
->rtsp_streams
);
859 avformat_close_input(&rt
->asf_ctx
);
861 if (CONFIG_RTPDEC
&& rt
->ts
)
862 avpriv_mpegts_parse_close(rt
->ts
);
864 av_freep(&rt
->recvbuf
);
867 int ff_rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
869 RTSPState
*rt
= s
->priv_data
;
871 int reordering_queue_size
= rt
->reordering_queue_size
;
872 if (reordering_queue_size
< 0) {
873 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
|| !s
->max_delay
)
874 reordering_queue_size
= 0;
876 reordering_queue_size
= RTP_REORDER_QUEUE_DEFAULT_SIZE
;
879 /* open the RTP context */
880 if (rtsp_st
->stream_index
>= 0)
881 st
= s
->streams
[rtsp_st
->stream_index
];
883 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
885 if (CONFIG_RTSP_MUXER
&& s
->oformat
&& st
) {
886 int ret
= ff_rtp_chain_mux_open((AVFormatContext
**)&rtsp_st
->transport_priv
,
887 s
, st
, rtsp_st
->rtp_handle
,
889 rtsp_st
->stream_index
);
890 /* Ownership of rtp_handle is passed to the rtp mux context */
891 rtsp_st
->rtp_handle
= NULL
;
894 st
->time_base
= ((AVFormatContext
*)rtsp_st
->transport_priv
)->streams
[0]->time_base
;
895 } else if (rt
->transport
== RTSP_TRANSPORT_RAW
) {
896 return 0; // Don't need to open any parser here
897 } else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RDT
&& st
)
898 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
899 rtsp_st
->dynamic_protocol_context
,
900 rtsp_st
->dynamic_handler
);
901 else if (CONFIG_RTPDEC
)
902 rtsp_st
->transport_priv
= ff_rtp_parse_open(s
, st
,
903 rtsp_st
->sdp_payload_type
,
904 reordering_queue_size
);
906 if (!rtsp_st
->transport_priv
) {
907 return AVERROR(ENOMEM
);
908 } else if (CONFIG_RTPDEC
&& rt
->transport
== RTSP_TRANSPORT_RTP
&&
910 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
911 rtpctx
->ssrc
= rtsp_st
->ssrc
;
912 if (rtsp_st
->dynamic_handler
) {
913 ff_rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
914 rtsp_st
->dynamic_protocol_context
,
915 rtsp_st
->dynamic_handler
);
917 if (rtsp_st
->crypto_suite
[0])
918 ff_rtp_parse_set_crypto(rtsp_st
->transport_priv
,
919 rtsp_st
->crypto_suite
,
920 rtsp_st
->crypto_params
);
926 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
927 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
934 q
+= strspn(q
, SPACE_CHARS
);
935 v
= strtol(q
, &p
, 10);
939 v
= strtol(p
, &p
, 10);
948 /* XXX: only one transport specification is parsed */
949 static void rtsp_parse_transport(AVFormatContext
*s
,
950 RTSPMessageHeader
*reply
, const char *p
)
952 char transport_protocol
[16];
954 char lower_transport
[16];
956 RTSPTransportField
*th
;
959 reply
->nb_transports
= 0;
962 p
+= strspn(p
, SPACE_CHARS
);
966 th
= &reply
->transports
[reply
->nb_transports
];
968 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
970 if (!av_strcasecmp (transport_protocol
, "rtp")) {
971 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
972 lower_transport
[0] = '\0';
973 /* rtp/avp/<protocol> */
975 get_word_sep(lower_transport
, sizeof(lower_transport
),
978 th
->transport
= RTSP_TRANSPORT_RTP
;
979 } else if (!av_strcasecmp (transport_protocol
, "x-pn-tng") ||
980 !av_strcasecmp (transport_protocol
, "x-real-rdt")) {
981 /* x-pn-tng/<protocol> */
982 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
984 th
->transport
= RTSP_TRANSPORT_RDT
;
985 } else if (!av_strcasecmp(transport_protocol
, "raw")) {
986 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
987 lower_transport
[0] = '\0';
988 /* raw/raw/<protocol> */
990 get_word_sep(lower_transport
, sizeof(lower_transport
),
993 th
->transport
= RTSP_TRANSPORT_RAW
;
997 if (!av_strcasecmp(lower_transport
, "TCP"))
998 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
1000 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
1004 /* get each parameter */
1005 while (*p
!= '\0' && *p
!= ',') {
1006 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
1007 if (!strcmp(parameter
, "port")) {
1010 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
1012 } else if (!strcmp(parameter
, "client_port")) {
1015 rtsp_parse_range(&th
->client_port_min
,
1016 &th
->client_port_max
, &p
);
1018 } else if (!strcmp(parameter
, "server_port")) {
1021 rtsp_parse_range(&th
->server_port_min
,
1022 &th
->server_port_max
, &p
);
1024 } else if (!strcmp(parameter
, "interleaved")) {
1027 rtsp_parse_range(&th
->interleaved_min
,
1028 &th
->interleaved_max
, &p
);
1030 } else if (!strcmp(parameter
, "multicast")) {
1031 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
1032 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
1033 } else if (!strcmp(parameter
, "ttl")) {
1037 th
->ttl
= strtol(p
, &end
, 10);
1040 } else if (!strcmp(parameter
, "destination")) {
1043 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
1044 get_sockaddr(s
, buf
, &th
->destination
);
1046 } else if (!strcmp(parameter
, "source")) {
1049 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
1050 av_strlcpy(th
->source
, buf
, sizeof(th
->source
));
1052 } else if (!strcmp(parameter
, "mode")) {
1055 get_word_sep(buf
, sizeof(buf
), ";, ", &p
);
1056 if (!av_strcasecmp(buf
, "record") ||
1057 !av_strcasecmp(buf
, "receive"))
1058 th
->mode_record
= 1;
1062 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
1070 reply
->nb_transports
++;
1071 if (reply
->nb_transports
>= RTSP_MAX_TRANSPORTS
)
1076 static void handle_rtp_info(RTSPState
*rt
, const char *url
,
1077 uint32_t seq
, uint32_t rtptime
)
1080 if (!rtptime
|| !url
[0])
1082 if (rt
->transport
!= RTSP_TRANSPORT_RTP
)
1084 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1085 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
1086 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1089 if (!strcmp(rtsp_st
->control_url
, url
)) {
1090 rtpctx
->base_timestamp
= rtptime
;
1096 static void rtsp_parse_rtp_info(RTSPState
*rt
, const char *p
)
1099 char key
[20], value
[MAX_URL_SIZE
], url
[MAX_URL_SIZE
] = "";
1100 uint32_t seq
= 0, rtptime
= 0;
1103 p
+= strspn(p
, SPACE_CHARS
);
1106 get_word_sep(key
, sizeof(key
), "=", &p
);
1110 get_word_sep(value
, sizeof(value
), ";, ", &p
);
1112 if (!strcmp(key
, "url"))
1113 av_strlcpy(url
, value
, sizeof(url
));
1114 else if (!strcmp(key
, "seq"))
1115 seq
= strtoul(value
, NULL
, 10);
1116 else if (!strcmp(key
, "rtptime"))
1117 rtptime
= strtoul(value
, NULL
, 10);
1119 handle_rtp_info(rt
, url
, seq
, rtptime
);
1128 handle_rtp_info(rt
, url
, seq
, rtptime
);
1131 void ff_rtsp_parse_line(AVFormatContext
*s
,
1132 RTSPMessageHeader
*reply
, const char *buf
,
1133 RTSPState
*rt
, const char *method
)
1137 /* NOTE: we do case independent match for broken servers */
1139 if (av_stristart(p
, "Session:", &p
)) {
1141 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
1142 if (av_stristart(p
, ";timeout=", &p
) &&
1143 (t
= strtol(p
, NULL
, 10)) > 0) {
1146 } else if (av_stristart(p
, "Content-Length:", &p
)) {
1147 reply
->content_length
= strtol(p
, NULL
, 10);
1148 } else if (av_stristart(p
, "Transport:", &p
)) {
1149 rtsp_parse_transport(s
, reply
, p
);
1150 } else if (av_stristart(p
, "CSeq:", &p
)) {
1151 reply
->seq
= strtol(p
, NULL
, 10);
1152 } else if (av_stristart(p
, "Range:", &p
)) {
1153 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
1154 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
1155 p
+= strspn(p
, SPACE_CHARS
);
1156 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
1157 } else if (av_stristart(p
, "Server:", &p
)) {
1158 p
+= strspn(p
, SPACE_CHARS
);
1159 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
1160 } else if (av_stristart(p
, "Notice:", &p
) ||
1161 av_stristart(p
, "X-Notice:", &p
)) {
1162 reply
->notice
= strtol(p
, NULL
, 10);
1163 } else if (av_stristart(p
, "Location:", &p
)) {
1164 p
+= strspn(p
, SPACE_CHARS
);
1165 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
1166 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && rt
) {
1167 p
+= strspn(p
, SPACE_CHARS
);
1168 ff_http_auth_handle_header(&rt
->auth_state
, "WWW-Authenticate", p
);
1169 } else if (av_stristart(p
, "Authentication-Info:", &p
) && rt
) {
1170 p
+= strspn(p
, SPACE_CHARS
);
1171 ff_http_auth_handle_header(&rt
->auth_state
, "Authentication-Info", p
);
1172 } else if (av_stristart(p
, "Content-Base:", &p
) && rt
) {
1173 p
+= strspn(p
, SPACE_CHARS
);
1174 if (method
&& !strcmp(method
, "DESCRIBE"))
1175 av_strlcpy(rt
->control_uri
, p
, sizeof(rt
->control_uri
));
1176 } else if (av_stristart(p
, "RTP-Info:", &p
) && rt
) {
1177 p
+= strspn(p
, SPACE_CHARS
);
1178 if (method
&& !strcmp(method
, "PLAY"))
1179 rtsp_parse_rtp_info(rt
, p
);
1180 } else if (av_stristart(p
, "Public:", &p
) && rt
) {
1181 if (strstr(p
, "GET_PARAMETER") &&
1182 method
&& !strcmp(method
, "OPTIONS"))
1183 rt
->get_parameter_supported
= 1;
1184 } else if (av_stristart(p
, "x-Accept-Dynamic-Rate:", &p
) && rt
) {
1185 p
+= strspn(p
, SPACE_CHARS
);
1186 rt
->accept_dynamic_rate
= atoi(p
);
1187 } else if (av_stristart(p
, "Content-Type:", &p
)) {
1188 p
+= strspn(p
, SPACE_CHARS
);
1189 av_strlcpy(reply
->content_type
, p
, sizeof(reply
->content_type
));
1190 } else if (av_stristart(p
, "com.ses.streamID:", &p
)) {
1191 p
+= strspn(p
, SPACE_CHARS
);
1192 av_strlcpy(reply
->stream_id
, p
, sizeof(reply
->stream_id
));
1196 /* skip a RTP/TCP interleaved packet */
1197 int ff_rtsp_skip_packet(AVFormatContext
*s
)
1199 RTSPState
*rt
= s
->priv_data
;
1201 uint8_t buf
[MAX_URL_SIZE
];
1203 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, 3);
1205 return ret
< 0 ? ret
: AVERROR(EIO
);
1206 len
= AV_RB16(buf
+ 1);
1208 av_log(s
, AV_LOG_TRACE
, "skipping RTP packet len=%d\n", len
);
1213 if (len1
> sizeof(buf
))
1215 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, len1
);
1217 return ret
< 0 ? ret
: AVERROR(EIO
);
1224 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
1225 unsigned char **content_ptr
,
1226 int return_on_interleaved_data
, const char *method
)
1228 RTSPState
*rt
= s
->priv_data
;
1229 char buf
[MAX_URL_SIZE
], buf1
[MAX_URL_SIZE
], *q
;
1232 int ret
, content_length
, line_count
, request
;
1233 unsigned char *content
;
1239 memset(reply
, 0, sizeof(*reply
));
1241 /* parse reply (XXX: use buffers) */
1242 rt
->last_reply
[0] = '\0';
1246 ret
= ffurl_read_complete(rt
->rtsp_hd
, &ch
, 1);
1248 ret
= (ret
< 0) ? ret
: AVERROR(EIO
);
1249 av_log(s
, AV_LOG_WARNING
, "Failed reading RTSP data: %s\n", av_err2str(ret
));
1252 av_log(s
, AV_LOG_TRACE
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
1255 if (ch
== '$' && q
== buf
) {
1256 if (return_on_interleaved_data
) {
1259 ret
= ff_rtsp_skip_packet(s
);
1263 } else if (ch
!= '\r') {
1264 if ((q
- buf
) < sizeof(buf
) - 1)
1270 av_log(s
, AV_LOG_TRACE
, "line='%s'\n", buf
);
1272 /* test if last line */
1276 if (line_count
== 0) {
1277 /* get reply code */
1278 get_word(buf1
, sizeof(buf1
), &p
);
1279 if (!strncmp(buf1
, "RTSP/", 5)) {
1280 get_word(buf1
, sizeof(buf1
), &p
);
1281 reply
->status_code
= atoi(buf1
);
1282 p
+= strspn(p
, SPACE_CHARS
);
1283 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
1285 av_strlcpy(reply
->reason
, buf1
, sizeof(reply
->reason
)); // method
1286 get_word(buf1
, sizeof(buf1
), &p
); // object
1290 ff_rtsp_parse_line(s
, reply
, p
, rt
, method
);
1291 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
1292 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
1297 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0' && !request
)
1298 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
1300 content_length
= reply
->content_length
;
1301 if (content_length
> 0) {
1302 /* leave some room for a trailing '\0' (useful for simple parsing) */
1303 content
= av_malloc(content_length
+ 1);
1305 return AVERROR(ENOMEM
);
1306 if ((ret
= ffurl_read_complete(rt
->rtsp_hd
, content
, content_length
)) != content_length
) {
1308 return ret
< 0 ? ret
: AVERROR(EIO
);
1310 content
[content_length
] = '\0';
1313 *content_ptr
= content
;
1318 char buf
[MAX_URL_SIZE
];
1319 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1320 const char* ptr
= buf
;
1322 if (!strcmp(reply
->reason
, "OPTIONS") ||
1323 !strcmp(reply
->reason
, "GET_PARAMETER")) {
1324 snprintf(buf
, sizeof(buf
), "RTSP/1.0 200 OK\r\n");
1326 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", reply
->seq
);
1327 if (reply
->session_id
[0])
1328 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n",
1331 snprintf(buf
, sizeof(buf
), "RTSP/1.0 501 Not Implemented\r\n");
1333 av_strlcat(buf
, "\r\n", sizeof(buf
));
1335 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1336 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1339 ffurl_write(rt
->rtsp_hd_out
, ptr
, strlen(ptr
));
1341 rt
->last_cmd_time
= av_gettime_relative();
1342 /* Even if the request from the server had data, it is not the data
1343 * that the caller wants or expects. The memory could also be leaked
1344 * if the actual following reply has content data. */
1346 av_freep(content_ptr
);
1347 /* If method is set, this is called from ff_rtsp_send_cmd,
1348 * where a reply to exactly this request is awaited. For
1349 * callers from within packet receiving, we just want to
1350 * return to the caller and go back to receiving packets. */
1356 if (rt
->seq
!= reply
->seq
) {
1357 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
1358 rt
->seq
, reply
->seq
);
1362 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
1363 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
1364 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
1365 rt
->state
= RTSP_STATE_IDLE
;
1366 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
1367 return AVERROR(EIO
); /* data or server error */
1368 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
1369 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
1370 return AVERROR(EPERM
);
1376 * Send a command to the RTSP server without waiting for the reply.
1378 * @param s RTSP (de)muxer context
1379 * @param method the method for the request
1380 * @param url the target url for the request
1381 * @param headers extra header lines to include in the request
1382 * @param send_content if non-null, the data to send as request body content
1383 * @param send_content_length the length of the send_content data, or 0 if
1384 * send_content is null
1386 * @return zero if success, nonzero otherwise
1388 static int rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
1389 const char *method
, const char *url
,
1390 const char *headers
,
1391 const unsigned char *send_content
,
1392 int send_content_length
)
1394 RTSPState
*rt
= s
->priv_data
;
1395 char buf
[MAX_URL_SIZE
], *out_buf
;
1396 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1398 if (!rt
->rtsp_hd_out
)
1399 return AVERROR(ENOTCONN
);
1401 /* Add in RTSP headers */
1404 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
1406 av_strlcat(buf
, headers
, sizeof(buf
));
1407 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
1408 av_strlcatf(buf
, sizeof(buf
), "User-Agent: %s\r\n", rt
->user_agent
);
1409 if (rt
->session_id
[0] != '\0' && (!headers
||
1410 !strstr(headers
, "\nIf-Match:"))) {
1411 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
1414 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
1415 rt
->auth
, url
, method
);
1417 av_strlcat(buf
, str
, sizeof(buf
));
1420 if (send_content_length
> 0 && send_content
)
1421 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
1422 av_strlcat(buf
, "\r\n", sizeof(buf
));
1424 /* base64 encode rtsp if tunneling */
1425 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1426 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1427 out_buf
= base64buf
;
1430 av_log(s
, AV_LOG_TRACE
, "Sending:\n%s--\n", buf
);
1432 ffurl_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
1433 if (send_content_length
> 0 && send_content
) {
1434 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1435 avpriv_report_missing_feature(s
, "Tunneling of RTSP requests with content data");
1436 return AVERROR_PATCHWELCOME
;
1438 ffurl_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
1440 rt
->last_cmd_time
= av_gettime_relative();
1445 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
1446 const char *url
, const char *headers
)
1448 return rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
1451 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
1452 const char *headers
, RTSPMessageHeader
*reply
,
1453 unsigned char **content_ptr
)
1455 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
1456 content_ptr
, NULL
, 0);
1459 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
1460 const char *method
, const char *url
,
1462 RTSPMessageHeader
*reply
,
1463 unsigned char **content_ptr
,
1464 const unsigned char *send_content
,
1465 int send_content_length
)
1467 RTSPState
*rt
= s
->priv_data
;
1468 HTTPAuthType cur_auth_type
;
1469 int ret
, attempts
= 0;
1472 cur_auth_type
= rt
->auth_state
.auth_type
;
1473 if ((ret
= rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
1475 send_content_length
)) < 0)
1478 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0, method
) ) < 0)
1482 if (reply
->status_code
== 401 &&
1483 (cur_auth_type
== HTTP_AUTH_NONE
|| rt
->auth_state
.stale
) &&
1484 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
&& attempts
< 2)
1487 if (reply
->status_code
> 400){
1488 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
1492 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
1498 int ff_rtsp_make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
1499 int lower_transport
, const char *real_challenge
)
1501 RTSPState
*rt
= s
->priv_data
;
1502 int rtx
= 0, j
, i
, err
, interleave
= 0, port_off
= 0;
1503 RTSPStream
*rtsp_st
;
1504 RTSPMessageHeader reply1
, *reply
= &reply1
;
1505 char cmd
[MAX_URL_SIZE
];
1506 const char *trans_pref
;
1508 memset(&reply1
, 0, sizeof(reply1
));
1510 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
1511 trans_pref
= "x-pn-tng";
1512 else if (rt
->transport
== RTSP_TRANSPORT_RAW
)
1513 trans_pref
= "RAW/RAW";
1515 trans_pref
= "RTP/AVP";
1517 /* default timeout: 1 minute */
1520 /* Choose a random starting offset within the first half of the
1521 * port range, to allow for a number of ports to try even if the offset
1522 * happens to be at the end of the random range. */
1523 if (rt
->rtp_port_max
- rt
->rtp_port_min
>= 4) {
1524 port_off
= av_get_random_seed() % ((rt
->rtp_port_max
- rt
->rtp_port_min
)/2);
1525 /* even random offset */
1526 port_off
-= port_off
& 0x01;
1529 for (j
= rt
->rtp_port_min
+ port_off
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1530 char transport
[MAX_URL_SIZE
];
1533 * WMS serves all UDP data over a single connection, the RTX, which
1534 * isn't necessarily the first in the SDP but has to be the first
1535 * to be set up, else the second/third SETUP will fail with a 461.
1537 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1538 rt
->server_type
== RTSP_SERVER_WMS
) {
1541 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1542 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1544 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1548 if (rtx
== rt
->nb_rtsp_streams
)
1549 return -1; /* no RTX found */
1550 rtsp_st
= rt
->rtsp_streams
[rtx
];
1552 rtsp_st
= rt
->rtsp_streams
[i
> rtx
? i
: i
- 1];
1554 rtsp_st
= rt
->rtsp_streams
[i
];
1557 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1560 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1561 port
= reply
->transports
[0].client_port_min
;
1565 /* first try in specified port range */
1566 while (j
+ 1 <= rt
->rtp_port_max
) {
1567 AVDictionary
*opts
= map_to_opts(rt
);
1569 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1570 "?localport=%d", j
);
1571 /* we will use two ports per rtp stream (rtp and rtcp) */
1573 err
= ffurl_open_whitelist(&rtsp_st
->rtp_handle
, buf
, AVIO_FLAG_READ_WRITE
,
1574 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
1576 av_dict_free(&opts
);
1581 av_log(s
, AV_LOG_ERROR
, "Unable to open an input RTP port\n");
1586 port
= ff_rtp_get_local_rtp_port(rtsp_st
->rtp_handle
);
1588 av_strlcpy(transport
, trans_pref
, sizeof(transport
));
1589 av_strlcat(transport
,
1590 rt
->server_type
== RTSP_SERVER_SATIP
? ";" : "/UDP;",
1592 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1593 av_strlcat(transport
, "unicast;", sizeof(transport
));
1594 av_strlcatf(transport
, sizeof(transport
),
1595 "client_port=%d", port
);
1596 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1597 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1598 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1602 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1603 /* For WMS streams, the application streams are only used for
1604 * UDP. When trying to set it up for TCP streams, the server
1605 * will return an error. Therefore, we skip those streams. */
1606 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1607 (rtsp_st
->stream_index
< 0 ||
1608 s
->streams
[rtsp_st
->stream_index
]->codecpar
->codec_type
==
1611 snprintf(transport
, sizeof(transport
) - 1,
1612 "%s/TCP;", trans_pref
);
1613 if (rt
->transport
!= RTSP_TRANSPORT_RDT
)
1614 av_strlcat(transport
, "unicast;", sizeof(transport
));
1615 av_strlcatf(transport
, sizeof(transport
),
1616 "interleaved=%d-%d",
1617 interleave
, interleave
+ 1);
1621 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1622 snprintf(transport
, sizeof(transport
) - 1,
1623 "%s/UDP;multicast", trans_pref
);
1625 err
= AVERROR(EINVAL
);
1626 goto fail
; // transport would be uninitialized
1630 av_strlcat(transport
, ";mode=record", sizeof(transport
));
1631 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1632 rt
->server_type
== RTSP_SERVER_WMS
)
1633 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1634 snprintf(cmd
, sizeof(cmd
),
1635 "Transport: %s\r\n",
1637 if (rt
->accept_dynamic_rate
)
1638 av_strlcat(cmd
, "x-Dynamic-Rate: 0\r\n", sizeof(cmd
));
1639 if (CONFIG_RTPDEC
&& i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
) {
1640 char real_res
[41], real_csum
[9];
1641 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1643 av_strlcatf(cmd
, sizeof(cmd
),
1645 "RealChallenge2: %s, sd=%s\r\n",
1646 rt
->session_id
, real_res
, real_csum
);
1648 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1649 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1652 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1653 reply
->nb_transports
!= 1) {
1654 err
= ff_rtsp_averror(reply
->status_code
, AVERROR_INVALIDDATA
);
1658 if (rt
->server_type
== RTSP_SERVER_SATIP
&& reply
->stream_id
[0]) {
1659 char proto
[128], host
[128], path
[512], auth
[128];
1661 av_url_split(proto
, sizeof(proto
), auth
, sizeof(auth
), host
, sizeof(host
),
1662 &port
, path
, sizeof(path
), rt
->control_uri
);
1663 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), proto
, NULL
, host
,
1664 port
, "/stream=%s", reply
->stream_id
);
1667 /* XXX: same protocol for all streams is required */
1669 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1670 reply
->transports
[0].transport
!= rt
->transport
) {
1671 err
= AVERROR_INVALIDDATA
;
1675 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1676 rt
->transport
= reply
->transports
[0].transport
;
1679 /* Fail if the server responded with another lower transport mode
1680 * than what we requested. */
1681 if (reply
->transports
[0].lower_transport
!= lower_transport
) {
1682 av_log(s
, AV_LOG_ERROR
, "Nonmatching transport in server reply\n");
1683 err
= AVERROR_INVALIDDATA
;
1687 switch(reply
->transports
[0].lower_transport
) {
1688 case RTSP_LOWER_TRANSPORT_TCP
:
1689 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1690 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1693 case RTSP_LOWER_TRANSPORT_UDP
: {
1694 char url
[MAX_URL_SIZE
], options
[30] = "";
1695 const char *peer
= host
;
1697 if (rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
)
1698 av_strlcpy(options
, "?connect=1", sizeof(options
));
1699 /* Use source address if specified */
1700 if (reply
->transports
[0].source
[0])
1701 peer
= reply
->transports
[0].source
;
1702 ff_url_join(url
, sizeof(url
), "rtp", NULL
, peer
,
1703 reply
->transports
[0].server_port_min
, "%s", options
);
1704 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1705 ff_rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1706 err
= AVERROR_INVALIDDATA
;
1711 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1712 char url
[MAX_URL_SIZE
], namebuf
[50], optbuf
[20] = "";
1713 struct sockaddr_storage addr
;
1715 AVDictionary
*opts
= map_to_opts(rt
);
1717 if (reply
->transports
[0].destination
.ss_family
) {
1718 addr
= reply
->transports
[0].destination
;
1719 port
= reply
->transports
[0].port_min
;
1720 ttl
= reply
->transports
[0].ttl
;
1722 addr
= rtsp_st
->sdp_ip
;
1723 port
= rtsp_st
->sdp_port
;
1724 ttl
= rtsp_st
->sdp_ttl
;
1727 snprintf(optbuf
, sizeof(optbuf
), "?ttl=%d", ttl
);
1728 getnameinfo((struct sockaddr
*) &addr
, sizeof(addr
),
1729 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1730 ff_url_join(url
, sizeof(url
), "rtp", NULL
, namebuf
,
1731 port
, "%s", optbuf
);
1732 err
= ffurl_open_whitelist(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
,
1733 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
1734 av_dict_free(&opts
);
1737 err
= AVERROR_INVALIDDATA
;
1744 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
1748 if (rt
->nb_rtsp_streams
&& reply
->timeout
> 0)
1749 rt
->timeout
= reply
->timeout
;
1751 if (rt
->server_type
== RTSP_SERVER_REAL
)
1752 rt
->need_subscription
= 1;
1757 ff_rtsp_undo_setup(s
, 0);
1761 void ff_rtsp_close_connections(AVFormatContext
*s
)
1763 RTSPState
*rt
= s
->priv_data
;
1764 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
)
1765 ffurl_closep(&rt
->rtsp_hd_out
);
1766 rt
->rtsp_hd_out
= NULL
;
1767 ffurl_closep(&rt
->rtsp_hd
);
1770 int ff_rtsp_connect(AVFormatContext
*s
)
1772 RTSPState
*rt
= s
->priv_data
;
1773 char proto
[128], host
[1024], path
[2048];
1774 char tcpname
[1024], cmd
[MAX_URL_SIZE
], auth
[128];
1775 const char *lower_rtsp_proto
= "tcp";
1776 int port
, err
, tcp_fd
;
1777 RTSPMessageHeader reply1
, *reply
= &reply1
;
1778 int lower_transport_mask
= 0;
1779 int default_port
= RTSP_DEFAULT_PORT
;
1780 int https_tunnel
= 0;
1781 char real_challenge
[64] = "";
1782 struct sockaddr_storage peer
;
1783 socklen_t peer_len
= sizeof(peer
);
1785 if (rt
->rtp_port_max
< rt
->rtp_port_min
) {
1786 av_log(s
, AV_LOG_ERROR
, "Invalid UDP port range, max port %d less "
1787 "than min port %d\n", rt
->rtp_port_max
,
1789 return AVERROR(EINVAL
);
1792 if (!ff_network_init())
1793 return AVERROR(EIO
);
1795 if (s
->max_delay
< 0) /* Not set by the caller */
1796 s
->max_delay
= s
->iformat
? DEFAULT_REORDERING_DELAY
: 0;
1798 rt
->control_transport
= RTSP_MODE_PLAIN
;
1799 if (rt
->lower_transport_mask
& ((1 << RTSP_LOWER_TRANSPORT_HTTP
) |
1800 (1 << RTSP_LOWER_TRANSPORT_HTTPS
))) {
1801 https_tunnel
= !!(rt
->lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_HTTPS
));
1802 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1803 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1805 /* Only pass through valid flags from here */
1806 rt
->lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1809 memset(&reply1
, 0, sizeof(reply1
));
1810 /* extract hostname and port */
1811 av_url_split(proto
, sizeof(proto
), auth
, sizeof(auth
),
1812 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->url
);
1814 if (!strcmp(proto
, "rtsps")) {
1815 lower_rtsp_proto
= "tls";
1816 default_port
= RTSPS_DEFAULT_PORT
;
1817 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1818 } else if (!strcmp(proto
, "satip")) {
1819 av_strlcpy(proto
, "rtsp", sizeof(proto
));
1820 rt
->server_type
= RTSP_SERVER_SATIP
;
1824 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1827 port
= default_port
;
1829 lower_transport_mask
= rt
->lower_transport_mask
;
1831 if (!lower_transport_mask
)
1832 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1835 /* Only UDP or TCP - UDP multicast isn't supported. */
1836 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1837 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1838 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1839 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1840 "only UDP and TCP are supported for output.\n");
1841 err
= AVERROR(EINVAL
);
1846 /* Construct the URI used in request; this is similar to s->url,
1847 * but with authentication credentials removed and RTSP specific options
1849 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), proto
, NULL
,
1850 host
, port
, "%s", path
);
1852 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1853 /* set up initial handshake for tunneling */
1854 char httpname
[1024];
1855 char sessioncookie
[17];
1857 AVDictionary
*options
= NULL
;
1859 av_dict_set_int(&options
, "timeout", rt
->stimeout
, 0);
1861 int ret
= copy_tls_opts_dict(rt
, &options
);
1863 av_dict_free(&options
);
1869 ff_url_join(httpname
, sizeof(httpname
), https_tunnel
? "https" : "http", auth
, host
, port
, "%s", path
);
1870 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1871 av_get_random_seed(), av_get_random_seed());
1874 if (ffurl_alloc(&rt
->rtsp_hd
, httpname
, AVIO_FLAG_READ
,
1875 &s
->interrupt_callback
) < 0) {
1876 av_dict_free(&options
);
1881 /* generate GET headers */
1882 snprintf(headers
, sizeof(headers
),
1883 "x-sessioncookie: %s\r\n"
1884 "Accept: application/x-rtsp-tunnelled\r\n"
1885 "Pragma: no-cache\r\n"
1886 "Cache-Control: no-cache\r\n",
1888 av_opt_set(rt
->rtsp_hd
->priv_data
, "headers", headers
, 0);
1890 if (!rt
->rtsp_hd
->protocol_whitelist
&& s
->protocol_whitelist
) {
1891 rt
->rtsp_hd
->protocol_whitelist
= av_strdup(s
->protocol_whitelist
);
1892 if (!rt
->rtsp_hd
->protocol_whitelist
) {
1893 av_dict_free(&options
);
1894 err
= AVERROR(ENOMEM
);
1899 /* complete the connection */
1900 if (ffurl_connect(rt
->rtsp_hd
, &options
)) {
1901 av_dict_free(&options
);
1907 if (ffurl_alloc(&rt
->rtsp_hd_out
, httpname
, AVIO_FLAG_WRITE
,
1908 &s
->interrupt_callback
) < 0 ) {
1909 av_dict_free(&options
);
1914 /* generate POST headers */
1915 snprintf(headers
, sizeof(headers
),
1916 "x-sessioncookie: %s\r\n"
1917 "Content-Type: application/x-rtsp-tunnelled\r\n"
1918 "Pragma: no-cache\r\n"
1919 "Cache-Control: no-cache\r\n"
1920 "Content-Length: 32767\r\n"
1921 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1923 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "headers", headers
, 0);
1924 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "chunked_post", "0", 0);
1925 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "send_expect_100", "0", 0);
1927 /* Initialize the authentication state for the POST session. The HTTP
1928 * protocol implementation doesn't properly handle multi-pass
1929 * authentication for POST requests, since it would require one of
1931 * - implementing Expect: 100-continue, which many HTTP servers
1932 * don't support anyway, even less the RTSP servers that do HTTP
1934 * - sending the whole POST data until getting a 401 reply specifying
1935 * what authentication method to use, then resending all that data
1936 * - waiting for potential 401 replies directly after sending the
1937 * POST header (waiting for some unspecified time)
1938 * Therefore, we copy the full auth state, which works for both basic
1939 * and digest. (For digest, we would have to synchronize the nonce
1940 * count variable between the two sessions, if we'd do more requests
1941 * with the original session, though.)
1943 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1945 /* complete the connection */
1946 if (ffurl_connect(rt
->rtsp_hd_out
, &options
)) {
1947 av_dict_free(&options
);
1951 av_dict_free(&options
);
1954 /* open the tcp connection */
1955 AVDictionary
*proto_opts
= NULL
;
1956 if (strcmp("tls", lower_rtsp_proto
) == 0) {
1957 ret
= copy_tls_opts_dict(rt
, &proto_opts
);
1959 av_dict_free(&proto_opts
);
1965 ff_url_join(tcpname
, sizeof(tcpname
), lower_rtsp_proto
, NULL
,
1967 "?timeout=%"PRId64
, rt
->stimeout
);
1968 if ((ret
= ffurl_open_whitelist(&rt
->rtsp_hd
, tcpname
, AVIO_FLAG_READ_WRITE
,
1969 &s
->interrupt_callback
, &proto_opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
)) < 0) {
1970 av_dict_free(&proto_opts
);
1974 av_dict_free(&proto_opts
);
1975 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1979 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1984 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1985 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1986 NULL
, 0, NI_NUMERICHOST
);
1989 /* request options supported by the server; this also detects server
1991 if (rt
->server_type
!= RTSP_SERVER_SATIP
)
1992 rt
->server_type
= RTSP_SERVER_RTP
;
1995 if (rt
->server_type
== RTSP_SERVER_REAL
)
1998 * The following entries are required for proper
1999 * streaming from a Realmedia server. They are
2000 * interdependent in some way although we currently
2001 * don't quite understand how. Values were copied
2002 * from mplayer SVN r23589.
2003 * ClientChallenge is a 16-byte ID in hex
2004 * CompanyID is a 16-byte ID in base64
2006 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
2007 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
2008 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
2009 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
2011 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
2012 if (reply
->status_code
!= RTSP_STATUS_OK
) {
2013 err
= ff_rtsp_averror(reply
->status_code
, AVERROR_INVALIDDATA
);
2017 /* detect server type if not standard-compliant RTP */
2018 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
2019 rt
->server_type
= RTSP_SERVER_REAL
;
2021 } else if (!av_strncasecmp(reply
->server
, "WMServer/", 9)) {
2022 rt
->server_type
= RTSP_SERVER_WMS
;
2023 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
2024 strcpy(real_challenge
, reply
->real_challenge
);
2028 #if CONFIG_RTSP_DEMUXER
2030 if (rt
->server_type
== RTSP_SERVER_SATIP
)
2031 err
= init_satip_stream(s
);
2033 err
= ff_rtsp_setup_input_streams(s
, reply
);
2036 if (CONFIG_RTSP_MUXER
)
2037 err
= ff_rtsp_setup_output_streams(s
, host
);
2039 av_unreachable("Either muxer or demuxer must be enabled");
2044 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
2045 ~(lower_transport_mask
- 1)];
2047 if ((lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_TCP
))
2048 && (rt
->rtsp_flags
& RTSP_FLAG_PREFER_TCP
))
2049 lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
2051 err
= ff_rtsp_make_setup_request(s
, host
, port
, lower_transport
,
2052 rt
->server_type
== RTSP_SERVER_REAL
?
2053 real_challenge
: NULL
);
2056 lower_transport_mask
&= ~(1 << lower_transport
);
2057 if (lower_transport_mask
== 0 && err
== 1) {
2058 err
= AVERROR(EPROTONOSUPPORT
);
2063 rt
->lower_transport_mask
= lower_transport_mask
;
2064 av_strlcpy(rt
->real_challenge
, real_challenge
, sizeof(rt
->real_challenge
));
2065 rt
->state
= RTSP_STATE_IDLE
;
2066 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
2069 ff_rtsp_close_streams(s
);
2070 ff_rtsp_close_connections(s
);
2071 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
2072 char *new_url
= av_strdup(reply
->location
);
2074 err
= AVERROR(ENOMEM
);
2077 ff_format_set_url(s
, new_url
);
2078 rt
->session_id
[0] = '\0';
2079 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
2088 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
2091 #if CONFIG_RTSP_DEMUXER
2092 static int parse_rtsp_message(AVFormatContext
*s
)
2094 RTSPState
*rt
= s
->priv_data
;
2097 if (rt
->rtsp_flags
& RTSP_FLAG_LISTEN
) {
2098 if (rt
->state
== RTSP_STATE_STREAMING
) {
2099 return ff_rtsp_parse_streaming_commands(s
);
2103 RTSPMessageHeader reply
;
2104 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0, NULL
);
2107 /* XXX: parse message */
2108 if (rt
->state
!= RTSP_STATE_STREAMING
)
2116 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
2117 uint8_t *buf
, int buf_size
, int64_t wait_end
)
2119 RTSPState
*rt
= s
->priv_data
;
2120 RTSPStream
*rtsp_st
;
2122 struct pollfd
*p
= rt
->p
;
2123 int *fds
= NULL
, fdsnum
, fdsidx
;
2124 int64_t runs
= rt
->stimeout
/ POLLING_TIME
/ 1000;
2127 p
= rt
->p
= av_malloc_array(2 * rt
->nb_rtsp_streams
+ 1, sizeof(*p
));
2129 return AVERROR(ENOMEM
);
2132 p
[rt
->max_p
].fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
2133 p
[rt
->max_p
++].events
= POLLIN
;
2135 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2136 rtsp_st
= rt
->rtsp_streams
[i
];
2137 if (rtsp_st
->rtp_handle
) {
2138 if (ret
= ffurl_get_multi_file_handle(rtsp_st
->rtp_handle
,
2140 av_log(s
, AV_LOG_ERROR
, "Unable to recover rtp ports\n");
2144 av_log(s
, AV_LOG_ERROR
,
2145 "Number of fds %d not supported\n", fdsnum
);
2147 return AVERROR_INVALIDDATA
;
2149 for (fdsidx
= 0; fdsidx
< fdsnum
; fdsidx
++) {
2150 p
[rt
->max_p
].fd
= fds
[fdsidx
];
2151 p
[rt
->max_p
++].events
= POLLIN
;
2159 if (ff_check_interrupt(&s
->interrupt_callback
))
2160 return AVERROR_EXIT
;
2161 if (wait_end
&& wait_end
- av_gettime_relative() < 0)
2162 return AVERROR(EAGAIN
);
2163 n
= poll(p
, rt
->max_p
, POLLING_TIME
);
2165 int j
= rt
->rtsp_hd
? 1 : 0;
2166 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2167 rtsp_st
= rt
->rtsp_streams
[i
];
2168 if (rtsp_st
->rtp_handle
) {
2169 if (p
[j
].revents
& POLLIN
|| p
[j
+1].revents
& POLLIN
) {
2170 ret
= ffurl_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
2172 *prtsp_st
= rtsp_st
;
2179 #if CONFIG_RTSP_DEMUXER
2180 if (rt
->rtsp_hd
&& p
[0].revents
& POLLIN
) {
2181 if ((ret
= parse_rtsp_message(s
)) < 0) {
2186 } else if (n
== 0 && rt
->stimeout
> 0 && --runs
<= 0) {
2187 return AVERROR(ETIMEDOUT
);
2188 } else if (n
< 0 && errno
!= EINTR
)
2189 return AVERROR(errno
);
2193 static int pick_stream(AVFormatContext
*s
, RTSPStream
**rtsp_st
,
2194 const uint8_t *buf
, int len
)
2196 RTSPState
*rt
= s
->priv_data
;
2200 if (rt
->nb_rtsp_streams
== 1) {
2201 *rtsp_st
= rt
->rtsp_streams
[0];
2204 if (len
>= 8 && rt
->transport
== RTSP_TRANSPORT_RTP
) {
2205 if (RTP_PT_IS_RTCP(rt
->recvbuf
[1])) {
2207 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2208 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
2211 if (rtpctx
->ssrc
== AV_RB32(&buf
[4])) {
2212 *rtsp_st
= rt
->rtsp_streams
[i
];
2219 av_log(s
, AV_LOG_WARNING
,
2220 "Unable to pick stream for packet - SSRC not known for "
2222 return AVERROR(EAGAIN
);
2225 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2226 if ((buf
[1] & 0x7f) == rt
->rtsp_streams
[i
]->sdp_payload_type
) {
2227 *rtsp_st
= rt
->rtsp_streams
[i
];
2233 av_log(s
, AV_LOG_WARNING
, "Unable to pick stream for packet\n");
2234 return AVERROR(EAGAIN
);
2237 static int read_packet(AVFormatContext
*s
,
2238 RTSPStream
**rtsp_st
, RTSPStream
*first_queue_st
,
2241 RTSPState
*rt
= s
->priv_data
;
2244 switch(rt
->lower_transport
) {
2246 #if CONFIG_RTSP_DEMUXER
2247 case RTSP_LOWER_TRANSPORT_TCP
:
2248 len
= ff_rtsp_tcp_read_packet(s
, rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
);
2251 case RTSP_LOWER_TRANSPORT_UDP
:
2252 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
2253 len
= udp_read_packet(s
, rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
, wait_end
);
2254 if (len
> 0 && (*rtsp_st
)->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
2255 ff_rtp_check_and_send_back_rr((*rtsp_st
)->transport_priv
, (*rtsp_st
)->rtp_handle
, NULL
, len
);
2257 case RTSP_LOWER_TRANSPORT_CUSTOM
:
2258 if (first_queue_st
&& rt
->transport
== RTSP_TRANSPORT_RTP
&&
2259 wait_end
&& wait_end
< av_gettime_relative())
2260 len
= AVERROR(EAGAIN
);
2262 len
= avio_read_partial(s
->pb
, rt
->recvbuf
, RECVBUF_SIZE
);
2263 len
= pick_stream(s
, rtsp_st
, rt
->recvbuf
, len
);
2264 if (len
> 0 && (*rtsp_st
)->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
2265 ff_rtp_check_and_send_back_rr((*rtsp_st
)->transport_priv
, NULL
, s
->pb
, len
);
2275 int ff_rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
2277 RTSPState
*rt
= s
->priv_data
;
2279 RTSPStream
*rtsp_st
, *first_queue_st
= NULL
;
2280 int64_t wait_end
= 0;
2282 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
2285 /* get next frames from the same RTP packet */
2286 if (rt
->cur_transport_priv
) {
2287 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
2288 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
2289 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
2290 ret
= ff_rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
2291 } else if (CONFIG_RTPDEC
&& rt
->ts
) {
2292 ret
= avpriv_mpegts_parse_packet(rt
->ts
, pkt
, rt
->recvbuf
+ rt
->recvbuf_pos
, rt
->recvbuf_len
- rt
->recvbuf_pos
);
2294 rt
->recvbuf_pos
+= ret
;
2295 ret
= rt
->recvbuf_pos
< rt
->recvbuf_len
;
2300 rt
->cur_transport_priv
= NULL
;
2302 } else if (ret
== 1) {
2305 rt
->cur_transport_priv
= NULL
;
2309 if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
2311 int64_t first_queue_time
= 0;
2312 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2313 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
2317 queue_time
= ff_rtp_queued_packet_time(rtpctx
);
2318 if (queue_time
&& (queue_time
- first_queue_time
< 0 ||
2319 !first_queue_time
)) {
2320 first_queue_time
= queue_time
;
2321 first_queue_st
= rt
->rtsp_streams
[i
];
2324 if (first_queue_time
) {
2325 wait_end
= first_queue_time
+ s
->max_delay
;
2328 first_queue_st
= NULL
;
2332 /* read next RTP packet */
2334 rt
->recvbuf
= av_malloc(RECVBUF_SIZE
);
2336 return AVERROR(ENOMEM
);
2339 len
= read_packet(s
, &rtsp_st
, first_queue_st
, wait_end
);
2340 if (len
== AVERROR(EAGAIN
) && first_queue_st
&&
2341 rt
->transport
== RTSP_TRANSPORT_RTP
) {
2342 av_log(s
, AV_LOG_WARNING
,
2343 "max delay reached. need to consume packet\n");
2344 rtsp_st
= first_queue_st
;
2345 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, NULL
, 0);
2351 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
2352 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
2353 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
2354 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
2355 if (rtsp_st
->feedback
) {
2356 AVIOContext
*pb
= NULL
;
2357 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_CUSTOM
)
2359 ff_rtp_send_rtcp_feedback(rtsp_st
->transport_priv
, rtsp_st
->rtp_handle
, pb
);
2362 /* Either bad packet, or a RTCP packet. Check if the
2363 * first_rtcp_ntp_time field was initialized. */
2364 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
2365 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
2366 /* first_rtcp_ntp_time has been initialized for this stream,
2367 * copy the same value to all other uninitialized streams,
2368 * in order to map their timestamp origin to the same ntp time
2371 AVStream
*st
= NULL
;
2372 if (rtsp_st
->stream_index
>= 0)
2373 st
= s
->streams
[rtsp_st
->stream_index
];
2374 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2375 RTPDemuxContext
*rtpctx2
= rt
->rtsp_streams
[i
]->transport_priv
;
2376 AVStream
*st2
= NULL
;
2377 if (rt
->rtsp_streams
[i
]->stream_index
>= 0)
2378 st2
= s
->streams
[rt
->rtsp_streams
[i
]->stream_index
];
2379 if (rtpctx2
&& st
&& st2
&&
2380 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) {
2381 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
2382 rtpctx2
->rtcp_ts_offset
= av_rescale_q(
2383 rtpctx
->rtcp_ts_offset
, st
->time_base
,
2387 // Make real NTP start time available in AVFormatContext
2388 if (s
->start_time_realtime
== AV_NOPTS_VALUE
) {
2389 s
->start_time_realtime
= ff_parse_ntp_time(rtpctx
->first_rtcp_ntp_time
) - NTP_OFFSET_US
;
2391 s
->start_time_realtime
-=
2392 av_rescale_q (rtpctx
->rtcp_ts_offset
, rtpctx
->st
->time_base
, AV_TIME_BASE_Q
);
2396 if (ret
== -RTCP_BYE
) {
2399 av_log(s
, AV_LOG_DEBUG
, "Received BYE for stream %d (%d/%d)\n",
2400 rtsp_st
->stream_index
, rt
->nb_byes
, rt
->nb_rtsp_streams
);
2402 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
2406 } else if (CONFIG_RTPDEC
&& rt
->ts
) {
2407 ret
= avpriv_mpegts_parse_packet(rt
->ts
, pkt
, rt
->recvbuf
, len
);
2410 rt
->recvbuf_len
= len
;
2411 rt
->recvbuf_pos
= ret
;
2412 rt
->cur_transport_priv
= rt
->ts
;
2419 return AVERROR_INVALIDDATA
;
2425 /* more packets may follow, so we save the RTP context */
2426 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
2430 #endif /* CONFIG_RTPDEC */
2432 #if CONFIG_SDP_DEMUXER
2433 static int sdp_probe(const AVProbeData
*p1
)
2435 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
2437 /* we look for a line beginning "c=IN IP" */
2438 while (p
< p_end
&& *p
!= '\0') {
2439 if (sizeof("c=IN IP") - 1 < p_end
- p
&&
2440 av_strstart(p
, "c=IN IP", NULL
))
2441 return AVPROBE_SCORE_EXTENSION
;
2443 while (p
< p_end
- 1 && *p
!= '\n') p
++;
2452 static void append_source_addrs(char *buf
, int size
, const char *name
,
2453 int count
, struct RTSPSource
**addrs
)
2458 av_strlcatf(buf
, size
, "&%s=%s", name
, addrs
[0]->addr
);
2459 for (i
= 1; i
< count
; i
++)
2460 av_strlcatf(buf
, size
, ",%s", addrs
[i
]->addr
);
2463 static int sdp_read_header(AVFormatContext
*s
)
2465 RTSPState
*rt
= s
->priv_data
;
2466 RTSPStream
*rtsp_st
;
2468 char url
[MAX_URL_SIZE
];
2471 if (!ff_network_init())
2472 return AVERROR(EIO
);
2474 if (s
->max_delay
< 0) /* Not set by the caller */
2475 s
->max_delay
= DEFAULT_REORDERING_DELAY
;
2476 if (rt
->rtsp_flags
& RTSP_FLAG_CUSTOM_IO
)
2477 rt
->lower_transport
= RTSP_LOWER_TRANSPORT_CUSTOM
;
2479 /* read the whole sdp file */
2480 av_bprint_init(&bp
, 0, AV_BPRINT_SIZE_UNLIMITED
);
2481 err
= avio_read_to_bprint(s
->pb
, &bp
, INT_MAX
);
2484 av_bprint_finalize(&bp
, NULL
);
2487 err
= ff_sdp_parse(s
, bp
.str
);
2488 av_bprint_finalize(&bp
, NULL
);
2491 /* open each RTP stream */
2492 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2494 rtsp_st
= rt
->rtsp_streams
[i
];
2496 if (!(rt
->rtsp_flags
& RTSP_FLAG_CUSTOM_IO
)) {
2497 AVDictionary
*opts
= map_to_opts(rt
);
2498 char buf
[MAX_URL_SIZE
];
2501 err
= getnameinfo((struct sockaddr
*) &rtsp_st
->sdp_ip
,
2502 sizeof(rtsp_st
->sdp_ip
),
2503 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
2505 av_log(s
, AV_LOG_ERROR
, "getnameinfo: %s\n", gai_strerror(err
));
2507 av_dict_free(&opts
);
2510 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
2511 namebuf
, rtsp_st
->sdp_port
,
2512 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2513 rtsp_st
->sdp_port
, rtsp_st
->sdp_ttl
,
2514 rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
? 1 : 0,
2515 rt
->rtsp_flags
& RTSP_FLAG_RTCP_TO_SOURCE
? 1 : 0);
2517 p
= strchr(s
->url
, '?');
2518 if (p
&& av_find_info_tag(buf
, sizeof(buf
), "localaddr", p
))
2519 av_strlcatf(url
, sizeof(url
), "&localaddr=%s", buf
);
2520 else if (rt
->localaddr
&& rt
->localaddr
[0])
2521 av_strlcatf(url
, sizeof(url
), "&localaddr=%s", rt
->localaddr
);
2522 append_source_addrs(url
, sizeof(url
), "sources",
2523 rtsp_st
->nb_include_source_addrs
,
2524 rtsp_st
->include_source_addrs
);
2525 append_source_addrs(url
, sizeof(url
), "block",
2526 rtsp_st
->nb_exclude_source_addrs
,
2527 rtsp_st
->exclude_source_addrs
);
2528 err
= ffurl_open_whitelist(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ
,
2529 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
2531 av_dict_free(&opts
);
2534 err
= AVERROR_INVALIDDATA
;
2538 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
2543 ff_rtsp_close_streams(s
);
2548 static int sdp_read_close(AVFormatContext
*s
)
2550 ff_rtsp_close_streams(s
);
2555 static const AVClass sdp_demuxer_class
= {
2556 .class_name
= "SDP demuxer",
2557 .item_name
= av_default_item_name
,
2558 .option
= sdp_options
,
2559 .version
= LIBAVUTIL_VERSION_INT
,
2562 const FFInputFormat ff_sdp_demuxer
= {
2564 .p
.long_name
= NULL_IF_CONFIG_SMALL("SDP"),
2565 .p
.priv_class
= &sdp_demuxer_class
,
2566 .priv_data_size
= sizeof(RTSPState
),
2567 .read_probe
= sdp_probe
,
2568 .read_header
= sdp_read_header
,
2569 .read_packet
= ff_rtsp_fetch_packet
,
2570 .read_close
= sdp_read_close
,
2572 #endif /* CONFIG_SDP_DEMUXER */
2574 #if CONFIG_RTP_DEMUXER
2575 static int rtp_probe(const AVProbeData
*p
)
2577 if (av_strstart(p
->filename
, "rtp:", NULL
))
2578 return AVPROBE_SCORE_MAX
;
2582 static int rtp_read_header(AVFormatContext
*s
)
2584 uint8_t recvbuf
[RTP_MAX_PACKET_LENGTH
];
2585 char host
[500], filters_buf
[1000];
2587 URLContext
* in
= NULL
;
2589 AVCodecParameters
*par
= NULL
;
2590 struct sockaddr_storage addr
;
2592 socklen_t addrlen
= sizeof(addr
);
2593 RTSPState
*rt
= s
->priv_data
;
2596 AVDictionary
*opts
= NULL
;
2598 if (!ff_network_init())
2599 return AVERROR(EIO
);
2601 opts
= map_to_opts(rt
);
2602 ret
= ffurl_open_whitelist(&in
, s
->url
, AVIO_FLAG_READ
,
2603 &s
->interrupt_callback
, &opts
, s
->protocol_whitelist
, s
->protocol_blacklist
, NULL
);
2604 av_dict_free(&opts
);
2609 ret
= ffurl_read(in
, recvbuf
, sizeof(recvbuf
));
2610 if (ret
== AVERROR(EAGAIN
))
2615 av_log(s
, AV_LOG_WARNING
, "Received too short packet\n");
2619 if ((recvbuf
[0] & 0xc0) != 0x80) {
2620 av_log(s
, AV_LOG_WARNING
, "Unsupported RTP version packet "
2625 if (RTP_PT_IS_RTCP(recvbuf
[1]))
2628 payload_type
= recvbuf
[1] & 0x7f;
2631 getsockname(ffurl_get_file_handle(in
), (struct sockaddr
*) &addr
, &addrlen
);
2634 par
= avcodec_parameters_alloc();
2636 ret
= AVERROR(ENOMEM
);
2640 if (ff_rtp_get_codec_info(par
, payload_type
)) {
2641 av_log(s
, AV_LOG_ERROR
, "Unable to receive RTP payload type %d "
2642 "without an SDP file describing it\n",
2644 ret
= AVERROR_INVALIDDATA
;
2647 if (par
->codec_type
!= AVMEDIA_TYPE_DATA
) {
2648 av_log(s
, AV_LOG_WARNING
, "Guessing on RTP content - if not received "
2649 "properly you need an SDP file "
2653 av_url_split(NULL
, 0, NULL
, 0, host
, sizeof(host
), &port
,
2656 av_bprint_init(&sdp
, 0, AV_BPRINT_SIZE_UNLIMITED
);
2657 av_bprintf(&sdp
, "v=0\r\nc=IN IP%d %s\r\n",
2658 addr
.ss_family
== AF_INET
? 4 : 6, host
);
2660 p
= strchr(s
->url
, '?');
2662 static const char filters
[][2][8] = { { "sources", "incl" },
2663 { "block", "excl" } };
2666 for (i
= 0; i
< FF_ARRAY_ELEMS(filters
); i
++) {
2667 if (av_find_info_tag(filters_buf
, sizeof(filters_buf
), filters
[i
][0], p
)) {
2669 while ((q
= strchr(q
, ',')) != NULL
)
2671 av_bprintf(&sdp
, "a=source-filter:%s IN IP%d %s %s\r\n",
2673 addr
.ss_family
== AF_INET
? 4 : 6, host
,
2679 av_bprintf(&sdp
, "m=%s %d RTP/AVP %d\r\n",
2680 par
->codec_type
== AVMEDIA_TYPE_DATA
? "application" :
2681 par
->codec_type
== AVMEDIA_TYPE_VIDEO
? "video" : "audio",
2682 port
, payload_type
);
2683 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
.str
);
2684 if (!av_bprint_is_complete(&sdp
))
2686 avcodec_parameters_free(&par
);
2688 ffio_init_read_context(&pb
, sdp
.str
, sdp
.len
);
2691 /* if sdp_read_header() fails then following ff_network_close() cancels out */
2692 /* ff_network_init() at the start of this function. Otherwise it cancels out */
2693 /* ff_network_init() inside sdp_read_header() */
2696 rt
->media_type_mask
= (1 << (AVMEDIA_TYPE_SUBTITLE
+1)) - 1;
2698 ret
= sdp_read_header(s
);
2700 av_bprint_finalize(&sdp
, NULL
);
2704 ret
= AVERROR(ENOMEM
);
2705 av_log(s
, AV_LOG_ERROR
, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2706 av_bprint_finalize(&sdp
, NULL
);
2708 avcodec_parameters_free(&par
);
2714 static const AVClass rtp_demuxer_class
= {
2715 .class_name
= "RTP demuxer",
2716 .item_name
= av_default_item_name
,
2717 .option
= rtp_options
,
2718 .version
= LIBAVUTIL_VERSION_INT
,
2721 const FFInputFormat ff_rtp_demuxer
= {
2723 .p
.long_name
= NULL_IF_CONFIG_SMALL("RTP input"),
2724 .p
.flags
= AVFMT_NOFILE
,
2725 .p
.priv_class
= &rtp_demuxer_class
,
2726 .priv_data_size
= sizeof(RTSPState
),
2727 .read_probe
= rtp_probe
,
2728 .read_header
= rtp_read_header
,
2729 .read_packet
= ff_rtsp_fetch_packet
,
2730 .read_close
= sdp_read_close
,
2732 #endif /* CONFIG_RTP_DEMUXER */