3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "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" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 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_DATA+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" }
77 #define RTSP_REORDERING_OPTS() \
78 { "reorder_queue_size", "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 const AVOption ff_rtsp_options
[] = {
81 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause
), AV_OPT_TYPE_INT
, {.i64
= 0}, 0, 1, DEC
},
82 FF_RTP_FLAG_OPTS(RTSPState
, rtp_muxer_flags
),
83 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask
), AV_OPT_TYPE_FLAGS
, {.i64
= 0}, INT_MIN
, INT_MAX
, DEC
|ENC
, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_UDP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_TCP
}, 0, 0, DEC
|ENC
, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST
, {.i64
= 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST
}, 0, 0, DEC
, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST
, {.i64
= (1 << RTSP_LOWER_TRANSPORT_HTTP
)}, 0, 0, DEC
, "rtsp_transport" },
88 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
89 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
90 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min
), AV_OPT_TYPE_INT
, {.i64
= RTSP_RTP_PORT_MIN
}, 0, 65535, DEC
|ENC
},
91 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max
), AV_OPT_TYPE_INT
, {.i64
= RTSP_RTP_PORT_MAX
}, 0, 65535, DEC
|ENC
},
92 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout
), AV_OPT_TYPE_INT
, {.i64
= -1}, INT_MIN
, INT_MAX
, DEC
},
93 RTSP_REORDERING_OPTS(),
97 static const AVOption sdp_options
[] = {
98 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
99 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST
, {.i64
= RTSP_FLAG_CUSTOM_IO
}, 0, 0, DEC
, "rtsp_flags" },
100 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
101 RTSP_REORDERING_OPTS(),
105 static const AVOption rtp_options
[] = {
106 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
107 RTSP_REORDERING_OPTS(),
111 static void get_word_until_chars(char *buf
, int buf_size
,
112 const char *sep
, const char **pp
)
118 p
+= strspn(p
, SPACE_CHARS
);
120 while (!strchr(sep
, *p
) && *p
!= '\0') {
121 if ((q
- buf
) < buf_size
- 1)
130 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
133 if (**pp
== '/') (*pp
)++;
134 get_word_until_chars(buf
, buf_size
, sep
, pp
);
137 static void get_word(char *buf
, int buf_size
, const char **pp
)
139 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
142 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
144 * Used for seeking in the rtp stream.
146 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
150 p
+= strspn(p
, SPACE_CHARS
);
151 if (!av_stristart(p
, "npt=", &p
))
154 *start
= AV_NOPTS_VALUE
;
155 *end
= AV_NOPTS_VALUE
;
157 get_word_sep(buf
, sizeof(buf
), "-", &p
);
158 av_parse_time(start
, buf
, 1);
161 get_word_sep(buf
, sizeof(buf
), "-", &p
);
162 av_parse_time(end
, buf
, 1);
166 static int get_sockaddr(const char *buf
, struct sockaddr_storage
*sock
)
168 struct addrinfo hints
= { 0 }, *ai
= NULL
;
169 hints
.ai_flags
= AI_NUMERICHOST
;
170 if (getaddrinfo(buf
, NULL
, &hints
, &ai
))
172 memcpy(sock
, ai
->ai_addr
, FFMIN(sizeof(*sock
), ai
->ai_addrlen
));
178 static void init_rtp_handler(RTPDynamicProtocolHandler
*handler
,
179 RTSPStream
*rtsp_st
, AVCodecContext
*codec
)
184 codec
->codec_id
= handler
->codec_id
;
185 rtsp_st
->dynamic_handler
= handler
;
186 if (handler
->alloc
) {
187 rtsp_st
->dynamic_protocol_context
= handler
->alloc();
188 if (!rtsp_st
->dynamic_protocol_context
)
189 rtsp_st
->dynamic_handler
= NULL
;
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext
*s
,
195 AVStream
*st
, RTSPStream
*rtsp_st
,
196 int payload_type
, const char *p
)
198 AVCodecContext
*codec
= st
->codec
;
204 /* See if we can handle this kind of payload.
205 * The space should normally not be there but some Real streams or
206 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207 * have a trailing space. */
208 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
209 if (payload_type
< RTP_PT_PRIVATE
) {
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
215 if (codec
->codec_id
== AV_CODEC_ID_NONE
) {
216 RTPDynamicProtocolHandler
*handler
=
217 ff_rtp_handler_find_by_name(buf
, codec
->codec_type
);
218 init_rtp_handler(handler
, rtsp_st
, codec
);
219 /* If no dynamic handler was found, check with the list of standard
220 * allocated types, if such a stream for some reason happens to
221 * use a private payload type. This isn't handled in rtpdec.c, since
222 * the format name from the rtpmap line never is passed into rtpdec. */
223 if (!rtsp_st
->dynamic_handler
)
224 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
227 c
= avcodec_find_decoder(codec
->codec_id
);
233 get_word_sep(buf
, sizeof(buf
), "/", &p
);
235 switch (codec
->codec_type
) {
236 case AVMEDIA_TYPE_AUDIO
:
237 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
238 codec
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
239 codec
->channels
= RTSP_DEFAULT_NB_AUDIO_CHANNELS
;
241 codec
->sample_rate
= i
;
242 avpriv_set_pts_info(st
, 32, 1, codec
->sample_rate
);
243 get_word_sep(buf
, sizeof(buf
), "/", &p
);
248 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
250 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
253 case AVMEDIA_TYPE_VIDEO
:
254 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
256 avpriv_set_pts_info(st
, 32, 1, i
);
261 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_handler
->init
)
262 rtsp_st
->dynamic_handler
->init(s
, st
->index
,
263 rtsp_st
->dynamic_protocol_context
);
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268 * is broken out as a function because it is used in rtp_h264.c, which is
270 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
271 char *value
, int value_size
)
273 *p
+= strspn(*p
, SPACE_CHARS
);
275 get_word_sep(attr
, attr_size
, "=", p
);
278 get_word_sep(value
, value_size
, ";", p
);
286 typedef struct SDPParseState
{
288 struct sockaddr_storage default_ip
;
290 int skip_media
; ///< set if an unknown m= line occurs
293 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
294 int letter
, const char *buf
)
296 RTSPState
*rt
= s
->priv_data
;
297 char buf1
[64], st_type
[64];
299 enum AVMediaType codec_type
;
303 struct sockaddr_storage sdp_ip
;
306 av_dlog(s
, "sdp: %c='%s'\n", letter
, buf
);
309 if (s1
->skip_media
&& letter
!= 'm')
313 get_word(buf1
, sizeof(buf1
), &p
);
314 if (strcmp(buf1
, "IN") != 0)
316 get_word(buf1
, sizeof(buf1
), &p
);
317 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6"))
319 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
320 if (get_sockaddr(buf1
, &sdp_ip
))
325 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
328 if (s
->nb_streams
== 0) {
329 s1
->default_ip
= sdp_ip
;
330 s1
->default_ttl
= ttl
;
332 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
333 rtsp_st
->sdp_ip
= sdp_ip
;
334 rtsp_st
->sdp_ttl
= ttl
;
338 av_dict_set(&s
->metadata
, "title", p
, 0);
341 if (s
->nb_streams
== 0) {
342 av_dict_set(&s
->metadata
, "comment", p
, 0);
349 codec_type
= AVMEDIA_TYPE_UNKNOWN
;
350 get_word(st_type
, sizeof(st_type
), &p
);
351 if (!strcmp(st_type
, "audio")) {
352 codec_type
= AVMEDIA_TYPE_AUDIO
;
353 } else if (!strcmp(st_type
, "video")) {
354 codec_type
= AVMEDIA_TYPE_VIDEO
;
355 } else if (!strcmp(st_type
, "application")) {
356 codec_type
= AVMEDIA_TYPE_DATA
;
358 if (codec_type
== AVMEDIA_TYPE_UNKNOWN
|| !(rt
->media_type_mask
& (1 << codec_type
))) {
362 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
365 rtsp_st
->stream_index
= -1;
366 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
368 rtsp_st
->sdp_ip
= s1
->default_ip
;
369 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
371 get_word(buf1
, sizeof(buf1
), &p
); /* port */
372 rtsp_st
->sdp_port
= atoi(buf1
);
374 get_word(buf1
, sizeof(buf1
), &p
); /* protocol */
375 if (!strcmp(buf1
, "udp"))
376 rt
->transport
= RTSP_TRANSPORT_RAW
;
377 else if (strstr(buf1
, "/AVPF") || strstr(buf1
, "/SAVPF"))
378 rtsp_st
->feedback
= 1;
380 /* XXX: handle list of formats */
381 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
382 rtsp_st
->sdp_payload_type
= atoi(buf1
);
384 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
385 /* no corresponding stream */
386 if (rt
->transport
== RTSP_TRANSPORT_RAW
) {
387 if (!rt
->ts
&& CONFIG_RTPDEC
)
388 rt
->ts
= ff_mpegts_parse_open(s
);
390 RTPDynamicProtocolHandler
*handler
;
391 handler
= ff_rtp_handler_find_by_id(
392 rtsp_st
->sdp_payload_type
, AVMEDIA_TYPE_DATA
);
393 init_rtp_handler(handler
, rtsp_st
, NULL
);
394 if (handler
&& handler
->init
)
395 handler
->init(s
, -1, rtsp_st
->dynamic_protocol_context
);
397 } else if (rt
->server_type
== RTSP_SERVER_WMS
&&
398 codec_type
== AVMEDIA_TYPE_DATA
) {
399 /* RTX stream, a stream that carries all the other actual
400 * audio/video streams. Don't expose this to the callers. */
402 st
= avformat_new_stream(s
, NULL
);
405 st
->id
= rt
->nb_rtsp_streams
- 1;
406 rtsp_st
->stream_index
= st
->index
;
407 st
->codec
->codec_type
= codec_type
;
408 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
409 RTPDynamicProtocolHandler
*handler
;
410 /* if standard payload type, we can find the codec right now */
411 ff_rtp_get_codec_info(st
->codec
, rtsp_st
->sdp_payload_type
);
412 if (st
->codec
->codec_type
== AVMEDIA_TYPE_AUDIO
&&
413 st
->codec
->sample_rate
> 0)
414 avpriv_set_pts_info(st
, 32, 1, st
->codec
->sample_rate
);
415 /* Even static payload types may need a custom depacketizer */
416 handler
= ff_rtp_handler_find_by_id(
417 rtsp_st
->sdp_payload_type
, st
->codec
->codec_type
);
418 init_rtp_handler(handler
, rtsp_st
, st
->codec
);
419 if (handler
&& handler
->init
)
420 handler
->init(s
, st
->index
,
421 rtsp_st
->dynamic_protocol_context
);
424 /* put a default control url */
425 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
426 sizeof(rtsp_st
->control_url
));
429 if (av_strstart(p
, "control:", &p
)) {
430 if (s
->nb_streams
== 0) {
431 if (!strncmp(p
, "rtsp://", 7))
432 av_strlcpy(rt
->control_uri
, p
,
433 sizeof(rt
->control_uri
));
436 /* get the control url */
437 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
439 /* XXX: may need to add full url resolution */
440 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
442 if (proto
[0] == '\0') {
443 /* relative control URL */
444 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
445 av_strlcat(rtsp_st
->control_url
, "/",
446 sizeof(rtsp_st
->control_url
));
447 av_strlcat(rtsp_st
->control_url
, p
,
448 sizeof(rtsp_st
->control_url
));
450 av_strlcpy(rtsp_st
->control_url
, p
,
451 sizeof(rtsp_st
->control_url
));
453 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
454 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
455 get_word(buf1
, sizeof(buf1
), &p
);
456 payload_type
= atoi(buf1
);
457 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
458 if (rtsp_st
->stream_index
>= 0) {
459 st
= s
->streams
[rtsp_st
->stream_index
];
460 sdp_parse_rtpmap(s
, st
, rtsp_st
, payload_type
, p
);
462 } else if (av_strstart(p
, "fmtp:", &p
) ||
463 av_strstart(p
, "framesize:", &p
)) {
464 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
465 // let dynamic protocol handlers have a stab at the line.
466 get_word(buf1
, sizeof(buf1
), &p
);
467 payload_type
= atoi(buf1
);
468 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
469 rtsp_st
= rt
->rtsp_streams
[i
];
470 if (rtsp_st
->sdp_payload_type
== payload_type
&&
471 rtsp_st
->dynamic_handler
&&
472 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
473 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, i
,
474 rtsp_st
->dynamic_protocol_context
, buf
);
476 } else if (av_strstart(p
, "range:", &p
)) {
479 // this is so that seeking on a streamed file can work.
480 rtsp_parse_range_npt(p
, &start
, &end
);
481 s
->start_time
= start
;
482 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
483 s
->duration
= (end
== AV_NOPTS_VALUE
) ?
484 AV_NOPTS_VALUE
: end
- start
;
485 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
487 rt
->transport
= RTSP_TRANSPORT_RDT
;
488 } else if (av_strstart(p
, "SampleRate:integer;", &p
) &&
490 st
= s
->streams
[s
->nb_streams
- 1];
491 st
->codec
->sample_rate
= atoi(p
);
492 } else if (av_strstart(p
, "crypto:", &p
) && s
->nb_streams
> 0) {
494 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
495 get_word(buf1
, sizeof(buf1
), &p
); // ignore tag
496 get_word(rtsp_st
->crypto_suite
, sizeof(rtsp_st
->crypto_suite
), &p
);
497 p
+= strspn(p
, SPACE_CHARS
);
498 if (av_strstart(p
, "inline:", &p
))
499 get_word(rtsp_st
->crypto_params
, sizeof(rtsp_st
->crypto_params
), &p
);
501 if (rt
->server_type
== RTSP_SERVER_WMS
)
502 ff_wms_parse_sdp_a_line(s
, p
);
503 if (s
->nb_streams
> 0) {
504 rtsp_st
= rt
->rtsp_streams
[rt
->nb_rtsp_streams
- 1];
506 if (rt
->server_type
== RTSP_SERVER_REAL
)
507 ff_real_parse_sdp_a_line(s
, rtsp_st
->stream_index
, p
);
509 if (rtsp_st
->dynamic_handler
&&
510 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
511 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
512 rtsp_st
->stream_index
,
513 rtsp_st
->dynamic_protocol_context
, buf
);
520 int ff_sdp_parse(AVFormatContext
*s
, const char *content
)
522 RTSPState
*rt
= s
->priv_data
;
525 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
526 * contain long SDP lines containing complete ASF Headers (several
527 * kB) or arrays of MDPR (RM stream descriptor) headers plus
528 * "rulebooks" describing their properties. Therefore, the SDP line
531 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
532 * in rtpdec_xiph.c. */
534 SDPParseState sdp_parse_state
= { { 0 } }, *s1
= &sdp_parse_state
;
538 p
+= strspn(p
, SPACE_CHARS
);
546 /* get the content */
548 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
549 if ((q
- buf
) < sizeof(buf
) - 1)
554 sdp_parse_line(s
, s1
, letter
, buf
);
556 while (*p
!= '\n' && *p
!= '\0')
561 rt
->p
= av_malloc(sizeof(struct pollfd
)*2*(rt
->nb_rtsp_streams
+1));
562 if (!rt
->p
) return AVERROR(ENOMEM
);
565 #endif /* CONFIG_RTPDEC */
567 void ff_rtsp_undo_setup(AVFormatContext
*s
)
569 RTSPState
*rt
= s
->priv_data
;
572 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
573 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
576 if (rtsp_st
->transport_priv
) {
578 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
579 av_write_trailer(rtpctx
);
580 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
582 avio_close_dyn_buf(rtpctx
->pb
, &ptr
);
585 avio_close(rtpctx
->pb
);
587 avformat_free_context(rtpctx
);
588 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
589 ff_rdt_parse_close(rtsp_st
->transport_priv
);
590 else if (rt
->transport
== RTSP_TRANSPORT_RTP
&& CONFIG_RTPDEC
)
591 ff_rtp_parse_close(rtsp_st
->transport_priv
);
593 rtsp_st
->transport_priv
= NULL
;
594 if (rtsp_st
->rtp_handle
)
595 ffurl_close(rtsp_st
->rtp_handle
);
596 rtsp_st
->rtp_handle
= NULL
;
600 /* close and free RTSP streams */
601 void ff_rtsp_close_streams(AVFormatContext
*s
)
603 RTSPState
*rt
= s
->priv_data
;
607 ff_rtsp_undo_setup(s
);
608 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
609 rtsp_st
= rt
->rtsp_streams
[i
];
611 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
)
612 rtsp_st
->dynamic_handler
->free(
613 rtsp_st
->dynamic_protocol_context
);
617 av_free(rt
->rtsp_streams
);
619 avformat_close_input(&rt
->asf_ctx
);
621 if (rt
->ts
&& CONFIG_RTPDEC
)
622 ff_mpegts_parse_close(rt
->ts
);
624 av_free(rt
->recvbuf
);
627 int ff_rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
629 RTSPState
*rt
= s
->priv_data
;
631 int reordering_queue_size
= rt
->reordering_queue_size
;
632 if (reordering_queue_size
< 0) {
633 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
|| !s
->max_delay
)
634 reordering_queue_size
= 0;
636 reordering_queue_size
= RTP_REORDER_QUEUE_DEFAULT_SIZE
;
639 /* open the RTP context */
640 if (rtsp_st
->stream_index
>= 0)
641 st
= s
->streams
[rtsp_st
->stream_index
];
643 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
645 if (s
->oformat
&& CONFIG_RTSP_MUXER
) {
646 int ret
= ff_rtp_chain_mux_open(&rtsp_st
->transport_priv
, s
, st
,
648 RTSP_TCP_MAX_PACKET_SIZE
,
649 rtsp_st
->stream_index
);
650 /* Ownership of rtp_handle is passed to the rtp mux context */
651 rtsp_st
->rtp_handle
= NULL
;
654 } else if (rt
->transport
== RTSP_TRANSPORT_RAW
) {
655 return 0; // Don't need to open any parser here
656 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
657 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
658 rtsp_st
->dynamic_protocol_context
,
659 rtsp_st
->dynamic_handler
);
660 else if (CONFIG_RTPDEC
)
661 rtsp_st
->transport_priv
= ff_rtp_parse_open(s
, st
,
662 rtsp_st
->sdp_payload_type
,
663 reordering_queue_size
);
665 if (!rtsp_st
->transport_priv
) {
666 return AVERROR(ENOMEM
);
667 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
&& CONFIG_RTPDEC
) {
668 if (rtsp_st
->dynamic_handler
) {
669 ff_rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
670 rtsp_st
->dynamic_protocol_context
,
671 rtsp_st
->dynamic_handler
);
673 if (rtsp_st
->crypto_suite
[0])
674 ff_rtp_parse_set_crypto(rtsp_st
->transport_priv
,
675 rtsp_st
->crypto_suite
,
676 rtsp_st
->crypto_params
);
682 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
683 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
690 q
+= strspn(q
, SPACE_CHARS
);
691 v
= strtol(q
, &p
, 10);
695 v
= strtol(p
, &p
, 10);
704 /* XXX: only one transport specification is parsed */
705 static void rtsp_parse_transport(RTSPMessageHeader
*reply
, const char *p
)
707 char transport_protocol
[16];
709 char lower_transport
[16];
711 RTSPTransportField
*th
;
714 reply
->nb_transports
= 0;
717 p
+= strspn(p
, SPACE_CHARS
);
721 th
= &reply
->transports
[reply
->nb_transports
];
723 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
725 if (!av_strcasecmp (transport_protocol
, "rtp")) {
726 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
727 lower_transport
[0] = '\0';
728 /* rtp/avp/<protocol> */
730 get_word_sep(lower_transport
, sizeof(lower_transport
),
733 th
->transport
= RTSP_TRANSPORT_RTP
;
734 } else if (!av_strcasecmp (transport_protocol
, "x-pn-tng") ||
735 !av_strcasecmp (transport_protocol
, "x-real-rdt")) {
736 /* x-pn-tng/<protocol> */
737 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
739 th
->transport
= RTSP_TRANSPORT_RDT
;
740 } else if (!av_strcasecmp(transport_protocol
, "raw")) {
741 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
742 lower_transport
[0] = '\0';
743 /* raw/raw/<protocol> */
745 get_word_sep(lower_transport
, sizeof(lower_transport
),
748 th
->transport
= RTSP_TRANSPORT_RAW
;
750 if (!av_strcasecmp(lower_transport
, "TCP"))
751 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
753 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
757 /* get each parameter */
758 while (*p
!= '\0' && *p
!= ',') {
759 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
760 if (!strcmp(parameter
, "port")) {
763 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
765 } else if (!strcmp(parameter
, "client_port")) {
768 rtsp_parse_range(&th
->client_port_min
,
769 &th
->client_port_max
, &p
);
771 } else if (!strcmp(parameter
, "server_port")) {
774 rtsp_parse_range(&th
->server_port_min
,
775 &th
->server_port_max
, &p
);
777 } else if (!strcmp(parameter
, "interleaved")) {
780 rtsp_parse_range(&th
->interleaved_min
,
781 &th
->interleaved_max
, &p
);
783 } else if (!strcmp(parameter
, "multicast")) {
784 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
785 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
786 } else if (!strcmp(parameter
, "ttl")) {
790 th
->ttl
= strtol(p
, &end
, 10);
793 } else if (!strcmp(parameter
, "destination")) {
796 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
797 get_sockaddr(buf
, &th
->destination
);
799 } else if (!strcmp(parameter
, "source")) {
802 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
803 av_strlcpy(th
->source
, buf
, sizeof(th
->source
));
805 } else if (!strcmp(parameter
, "mode")) {
808 get_word_sep(buf
, sizeof(buf
), ";, ", &p
);
809 if (!strcmp(buf
, "record") ||
810 !strcmp(buf
, "receive"))
815 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
823 reply
->nb_transports
++;
827 static void handle_rtp_info(RTSPState
*rt
, const char *url
,
828 uint32_t seq
, uint32_t rtptime
)
831 if (!rtptime
|| !url
[0])
833 if (rt
->transport
!= RTSP_TRANSPORT_RTP
)
835 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
836 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
837 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
840 if (!strcmp(rtsp_st
->control_url
, url
)) {
841 rtpctx
->base_timestamp
= rtptime
;
847 static void rtsp_parse_rtp_info(RTSPState
*rt
, const char *p
)
850 char key
[20], value
[1024], url
[1024] = "";
851 uint32_t seq
= 0, rtptime
= 0;
854 p
+= strspn(p
, SPACE_CHARS
);
857 get_word_sep(key
, sizeof(key
), "=", &p
);
861 get_word_sep(value
, sizeof(value
), ";, ", &p
);
863 if (!strcmp(key
, "url"))
864 av_strlcpy(url
, value
, sizeof(url
));
865 else if (!strcmp(key
, "seq"))
866 seq
= strtoul(value
, NULL
, 10);
867 else if (!strcmp(key
, "rtptime"))
868 rtptime
= strtoul(value
, NULL
, 10);
870 handle_rtp_info(rt
, url
, seq
, rtptime
);
879 handle_rtp_info(rt
, url
, seq
, rtptime
);
882 void ff_rtsp_parse_line(RTSPMessageHeader
*reply
, const char *buf
,
883 RTSPState
*rt
, const char *method
)
887 /* NOTE: we do case independent match for broken servers */
889 if (av_stristart(p
, "Session:", &p
)) {
891 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
892 if (av_stristart(p
, ";timeout=", &p
) &&
893 (t
= strtol(p
, NULL
, 10)) > 0) {
896 } else if (av_stristart(p
, "Content-Length:", &p
)) {
897 reply
->content_length
= strtol(p
, NULL
, 10);
898 } else if (av_stristart(p
, "Transport:", &p
)) {
899 rtsp_parse_transport(reply
, p
);
900 } else if (av_stristart(p
, "CSeq:", &p
)) {
901 reply
->seq
= strtol(p
, NULL
, 10);
902 } else if (av_stristart(p
, "Range:", &p
)) {
903 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
904 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
905 p
+= strspn(p
, SPACE_CHARS
);
906 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
907 } else if (av_stristart(p
, "Server:", &p
)) {
908 p
+= strspn(p
, SPACE_CHARS
);
909 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
910 } else if (av_stristart(p
, "Notice:", &p
) ||
911 av_stristart(p
, "X-Notice:", &p
)) {
912 reply
->notice
= strtol(p
, NULL
, 10);
913 } else if (av_stristart(p
, "Location:", &p
)) {
914 p
+= strspn(p
, SPACE_CHARS
);
915 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
916 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && rt
) {
917 p
+= strspn(p
, SPACE_CHARS
);
918 ff_http_auth_handle_header(&rt
->auth_state
, "WWW-Authenticate", p
);
919 } else if (av_stristart(p
, "Authentication-Info:", &p
) && rt
) {
920 p
+= strspn(p
, SPACE_CHARS
);
921 ff_http_auth_handle_header(&rt
->auth_state
, "Authentication-Info", p
);
922 } else if (av_stristart(p
, "Content-Base:", &p
) && rt
) {
923 p
+= strspn(p
, SPACE_CHARS
);
924 if (method
&& !strcmp(method
, "DESCRIBE"))
925 av_strlcpy(rt
->control_uri
, p
, sizeof(rt
->control_uri
));
926 } else if (av_stristart(p
, "RTP-Info:", &p
) && rt
) {
927 p
+= strspn(p
, SPACE_CHARS
);
928 if (method
&& !strcmp(method
, "PLAY"))
929 rtsp_parse_rtp_info(rt
, p
);
930 } else if (av_stristart(p
, "Public:", &p
) && rt
) {
931 if (strstr(p
, "GET_PARAMETER") &&
932 method
&& !strcmp(method
, "OPTIONS"))
933 rt
->get_parameter_supported
= 1;
934 } else if (av_stristart(p
, "x-Accept-Dynamic-Rate:", &p
) && rt
) {
935 p
+= strspn(p
, SPACE_CHARS
);
936 rt
->accept_dynamic_rate
= atoi(p
);
937 } else if (av_stristart(p
, "Content-Type:", &p
)) {
938 p
+= strspn(p
, SPACE_CHARS
);
939 av_strlcpy(reply
->content_type
, p
, sizeof(reply
->content_type
));
943 /* skip a RTP/TCP interleaved packet */
944 void ff_rtsp_skip_packet(AVFormatContext
*s
)
946 RTSPState
*rt
= s
->priv_data
;
950 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, 3);
953 len
= AV_RB16(buf
+ 1);
955 av_dlog(s
, "skipping RTP packet len=%d\n", len
);
960 if (len1
> sizeof(buf
))
962 ret
= ffurl_read_complete(rt
->rtsp_hd
, buf
, len1
);
969 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
970 unsigned char **content_ptr
,
971 int return_on_interleaved_data
, const char *method
)
973 RTSPState
*rt
= s
->priv_data
;
974 char buf
[4096], buf1
[1024], *q
;
977 int ret
, content_length
, line_count
= 0, request
= 0;
978 unsigned char *content
= NULL
;
984 memset(reply
, 0, sizeof(*reply
));
986 /* parse reply (XXX: use buffers) */
987 rt
->last_reply
[0] = '\0';
991 ret
= ffurl_read_complete(rt
->rtsp_hd
, &ch
, 1);
992 av_dlog(s
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
998 /* XXX: only parse it if first char on line ? */
999 if (return_on_interleaved_data
) {
1002 ff_rtsp_skip_packet(s
);
1003 } else if (ch
!= '\r') {
1004 if ((q
- buf
) < sizeof(buf
) - 1)
1010 av_dlog(s
, "line='%s'\n", buf
);
1012 /* test if last line */
1016 if (line_count
== 0) {
1017 /* get reply code */
1018 get_word(buf1
, sizeof(buf1
), &p
);
1019 if (!strncmp(buf1
, "RTSP/", 5)) {
1020 get_word(buf1
, sizeof(buf1
), &p
);
1021 reply
->status_code
= atoi(buf1
);
1022 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
1024 av_strlcpy(reply
->reason
, buf1
, sizeof(reply
->reason
)); // method
1025 get_word(buf1
, sizeof(buf1
), &p
); // object
1029 ff_rtsp_parse_line(reply
, p
, rt
, method
);
1030 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
1031 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
1036 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0' && !request
)
1037 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
1039 content_length
= reply
->content_length
;
1040 if (content_length
> 0) {
1041 /* leave some room for a trailing '\0' (useful for simple parsing) */
1042 content
= av_malloc(content_length
+ 1);
1043 ffurl_read_complete(rt
->rtsp_hd
, content
, content_length
);
1044 content
[content_length
] = '\0';
1047 *content_ptr
= content
;
1053 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1054 const char* ptr
= buf
;
1056 if (!strcmp(reply
->reason
, "OPTIONS")) {
1057 snprintf(buf
, sizeof(buf
), "RTSP/1.0 200 OK\r\n");
1059 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", reply
->seq
);
1060 if (reply
->session_id
[0])
1061 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n",
1064 snprintf(buf
, sizeof(buf
), "RTSP/1.0 501 Not Implemented\r\n");
1066 av_strlcat(buf
, "\r\n", sizeof(buf
));
1068 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1069 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1072 ffurl_write(rt
->rtsp_hd_out
, ptr
, strlen(ptr
));
1074 rt
->last_cmd_time
= av_gettime();
1075 /* Even if the request from the server had data, it is not the data
1076 * that the caller wants or expects. The memory could also be leaked
1077 * if the actual following reply has content data. */
1079 av_freep(content_ptr
);
1080 /* If method is set, this is called from ff_rtsp_send_cmd,
1081 * where a reply to exactly this request is awaited. For
1082 * callers from within packet receiving, we just want to
1083 * return to the caller and go back to receiving packets. */
1089 if (rt
->seq
!= reply
->seq
) {
1090 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
1091 rt
->seq
, reply
->seq
);
1095 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
1096 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
1097 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
1098 rt
->state
= RTSP_STATE_IDLE
;
1099 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
1100 return AVERROR(EIO
); /* data or server error */
1101 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
1102 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
1103 return AVERROR(EPERM
);
1109 * Send a command to the RTSP server without waiting for the reply.
1111 * @param s RTSP (de)muxer context
1112 * @param method the method for the request
1113 * @param url the target url for the request
1114 * @param headers extra header lines to include in the request
1115 * @param send_content if non-null, the data to send as request body content
1116 * @param send_content_length the length of the send_content data, or 0 if
1117 * send_content is null
1119 * @return zero if success, nonzero otherwise
1121 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
1122 const char *method
, const char *url
,
1123 const char *headers
,
1124 const unsigned char *send_content
,
1125 int send_content_length
)
1127 RTSPState
*rt
= s
->priv_data
;
1128 char buf
[4096], *out_buf
;
1129 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
1131 /* Add in RTSP headers */
1134 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
1136 av_strlcat(buf
, headers
, sizeof(buf
));
1137 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
1138 if (rt
->session_id
[0] != '\0' && (!headers
||
1139 !strstr(headers
, "\nIf-Match:"))) {
1140 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
1143 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
1144 rt
->auth
, url
, method
);
1146 av_strlcat(buf
, str
, sizeof(buf
));
1149 if (send_content_length
> 0 && send_content
)
1150 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
1151 av_strlcat(buf
, "\r\n", sizeof(buf
));
1153 /* base64 encode rtsp if tunneling */
1154 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1155 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
1156 out_buf
= base64buf
;
1159 av_dlog(s
, "Sending:\n%s--\n", buf
);
1161 ffurl_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
1162 if (send_content_length
> 0 && send_content
) {
1163 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1164 av_log(s
, AV_LOG_ERROR
, "tunneling of RTSP requests "
1165 "with content data not supported\n");
1166 return AVERROR_PATCHWELCOME
;
1168 ffurl_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
1170 rt
->last_cmd_time
= av_gettime();
1175 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
1176 const char *url
, const char *headers
)
1178 return ff_rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
1181 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
1182 const char *headers
, RTSPMessageHeader
*reply
,
1183 unsigned char **content_ptr
)
1185 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
1186 content_ptr
, NULL
, 0);
1189 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
1190 const char *method
, const char *url
,
1192 RTSPMessageHeader
*reply
,
1193 unsigned char **content_ptr
,
1194 const unsigned char *send_content
,
1195 int send_content_length
)
1197 RTSPState
*rt
= s
->priv_data
;
1198 HTTPAuthType cur_auth_type
;
1199 int ret
, attempts
= 0;
1202 cur_auth_type
= rt
->auth_state
.auth_type
;
1203 if ((ret
= ff_rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
1205 send_content_length
)))
1208 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0, method
) ) < 0)
1212 if (reply
->status_code
== 401 &&
1213 (cur_auth_type
== HTTP_AUTH_NONE
|| rt
->auth_state
.stale
) &&
1214 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
&& attempts
< 2)
1217 if (reply
->status_code
> 400){
1218 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
1222 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
1228 int ff_rtsp_make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
1229 int lower_transport
, const char *real_challenge
)
1231 RTSPState
*rt
= s
->priv_data
;
1232 int rtx
= 0, j
, i
, err
, interleave
= 0, port_off
;
1233 RTSPStream
*rtsp_st
;
1234 RTSPMessageHeader reply1
, *reply
= &reply1
;
1236 const char *trans_pref
;
1238 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
1239 trans_pref
= "x-pn-tng";
1240 else if (rt
->transport
== RTSP_TRANSPORT_RAW
)
1241 trans_pref
= "RAW/RAW";
1243 trans_pref
= "RTP/AVP";
1245 /* default timeout: 1 minute */
1248 /* for each stream, make the setup request */
1249 /* XXX: we assume the same server is used for the control of each
1252 /* Choose a random starting offset within the first half of the
1253 * port range, to allow for a number of ports to try even if the offset
1254 * happens to be at the end of the random range. */
1255 port_off
= av_get_random_seed() % ((rt
->rtp_port_max
- rt
->rtp_port_min
)/2);
1256 /* even random offset */
1257 port_off
-= port_off
& 0x01;
1259 for (j
= rt
->rtp_port_min
+ port_off
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1260 char transport
[2048];
1263 * WMS serves all UDP data over a single connection, the RTX, which
1264 * isn't necessarily the first in the SDP but has to be the first
1265 * to be set up, else the second/third SETUP will fail with a 461.
1267 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1268 rt
->server_type
== RTSP_SERVER_WMS
) {
1271 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1272 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1274 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1278 if (rtx
== rt
->nb_rtsp_streams
)
1279 return -1; /* no RTX found */
1280 rtsp_st
= rt
->rtsp_streams
[rtx
];
1282 rtsp_st
= rt
->rtsp_streams
[i
> rtx
? i
: i
- 1];
1284 rtsp_st
= rt
->rtsp_streams
[i
];
1287 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1290 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1291 port
= reply
->transports
[0].client_port_min
;
1295 /* first try in specified port range */
1296 while (j
<= rt
->rtp_port_max
) {
1297 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1298 "?localport=%d", j
);
1299 /* we will use two ports per rtp stream (rtp and rtcp) */
1301 if (!ffurl_open(&rtsp_st
->rtp_handle
, buf
, AVIO_FLAG_READ_WRITE
,
1302 &s
->interrupt_callback
, NULL
))
1306 av_log(s
, AV_LOG_ERROR
, "Unable to open an input RTP port\n");
1311 port
= ff_rtp_get_local_rtp_port(rtsp_st
->rtp_handle
);
1313 snprintf(transport
, sizeof(transport
) - 1,
1314 "%s/UDP;", trans_pref
);
1315 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1316 av_strlcat(transport
, "unicast;", sizeof(transport
));
1317 av_strlcatf(transport
, sizeof(transport
),
1318 "client_port=%d", port
);
1319 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1320 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1321 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1325 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1326 /* For WMS streams, the application streams are only used for
1327 * UDP. When trying to set it up for TCP streams, the server
1328 * will return an error. Therefore, we skip those streams. */
1329 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1330 (rtsp_st
->stream_index
< 0 ||
1331 s
->streams
[rtsp_st
->stream_index
]->codec
->codec_type
==
1334 snprintf(transport
, sizeof(transport
) - 1,
1335 "%s/TCP;", trans_pref
);
1336 if (rt
->transport
!= RTSP_TRANSPORT_RDT
)
1337 av_strlcat(transport
, "unicast;", sizeof(transport
));
1338 av_strlcatf(transport
, sizeof(transport
),
1339 "interleaved=%d-%d",
1340 interleave
, interleave
+ 1);
1344 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1345 snprintf(transport
, sizeof(transport
) - 1,
1346 "%s/UDP;multicast", trans_pref
);
1349 av_strlcat(transport
, ";mode=record", sizeof(transport
));
1350 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1351 rt
->server_type
== RTSP_SERVER_WMS
)
1352 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1353 snprintf(cmd
, sizeof(cmd
),
1354 "Transport: %s\r\n",
1356 if (rt
->accept_dynamic_rate
)
1357 av_strlcat(cmd
, "x-Dynamic-Rate: 0\r\n", sizeof(cmd
));
1358 if (i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
&& CONFIG_RTPDEC
) {
1359 char real_res
[41], real_csum
[9];
1360 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1362 av_strlcatf(cmd
, sizeof(cmd
),
1364 "RealChallenge2: %s, sd=%s\r\n",
1365 rt
->session_id
, real_res
, real_csum
);
1367 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1368 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1371 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1372 reply
->nb_transports
!= 1) {
1373 err
= AVERROR_INVALIDDATA
;
1377 /* XXX: same protocol for all streams is required */
1379 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1380 reply
->transports
[0].transport
!= rt
->transport
) {
1381 err
= AVERROR_INVALIDDATA
;
1385 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1386 rt
->transport
= reply
->transports
[0].transport
;
1389 /* Fail if the server responded with another lower transport mode
1390 * than what we requested. */
1391 if (reply
->transports
[0].lower_transport
!= lower_transport
) {
1392 av_log(s
, AV_LOG_ERROR
, "Nonmatching transport in server reply\n");
1393 err
= AVERROR_INVALIDDATA
;
1397 switch(reply
->transports
[0].lower_transport
) {
1398 case RTSP_LOWER_TRANSPORT_TCP
:
1399 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1400 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1403 case RTSP_LOWER_TRANSPORT_UDP
: {
1404 char url
[1024], options
[30] = "";
1406 if (rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
)
1407 av_strlcpy(options
, "?connect=1", sizeof(options
));
1408 /* Use source address if specified */
1409 if (reply
->transports
[0].source
[0]) {
1410 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1411 reply
->transports
[0].source
,
1412 reply
->transports
[0].server_port_min
, "%s", options
);
1414 ff_url_join(url
, sizeof(url
), "rtp", NULL
, host
,
1415 reply
->transports
[0].server_port_min
, "%s", options
);
1417 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1418 ff_rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1419 err
= AVERROR_INVALIDDATA
;
1422 /* Try to initialize the connection state in a
1423 * potential NAT router by sending dummy packets.
1424 * RTP/RTCP dummy packets are used for RDT, too.
1426 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) && s
->iformat
&&
1428 ff_rtp_send_punch_packets(rtsp_st
->rtp_handle
);
1431 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1432 char url
[1024], namebuf
[50], optbuf
[20] = "";
1433 struct sockaddr_storage addr
;
1436 if (reply
->transports
[0].destination
.ss_family
) {
1437 addr
= reply
->transports
[0].destination
;
1438 port
= reply
->transports
[0].port_min
;
1439 ttl
= reply
->transports
[0].ttl
;
1441 addr
= rtsp_st
->sdp_ip
;
1442 port
= rtsp_st
->sdp_port
;
1443 ttl
= rtsp_st
->sdp_ttl
;
1446 snprintf(optbuf
, sizeof(optbuf
), "?ttl=%d", ttl
);
1447 getnameinfo((struct sockaddr
*) &addr
, sizeof(addr
),
1448 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1449 ff_url_join(url
, sizeof(url
), "rtp", NULL
, namebuf
,
1450 port
, "%s", optbuf
);
1451 if (ffurl_open(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
,
1452 &s
->interrupt_callback
, NULL
) < 0) {
1453 err
= AVERROR_INVALIDDATA
;
1460 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
1464 if (rt
->nb_rtsp_streams
&& reply
->timeout
> 0)
1465 rt
->timeout
= reply
->timeout
;
1467 if (rt
->server_type
== RTSP_SERVER_REAL
)
1468 rt
->need_subscription
= 1;
1473 ff_rtsp_undo_setup(s
);
1477 void ff_rtsp_close_connections(AVFormatContext
*s
)
1479 RTSPState
*rt
= s
->priv_data
;
1480 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
) ffurl_close(rt
->rtsp_hd_out
);
1481 ffurl_close(rt
->rtsp_hd
);
1482 rt
->rtsp_hd
= rt
->rtsp_hd_out
= NULL
;
1485 int ff_rtsp_connect(AVFormatContext
*s
)
1487 RTSPState
*rt
= s
->priv_data
;
1488 char host
[1024], path
[1024], tcpname
[1024], cmd
[2048], auth
[128];
1489 int port
, err
, tcp_fd
;
1490 RTSPMessageHeader reply1
= {0}, *reply
= &reply1
;
1491 int lower_transport_mask
= 0;
1492 char real_challenge
[64] = "";
1493 struct sockaddr_storage peer
;
1494 socklen_t peer_len
= sizeof(peer
);
1496 if (rt
->rtp_port_max
< rt
->rtp_port_min
) {
1497 av_log(s
, AV_LOG_ERROR
, "Invalid UDP port range, max port %d less "
1498 "than min port %d\n", rt
->rtp_port_max
,
1500 return AVERROR(EINVAL
);
1503 if (!ff_network_init())
1504 return AVERROR(EIO
);
1506 if (s
->max_delay
< 0) /* Not set by the caller */
1507 s
->max_delay
= s
->iformat
? DEFAULT_REORDERING_DELAY
: 0;
1509 rt
->control_transport
= RTSP_MODE_PLAIN
;
1510 if (rt
->lower_transport_mask
& (1 << RTSP_LOWER_TRANSPORT_HTTP
)) {
1511 rt
->lower_transport_mask
= 1 << RTSP_LOWER_TRANSPORT_TCP
;
1512 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1514 /* Only pass through valid flags from here */
1515 rt
->lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1518 lower_transport_mask
= rt
->lower_transport_mask
;
1519 /* extract hostname and port */
1520 av_url_split(NULL
, 0, auth
, sizeof(auth
),
1521 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->filename
);
1523 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1526 port
= RTSP_DEFAULT_PORT
;
1528 if (!lower_transport_mask
)
1529 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1532 /* Only UDP or TCP - UDP multicast isn't supported. */
1533 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1534 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1535 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1536 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1537 "only UDP and TCP are supported for output.\n");
1538 err
= AVERROR(EINVAL
);
1543 /* Construct the URI used in request; this is similar to s->filename,
1544 * but with authentication credentials removed and RTSP specific options
1546 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), "rtsp", NULL
,
1547 host
, port
, "%s", path
);
1549 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1550 /* set up initial handshake for tunneling */
1551 char httpname
[1024];
1552 char sessioncookie
[17];
1555 ff_url_join(httpname
, sizeof(httpname
), "http", auth
, host
, port
, "%s", path
);
1556 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1557 av_get_random_seed(), av_get_random_seed());
1560 if (ffurl_alloc(&rt
->rtsp_hd
, httpname
, AVIO_FLAG_READ
,
1561 &s
->interrupt_callback
) < 0) {
1566 /* generate GET headers */
1567 snprintf(headers
, sizeof(headers
),
1568 "x-sessioncookie: %s\r\n"
1569 "Accept: application/x-rtsp-tunnelled\r\n"
1570 "Pragma: no-cache\r\n"
1571 "Cache-Control: no-cache\r\n",
1573 av_opt_set(rt
->rtsp_hd
->priv_data
, "headers", headers
, 0);
1575 /* complete the connection */
1576 if (ffurl_connect(rt
->rtsp_hd
, NULL
)) {
1582 if (ffurl_alloc(&rt
->rtsp_hd_out
, httpname
, AVIO_FLAG_WRITE
,
1583 &s
->interrupt_callback
) < 0 ) {
1588 /* generate POST headers */
1589 snprintf(headers
, sizeof(headers
),
1590 "x-sessioncookie: %s\r\n"
1591 "Content-Type: application/x-rtsp-tunnelled\r\n"
1592 "Pragma: no-cache\r\n"
1593 "Cache-Control: no-cache\r\n"
1594 "Content-Length: 32767\r\n"
1595 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1597 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "headers", headers
, 0);
1598 av_opt_set(rt
->rtsp_hd_out
->priv_data
, "chunked_post", "0", 0);
1600 /* Initialize the authentication state for the POST session. The HTTP
1601 * protocol implementation doesn't properly handle multi-pass
1602 * authentication for POST requests, since it would require one of
1604 * - implementing Expect: 100-continue, which many HTTP servers
1605 * don't support anyway, even less the RTSP servers that do HTTP
1607 * - sending the whole POST data until getting a 401 reply specifying
1608 * what authentication method to use, then resending all that data
1609 * - waiting for potential 401 replies directly after sending the
1610 * POST header (waiting for some unspecified time)
1611 * Therefore, we copy the full auth state, which works for both basic
1612 * and digest. (For digest, we would have to synchronize the nonce
1613 * count variable between the two sessions, if we'd do more requests
1614 * with the original session, though.)
1616 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1618 /* complete the connection */
1619 if (ffurl_connect(rt
->rtsp_hd_out
, NULL
)) {
1624 /* open the tcp connection */
1625 ff_url_join(tcpname
, sizeof(tcpname
), "tcp", NULL
, host
, port
, NULL
);
1626 if (ffurl_open(&rt
->rtsp_hd
, tcpname
, AVIO_FLAG_READ_WRITE
,
1627 &s
->interrupt_callback
, NULL
) < 0) {
1631 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1635 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1636 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1637 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1638 NULL
, 0, NI_NUMERICHOST
);
1641 /* request options supported by the server; this also detects server
1643 for (rt
->server_type
= RTSP_SERVER_RTP
;;) {
1645 if (rt
->server_type
== RTSP_SERVER_REAL
)
1648 * The following entries are required for proper
1649 * streaming from a Realmedia server. They are
1650 * interdependent in some way although we currently
1651 * don't quite understand how. Values were copied
1652 * from mplayer SVN r23589.
1653 * ClientChallenge is a 16-byte ID in hex
1654 * CompanyID is a 16-byte ID in base64
1656 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1657 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1658 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1659 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1661 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
1662 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1663 err
= AVERROR_INVALIDDATA
;
1667 /* detect server type if not standard-compliant RTP */
1668 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
1669 rt
->server_type
= RTSP_SERVER_REAL
;
1671 } else if (!av_strncasecmp(reply
->server
, "WMServer/", 9)) {
1672 rt
->server_type
= RTSP_SERVER_WMS
;
1673 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
1674 strcpy(real_challenge
, reply
->real_challenge
);
1678 if (s
->iformat
&& CONFIG_RTSP_DEMUXER
)
1679 err
= ff_rtsp_setup_input_streams(s
, reply
);
1680 else if (CONFIG_RTSP_MUXER
)
1681 err
= ff_rtsp_setup_output_streams(s
, host
);
1686 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
1687 ~(lower_transport_mask
- 1)];
1689 err
= ff_rtsp_make_setup_request(s
, host
, port
, lower_transport
,
1690 rt
->server_type
== RTSP_SERVER_REAL
?
1691 real_challenge
: NULL
);
1694 lower_transport_mask
&= ~(1 << lower_transport
);
1695 if (lower_transport_mask
== 0 && err
== 1) {
1696 err
= AVERROR(EPROTONOSUPPORT
);
1701 rt
->lower_transport_mask
= lower_transport_mask
;
1702 av_strlcpy(rt
->real_challenge
, real_challenge
, sizeof(rt
->real_challenge
));
1703 rt
->state
= RTSP_STATE_IDLE
;
1704 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
1707 ff_rtsp_close_streams(s
);
1708 ff_rtsp_close_connections(s
);
1709 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
1710 av_strlcpy(s
->filename
, reply
->location
, sizeof(s
->filename
));
1711 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
1719 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1722 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1723 uint8_t *buf
, int buf_size
, int64_t wait_end
)
1725 RTSPState
*rt
= s
->priv_data
;
1726 RTSPStream
*rtsp_st
;
1727 int n
, i
, ret
, tcp_fd
, timeout_cnt
= 0;
1729 struct pollfd
*p
= rt
->p
;
1730 int *fds
= NULL
, fdsnum
, fdsidx
;
1733 if (ff_check_interrupt(&s
->interrupt_callback
))
1734 return AVERROR_EXIT
;
1735 if (wait_end
&& wait_end
- av_gettime() < 0)
1736 return AVERROR(EAGAIN
);
1739 tcp_fd
= ffurl_get_file_handle(rt
->rtsp_hd
);
1740 p
[max_p
].fd
= tcp_fd
;
1741 p
[max_p
++].events
= POLLIN
;
1745 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1746 rtsp_st
= rt
->rtsp_streams
[i
];
1747 if (rtsp_st
->rtp_handle
) {
1748 if (ret
= ffurl_get_multi_file_handle(rtsp_st
->rtp_handle
,
1750 av_log(s
, AV_LOG_ERROR
, "Unable to recover rtp ports\n");
1754 av_log(s
, AV_LOG_ERROR
,
1755 "Number of fds %d not supported\n", fdsnum
);
1756 return AVERROR_INVALIDDATA
;
1758 for (fdsidx
= 0; fdsidx
< fdsnum
; fdsidx
++) {
1759 p
[max_p
].fd
= fds
[fdsidx
];
1760 p
[max_p
++].events
= POLLIN
;
1765 n
= poll(p
, max_p
, POLL_TIMEOUT_MS
);
1767 int j
= 1 - (tcp_fd
== -1);
1769 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1770 rtsp_st
= rt
->rtsp_streams
[i
];
1771 if (rtsp_st
->rtp_handle
) {
1772 if (p
[j
].revents
& POLLIN
|| p
[j
+1].revents
& POLLIN
) {
1773 ret
= ffurl_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
1775 *prtsp_st
= rtsp_st
;
1782 #if CONFIG_RTSP_DEMUXER
1783 if (tcp_fd
!= -1 && p
[0].revents
& POLLIN
) {
1784 if (rt
->rtsp_flags
& RTSP_FLAG_LISTEN
) {
1785 if (rt
->state
== RTSP_STATE_STREAMING
) {
1786 if (!ff_rtsp_parse_streaming_commands(s
))
1789 av_log(s
, AV_LOG_WARNING
,
1790 "Unable to answer to TEARDOWN\n");
1794 RTSPMessageHeader reply
;
1795 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0, NULL
);
1798 /* XXX: parse message */
1799 if (rt
->state
!= RTSP_STATE_STREAMING
)
1804 } else if (n
== 0 && ++timeout_cnt
>= MAX_TIMEOUTS
) {
1805 return AVERROR(ETIMEDOUT
);
1806 } else if (n
< 0 && errno
!= EINTR
)
1807 return AVERROR(errno
);
1811 static int pick_stream(AVFormatContext
*s
, RTSPStream
**rtsp_st
,
1812 const uint8_t *buf
, int len
)
1814 RTSPState
*rt
= s
->priv_data
;
1818 if (rt
->nb_rtsp_streams
== 1) {
1819 *rtsp_st
= rt
->rtsp_streams
[0];
1822 if (len
>= 8 && rt
->transport
== RTSP_TRANSPORT_RTP
) {
1823 if (RTP_PT_IS_RTCP(rt
->recvbuf
[1])) {
1825 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1826 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
1829 if (rtpctx
->ssrc
== AV_RB32(&buf
[4])) {
1830 *rtsp_st
= rt
->rtsp_streams
[i
];
1837 av_log(s
, AV_LOG_WARNING
,
1838 "Unable to pick stream for packet - SSRC not known for "
1840 return AVERROR(EAGAIN
);
1843 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1844 if ((buf
[1] & 0x7f) == rt
->rtsp_streams
[i
]->sdp_payload_type
) {
1845 *rtsp_st
= rt
->rtsp_streams
[i
];
1851 av_log(s
, AV_LOG_WARNING
, "Unable to pick stream for packet\n");
1852 return AVERROR(EAGAIN
);
1855 int ff_rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
1857 RTSPState
*rt
= s
->priv_data
;
1859 RTSPStream
*rtsp_st
, *first_queue_st
= NULL
;
1860 int64_t wait_end
= 0;
1862 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1865 /* get next frames from the same RTP packet */
1866 if (rt
->cur_transport_priv
) {
1867 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1868 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1869 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
1870 ret
= ff_rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1871 } else if (rt
->ts
&& CONFIG_RTPDEC
) {
1872 ret
= ff_mpegts_parse_packet(rt
->ts
, pkt
, rt
->recvbuf
+ rt
->recvbuf_pos
, rt
->recvbuf_len
- rt
->recvbuf_pos
);
1874 rt
->recvbuf_pos
+= ret
;
1875 ret
= rt
->recvbuf_pos
< rt
->recvbuf_len
;
1880 rt
->cur_transport_priv
= NULL
;
1882 } else if (ret
== 1) {
1885 rt
->cur_transport_priv
= NULL
;
1889 if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
1891 int64_t first_queue_time
= 0;
1892 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1893 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
1897 queue_time
= ff_rtp_queued_packet_time(rtpctx
);
1898 if (queue_time
&& (queue_time
- first_queue_time
< 0 ||
1899 !first_queue_time
)) {
1900 first_queue_time
= queue_time
;
1901 first_queue_st
= rt
->rtsp_streams
[i
];
1904 if (first_queue_time
) {
1905 wait_end
= first_queue_time
+ s
->max_delay
;
1908 first_queue_st
= NULL
;
1912 /* read next RTP packet */
1914 rt
->recvbuf
= av_malloc(RECVBUF_SIZE
);
1916 return AVERROR(ENOMEM
);
1919 switch(rt
->lower_transport
) {
1921 #if CONFIG_RTSP_DEMUXER
1922 case RTSP_LOWER_TRANSPORT_TCP
:
1923 len
= ff_rtsp_tcp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
);
1926 case RTSP_LOWER_TRANSPORT_UDP
:
1927 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
1928 len
= udp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
, wait_end
);
1929 if (len
> 0 && rtsp_st
->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
1930 ff_rtp_check_and_send_back_rr(rtsp_st
->transport_priv
, rtsp_st
->rtp_handle
, NULL
, len
);
1932 case RTSP_LOWER_TRANSPORT_CUSTOM
:
1933 if (first_queue_st
&& rt
->transport
== RTSP_TRANSPORT_RTP
&&
1934 wait_end
&& wait_end
< av_gettime())
1935 len
= AVERROR(EAGAIN
);
1937 len
= ffio_read_partial(s
->pb
, rt
->recvbuf
, RECVBUF_SIZE
);
1938 len
= pick_stream(s
, &rtsp_st
, rt
->recvbuf
, len
);
1939 if (len
> 0 && rtsp_st
->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
1940 ff_rtp_check_and_send_back_rr(rtsp_st
->transport_priv
, NULL
, s
->pb
, len
);
1943 if (len
== AVERROR(EAGAIN
) && first_queue_st
&&
1944 rt
->transport
== RTSP_TRANSPORT_RTP
) {
1945 rtsp_st
= first_queue_st
;
1946 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, NULL
, 0);
1953 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1954 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1955 } else if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
1956 ret
= ff_rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1957 if (rtsp_st
->feedback
) {
1958 AVIOContext
*pb
= NULL
;
1959 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_CUSTOM
)
1961 ff_rtp_send_rtcp_feedback(rtsp_st
->transport_priv
, rtsp_st
->rtp_handle
, pb
);
1964 /* Either bad packet, or a RTCP packet. Check if the
1965 * first_rtcp_ntp_time field was initialized. */
1966 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1967 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
1968 /* first_rtcp_ntp_time has been initialized for this stream,
1969 * copy the same value to all other uninitialized streams,
1970 * in order to map their timestamp origin to the same ntp time
1973 AVStream
*st
= NULL
;
1974 if (rtsp_st
->stream_index
>= 0)
1975 st
= s
->streams
[rtsp_st
->stream_index
];
1976 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1977 RTPDemuxContext
*rtpctx2
= rt
->rtsp_streams
[i
]->transport_priv
;
1978 AVStream
*st2
= NULL
;
1979 if (rt
->rtsp_streams
[i
]->stream_index
>= 0)
1980 st2
= s
->streams
[rt
->rtsp_streams
[i
]->stream_index
];
1981 if (rtpctx2
&& st
&& st2
&&
1982 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) {
1983 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
1984 rtpctx2
->rtcp_ts_offset
= av_rescale_q(
1985 rtpctx
->rtcp_ts_offset
, st
->time_base
,
1990 if (ret
== -RTCP_BYE
) {
1993 av_log(s
, AV_LOG_DEBUG
, "Received BYE for stream %d (%d/%d)\n",
1994 rtsp_st
->stream_index
, rt
->nb_byes
, rt
->nb_rtsp_streams
);
1996 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
2000 } else if (rt
->ts
&& CONFIG_RTPDEC
) {
2001 ret
= ff_mpegts_parse_packet(rt
->ts
, pkt
, rt
->recvbuf
, len
);
2004 rt
->recvbuf_len
= len
;
2005 rt
->recvbuf_pos
= ret
;
2006 rt
->cur_transport_priv
= rt
->ts
;
2013 return AVERROR_INVALIDDATA
;
2019 /* more packets may follow, so we save the RTP context */
2020 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
2024 #endif /* CONFIG_RTPDEC */
2026 #if CONFIG_SDP_DEMUXER
2027 static int sdp_probe(AVProbeData
*p1
)
2029 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
2031 /* we look for a line beginning "c=IN IP" */
2032 while (p
< p_end
&& *p
!= '\0') {
2033 if (p
+ sizeof("c=IN IP") - 1 < p_end
&&
2034 av_strstart(p
, "c=IN IP", NULL
))
2035 return AVPROBE_SCORE_MAX
/ 2;
2037 while (p
< p_end
- 1 && *p
!= '\n') p
++;
2046 static int sdp_read_header(AVFormatContext
*s
)
2048 RTSPState
*rt
= s
->priv_data
;
2049 RTSPStream
*rtsp_st
;
2054 if (!ff_network_init())
2055 return AVERROR(EIO
);
2057 if (s
->max_delay
< 0) /* Not set by the caller */
2058 s
->max_delay
= DEFAULT_REORDERING_DELAY
;
2059 if (rt
->rtsp_flags
& RTSP_FLAG_CUSTOM_IO
)
2060 rt
->lower_transport
= RTSP_LOWER_TRANSPORT_CUSTOM
;
2062 /* read the whole sdp file */
2063 /* XXX: better loading */
2064 content
= av_malloc(SDP_MAX_SIZE
);
2065 size
= avio_read(s
->pb
, content
, SDP_MAX_SIZE
- 1);
2068 return AVERROR_INVALIDDATA
;
2070 content
[size
] ='\0';
2072 err
= ff_sdp_parse(s
, content
);
2076 /* open each RTP stream */
2077 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2079 rtsp_st
= rt
->rtsp_streams
[i
];
2081 if (!(rt
->rtsp_flags
& RTSP_FLAG_CUSTOM_IO
)) {
2082 getnameinfo((struct sockaddr
*) &rtsp_st
->sdp_ip
, sizeof(rtsp_st
->sdp_ip
),
2083 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
2084 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
2085 namebuf
, rtsp_st
->sdp_port
,
2086 "?localport=%d&ttl=%d&connect=%d", rtsp_st
->sdp_port
,
2088 rt
->rtsp_flags
& RTSP_FLAG_FILTER_SRC
? 1 : 0);
2089 if (ffurl_open(&rtsp_st
->rtp_handle
, url
, AVIO_FLAG_READ_WRITE
,
2090 &s
->interrupt_callback
, NULL
) < 0) {
2091 err
= AVERROR_INVALIDDATA
;
2095 if ((err
= ff_rtsp_open_transport_ctx(s
, rtsp_st
)))
2100 ff_rtsp_close_streams(s
);
2105 static int sdp_read_close(AVFormatContext
*s
)
2107 ff_rtsp_close_streams(s
);
2112 static const AVClass sdp_demuxer_class
= {
2113 .class_name
= "SDP demuxer",
2114 .item_name
= av_default_item_name
,
2115 .option
= sdp_options
,
2116 .version
= LIBAVUTIL_VERSION_INT
,
2119 AVInputFormat ff_sdp_demuxer
= {
2121 .long_name
= NULL_IF_CONFIG_SMALL("SDP"),
2122 .priv_data_size
= sizeof(RTSPState
),
2123 .read_probe
= sdp_probe
,
2124 .read_header
= sdp_read_header
,
2125 .read_packet
= ff_rtsp_fetch_packet
,
2126 .read_close
= sdp_read_close
,
2127 .priv_class
= &sdp_demuxer_class
,
2129 #endif /* CONFIG_SDP_DEMUXER */
2131 #if CONFIG_RTP_DEMUXER
2132 static int rtp_probe(AVProbeData
*p
)
2134 if (av_strstart(p
->filename
, "rtp:", NULL
))
2135 return AVPROBE_SCORE_MAX
;
2139 static int rtp_read_header(AVFormatContext
*s
)
2141 uint8_t recvbuf
[1500];
2142 char host
[500], sdp
[500];
2144 URLContext
* in
= NULL
;
2146 AVCodecContext codec
= { 0 };
2147 struct sockaddr_storage addr
;
2149 socklen_t addrlen
= sizeof(addr
);
2150 RTSPState
*rt
= s
->priv_data
;
2152 if (!ff_network_init())
2153 return AVERROR(EIO
);
2155 ret
= ffurl_open(&in
, s
->filename
, AVIO_FLAG_READ
,
2156 &s
->interrupt_callback
, NULL
);
2161 ret
= ffurl_read(in
, recvbuf
, sizeof(recvbuf
));
2162 if (ret
== AVERROR(EAGAIN
))
2167 av_log(s
, AV_LOG_WARNING
, "Received too short packet\n");
2171 if ((recvbuf
[0] & 0xc0) != 0x80) {
2172 av_log(s
, AV_LOG_WARNING
, "Unsupported RTP version packet "
2177 if (RTP_PT_IS_RTCP(recvbuf
[1]))
2180 payload_type
= recvbuf
[1] & 0x7f;
2183 getsockname(ffurl_get_file_handle(in
), (struct sockaddr
*) &addr
, &addrlen
);
2187 if (ff_rtp_get_codec_info(&codec
, payload_type
)) {
2188 av_log(s
, AV_LOG_ERROR
, "Unable to receive RTP payload type %d "
2189 "without an SDP file describing it\n",
2193 if (codec
.codec_type
!= AVMEDIA_TYPE_DATA
) {
2194 av_log(s
, AV_LOG_WARNING
, "Guessing on RTP content - if not received "
2195 "properly you need an SDP file "
2199 av_url_split(NULL
, 0, NULL
, 0, host
, sizeof(host
), &port
,
2200 NULL
, 0, s
->filename
);
2202 snprintf(sdp
, sizeof(sdp
),
2203 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2204 addr
.ss_family
== AF_INET
? 4 : 6, host
,
2205 codec
.codec_type
== AVMEDIA_TYPE_DATA
? "application" :
2206 codec
.codec_type
== AVMEDIA_TYPE_VIDEO
? "video" : "audio",
2207 port
, payload_type
);
2208 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
);
2210 ffio_init_context(&pb
, sdp
, strlen(sdp
), 0, NULL
, NULL
, NULL
, NULL
);
2213 /* sdp_read_header initializes this again */
2216 rt
->media_type_mask
= (1 << (AVMEDIA_TYPE_DATA
+1)) - 1;
2218 ret
= sdp_read_header(s
);
2229 static const AVClass rtp_demuxer_class
= {
2230 .class_name
= "RTP demuxer",
2231 .item_name
= av_default_item_name
,
2232 .option
= rtp_options
,
2233 .version
= LIBAVUTIL_VERSION_INT
,
2236 AVInputFormat ff_rtp_demuxer
= {
2238 .long_name
= NULL_IF_CONFIG_SMALL("RTP input"),
2239 .priv_data_size
= sizeof(RTSPState
),
2240 .read_probe
= rtp_probe
,
2241 .read_header
= rtp_read_header
,
2242 .read_packet
= ff_rtsp_fetch_packet
,
2243 .read_close
= sdp_read_close
,
2244 .flags
= AVFMT_NOFILE
,
2245 .priv_class
= &rtp_demuxer_class
,
2247 #endif /* CONFIG_RTP_DEMUXER */