blob: 852fd6704e1de4777b9e3e41dbc2d30ac9cc4b3e
1 | /* |
2 | * RTSP definitions |
3 | * Copyright (c) 2002 Fabrice Bellard |
4 | * |
5 | * This file is part of FFmpeg. |
6 | * |
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. |
11 | * |
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. |
16 | * |
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 |
20 | */ |
21 | #ifndef AVFORMAT_RTSP_H |
22 | #define AVFORMAT_RTSP_H |
23 | |
24 | #include <stdint.h> |
25 | #include "avformat.h" |
26 | #include "rtspcodes.h" |
27 | #include "rtpdec.h" |
28 | #include "network.h" |
29 | #include "httpauth.h" |
30 | |
31 | #include "libavutil/log.h" |
32 | #include "libavutil/opt.h" |
33 | |
34 | /** |
35 | * Network layer over which RTP/etc packet data will be transported. |
36 | */ |
37 | enum RTSPLowerTransport { |
38 | RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */ |
39 | RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */ |
40 | RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */ |
41 | RTSP_LOWER_TRANSPORT_NB, |
42 | RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper |
43 | transport mode as such, |
44 | only for use via AVOptions */ |
45 | RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public |
46 | option for lower_transport_mask, |
47 | but set in the SDP demuxer based |
48 | on a flag. */ |
49 | }; |
50 | |
51 | /** |
52 | * Packet profile of the data that we will be receiving. Real servers |
53 | * commonly send RDT (although they can sometimes send RTP as well), |
54 | * whereas most others will send RTP. |
55 | */ |
56 | enum RTSPTransport { |
57 | RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ |
58 | RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ |
59 | RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */ |
60 | RTSP_TRANSPORT_NB |
61 | }; |
62 | |
63 | /** |
64 | * Transport mode for the RTSP data. This may be plain, or |
65 | * tunneled, which is done over HTTP. |
66 | */ |
67 | enum RTSPControlTransport { |
68 | RTSP_MODE_PLAIN, /**< Normal RTSP */ |
69 | RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */ |
70 | }; |
71 | |
72 | #define RTSP_DEFAULT_PORT 554 |
73 | #define RTSPS_DEFAULT_PORT 322 |
74 | #define RTSP_MAX_TRANSPORTS 8 |
75 | #define RTSP_TCP_MAX_PACKET_SIZE 1472 |
76 | #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1 |
77 | #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 |
78 | #define RTSP_RTP_PORT_MIN 5000 |
79 | #define RTSP_RTP_PORT_MAX 65000 |
80 | |
81 | /** |
82 | * This describes a single item in the "Transport:" line of one stream as |
83 | * negotiated by the SETUP RTSP command. Multiple transports are comma- |
84 | * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp; |
85 | * client_port=1000-1001;server_port=1800-1801") and described in separate |
86 | * RTSPTransportFields. |
87 | */ |
88 | typedef struct RTSPTransportField { |
89 | /** interleave ids, if TCP transport; each TCP/RTSP data packet starts |
90 | * with a '$', stream length and stream ID. If the stream ID is within |
91 | * the range of this interleaved_min-max, then the packet belongs to |
92 | * this stream. */ |
93 | int interleaved_min, interleaved_max; |
94 | |
95 | /** UDP multicast port range; the ports to which we should connect to |
96 | * receive multicast UDP data. */ |
97 | int port_min, port_max; |
98 | |
99 | /** UDP client ports; these should be the local ports of the UDP RTP |
100 | * (and RTCP) sockets over which we receive RTP/RTCP data. */ |
101 | int client_port_min, client_port_max; |
102 | |
103 | /** UDP unicast server port range; the ports to which we should connect |
104 | * to receive unicast UDP RTP/RTCP data. */ |
105 | int server_port_min, server_port_max; |
106 | |
107 | /** time-to-live value (required for multicast); the amount of HOPs that |
108 | * packets will be allowed to make before being discarded. */ |
109 | int ttl; |
110 | |
111 | /** transport set to record data */ |
112 | int mode_record; |
113 | |
114 | struct sockaddr_storage destination; /**< destination IP address */ |
115 | char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */ |
116 | |
117 | /** data/packet transport protocol; e.g. RTP or RDT */ |
118 | enum RTSPTransport transport; |
119 | |
120 | /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */ |
121 | enum RTSPLowerTransport lower_transport; |
122 | } RTSPTransportField; |
123 | |
124 | /** |
125 | * This describes the server response to each RTSP command. |
126 | */ |
127 | typedef struct RTSPMessageHeader { |
128 | /** length of the data following this header */ |
129 | int content_length; |
130 | |
131 | enum RTSPStatusCode status_code; /**< response code from server */ |
132 | |
133 | /** number of items in the 'transports' variable below */ |
134 | int nb_transports; |
135 | |
136 | /** Time range of the streams that the server will stream. In |
137 | * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ |
138 | int64_t range_start, range_end; |
139 | |
140 | /** describes the complete "Transport:" line of the server in response |
141 | * to a SETUP RTSP command by the client */ |
142 | RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; |
143 | |
144 | int seq; /**< sequence number */ |
145 | |
146 | /** the "Session:" field. This value is initially set by the server and |
147 | * should be re-transmitted by the client in every RTSP command. */ |
148 | char session_id[512]; |
149 | |
150 | /** the "Location:" field. This value is used to handle redirection. |
151 | */ |
152 | char location[4096]; |
153 | |
154 | /** the "RealChallenge1:" field from the server */ |
155 | char real_challenge[64]; |
156 | |
157 | /** the "Server: field, which can be used to identify some special-case |
158 | * servers that are not 100% standards-compliant. We use this to identify |
159 | * Windows Media Server, which has a value "WMServer/v.e.r.sion", where |
160 | * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers |
161 | * use something like "Helix [..] Server Version v.e.r.sion (platform) |
162 | * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)", |
163 | * where platform is the output of $uname -msr | sed 's/ /-/g'. */ |
164 | char server[64]; |
165 | |
166 | /** The "timeout" comes as part of the server response to the "SETUP" |
167 | * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the |
168 | * time, in seconds, that the server will go without traffic over the |
169 | * RTSP/TCP connection before it closes the connection. To prevent |
170 | * this, sent dummy requests (e.g. OPTIONS) with intervals smaller |
171 | * than this value. */ |
172 | int timeout; |
173 | |
174 | /** The "Notice" or "X-Notice" field value. See |
175 | * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00 |
176 | * for a complete list of supported values. */ |
177 | int notice; |
178 | |
179 | /** The "reason" is meant to specify better the meaning of the error code |
180 | * returned |
181 | */ |
182 | char reason[256]; |
183 | |
184 | /** |
185 | * Content type header |
186 | */ |
187 | char content_type[64]; |
188 | } RTSPMessageHeader; |
189 | |
190 | /** |
191 | * Client state, i.e. whether we are currently receiving data (PLAYING) or |
192 | * setup-but-not-receiving (PAUSED). State can be changed in applications |
193 | * by calling av_read_play/pause(). |
194 | */ |
195 | enum RTSPClientState { |
196 | RTSP_STATE_IDLE, /**< not initialized */ |
197 | RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ |
198 | RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ |
199 | RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ |
200 | }; |
201 | |
202 | /** |
203 | * Identify particular servers that require special handling, such as |
204 | * standards-incompliant "Transport:" lines in the SETUP request. |
205 | */ |
206 | enum RTSPServerType { |
207 | RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ |
208 | RTSP_SERVER_REAL, /**< Realmedia-style server */ |
209 | RTSP_SERVER_WMS, /**< Windows Media server */ |
210 | RTSP_SERVER_NB |
211 | }; |
212 | |
213 | /** |
214 | * Private data for the RTSP demuxer. |
215 | * |
216 | * @todo Use AVIOContext instead of URLContext |
217 | */ |
218 | typedef struct RTSPState { |
219 | const AVClass *class; /**< Class for private options. */ |
220 | URLContext *rtsp_hd; /* RTSP TCP connection handle */ |
221 | |
222 | /** number of items in the 'rtsp_streams' variable */ |
223 | int nb_rtsp_streams; |
224 | |
225 | struct RTSPStream **rtsp_streams; /**< streams in this session */ |
226 | |
227 | /** indicator of whether we are currently receiving data from the |
228 | * server. Basically this isn't more than a simple cache of the |
229 | * last PLAY/PAUSE command sent to the server, to make sure we don't |
230 | * send 2x the same unexpectedly or commands in the wrong state. */ |
231 | enum RTSPClientState state; |
232 | |
233 | /** the seek value requested when calling av_seek_frame(). This value |
234 | * is subsequently used as part of the "Range" parameter when emitting |
235 | * the RTSP PLAY command. If we are currently playing, this command is |
236 | * called instantly. If we are currently paused, this command is called |
237 | * whenever we resume playback. Either way, the value is only used once, |
238 | * see rtsp_read_play() and rtsp_read_seek(). */ |
239 | int64_t seek_timestamp; |
240 | |
241 | int seq; /**< RTSP command sequence number */ |
242 | |
243 | /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session |
244 | * identifier that the client should re-transmit in each RTSP command */ |
245 | char session_id[512]; |
246 | |
247 | /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that |
248 | * the server will go without traffic on the RTSP/TCP line before it |
249 | * closes the connection. */ |
250 | int timeout; |
251 | |
252 | /** timestamp of the last RTSP command that we sent to the RTSP server. |
253 | * This is used to calculate when to send dummy commands to keep the |
254 | * connection alive, in conjunction with timeout. */ |
255 | int64_t last_cmd_time; |
256 | |
257 | /** the negotiated data/packet transport protocol; e.g. RTP or RDT */ |
258 | enum RTSPTransport transport; |
259 | |
260 | /** the negotiated network layer transport protocol; e.g. TCP or UDP |
261 | * uni-/multicast */ |
262 | enum RTSPLowerTransport lower_transport; |
263 | |
264 | /** brand of server that we're talking to; e.g. WMS, REAL or other. |
265 | * Detected based on the value of RTSPMessageHeader->server or the presence |
266 | * of RTSPMessageHeader->real_challenge */ |
267 | enum RTSPServerType server_type; |
268 | |
269 | /** the "RealChallenge1:" field from the server */ |
270 | char real_challenge[64]; |
271 | |
272 | /** plaintext authorization line (username:password) */ |
273 | char auth[128]; |
274 | |
275 | /** authentication state */ |
276 | HTTPAuthState auth_state; |
277 | |
278 | /** The last reply of the server to a RTSP command */ |
279 | char last_reply[2048]; /* XXX: allocate ? */ |
280 | |
281 | /** RTSPStream->transport_priv of the last stream that we read a |
282 | * packet from */ |
283 | void *cur_transport_priv; |
284 | |
285 | /** The following are used for Real stream selection */ |
286 | //@{ |
287 | /** whether we need to send a "SET_PARAMETER Subscribe:" command */ |
288 | int need_subscription; |
289 | |
290 | /** stream setup during the last frame read. This is used to detect if |
291 | * we need to subscribe or unsubscribe to any new streams. */ |
292 | enum AVDiscard *real_setup_cache; |
293 | |
294 | /** current stream setup. This is a temporary buffer used to compare |
295 | * current setup to previous frame setup. */ |
296 | enum AVDiscard *real_setup; |
297 | |
298 | /** the last value of the "SET_PARAMETER Subscribe:" RTSP command. |
299 | * this is used to send the same "Unsubscribe:" if stream setup changed, |
300 | * before sending a new "Subscribe:" command. */ |
301 | char last_subscription[1024]; |
302 | //@} |
303 | |
304 | /** The following are used for RTP/ASF streams */ |
305 | //@{ |
306 | /** ASF demuxer context for the embedded ASF stream from WMS servers */ |
307 | AVFormatContext *asf_ctx; |
308 | |
309 | /** cache for position of the asf demuxer, since we load a new |
310 | * data packet in the bytecontext for each incoming RTSP packet. */ |
311 | uint64_t asf_pb_pos; |
312 | //@} |
313 | |
314 | /** some MS RTSP streams contain a URL in the SDP that we need to use |
315 | * for all subsequent RTSP requests, rather than the input URI; in |
316 | * other cases, this is a copy of AVFormatContext->filename. */ |
317 | char control_uri[1024]; |
318 | |
319 | /** The following are used for parsing raw mpegts in udp */ |
320 | //@{ |
321 | struct MpegTSContext *ts; |
322 | int recvbuf_pos; |
323 | int recvbuf_len; |
324 | //@} |
325 | |
326 | /** Additional output handle, used when input and output are done |
327 | * separately, eg for HTTP tunneling. */ |
328 | URLContext *rtsp_hd_out; |
329 | |
330 | /** RTSP transport mode, such as plain or tunneled. */ |
331 | enum RTSPControlTransport control_transport; |
332 | |
333 | /* Number of RTCP BYE packets the RTSP session has received. |
334 | * An EOF is propagated back if nb_byes == nb_streams. |
335 | * This is reset after a seek. */ |
336 | int nb_byes; |
337 | |
338 | /** Reusable buffer for receiving packets */ |
339 | uint8_t* recvbuf; |
340 | |
341 | /** |
342 | * A mask with all requested transport methods |
343 | */ |
344 | int lower_transport_mask; |
345 | |
346 | /** |
347 | * The number of returned packets |
348 | */ |
349 | uint64_t packets; |
350 | |
351 | /** |
352 | * Polling array for udp |
353 | */ |
354 | struct pollfd *p; |
355 | |
356 | /** |
357 | * Whether the server supports the GET_PARAMETER method. |
358 | */ |
359 | int get_parameter_supported; |
360 | |
361 | /** |
362 | * Do not begin to play the stream immediately. |
363 | */ |
364 | int initial_pause; |
365 | |
366 | /** |
367 | * Option flags for the chained RTP muxer. |
368 | */ |
369 | int rtp_muxer_flags; |
370 | |
371 | /** Whether the server accepts the x-Dynamic-Rate header */ |
372 | int accept_dynamic_rate; |
373 | |
374 | /** |
375 | * Various option flags for the RTSP muxer/demuxer. |
376 | */ |
377 | int rtsp_flags; |
378 | |
379 | /** |
380 | * Mask of all requested media types |
381 | */ |
382 | int media_type_mask; |
383 | |
384 | /** |
385 | * Minimum and maximum local UDP ports. |
386 | */ |
387 | int rtp_port_min, rtp_port_max; |
388 | |
389 | /** |
390 | * Timeout to wait for incoming connections. |
391 | */ |
392 | int initial_timeout; |
393 | |
394 | /** |
395 | * timeout of socket i/o operations. |
396 | */ |
397 | int stimeout; |
398 | |
399 | /** |
400 | * Size of RTP packet reordering queue. |
401 | */ |
402 | int reordering_queue_size; |
403 | |
404 | /** |
405 | * User-Agent string |
406 | */ |
407 | char *user_agent; |
408 | |
409 | char default_lang[4]; |
410 | int buffer_size; |
411 | } RTSPState; |
412 | |
413 | #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets - |
414 | receive packets only from the right |
415 | source address and port. */ |
416 | #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */ |
417 | #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */ |
418 | #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source |
419 | address of received packets. */ |
420 | #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */ |
421 | |
422 | typedef struct RTSPSource { |
423 | char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */ |
424 | } RTSPSource; |
425 | |
426 | /** |
427 | * Describe a single stream, as identified by a single m= line block in the |
428 | * SDP content. In the case of RDT, one RTSPStream can represent multiple |
429 | * AVStreams. In this case, each AVStream in this set has similar content |
430 | * (but different codec/bitrate). |
431 | */ |
432 | typedef struct RTSPStream { |
433 | URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ |
434 | void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */ |
435 | |
436 | /** corresponding stream index, if any. -1 if none (MPEG2TS case) */ |
437 | int stream_index; |
438 | |
439 | /** interleave IDs; copies of RTSPTransportField->interleaved_min/max |
440 | * for the selected transport. Only used for TCP. */ |
441 | int interleaved_min, interleaved_max; |
442 | |
443 | char control_url[1024]; /**< url for this stream (from SDP) */ |
444 | |
445 | /** The following are used only in SDP, not RTSP */ |
446 | //@{ |
447 | int sdp_port; /**< port (from SDP content) */ |
448 | struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */ |
449 | int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */ |
450 | struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */ |
451 | int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */ |
452 | struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */ |
453 | int sdp_ttl; /**< IP Time-To-Live (from SDP content) */ |
454 | int sdp_payload_type; /**< payload type */ |
455 | //@} |
456 | |
457 | /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */ |
458 | //@{ |
459 | /** handler structure */ |
460 | RTPDynamicProtocolHandler *dynamic_handler; |
461 | |
462 | /** private data associated with the dynamic protocol */ |
463 | PayloadContext *dynamic_protocol_context; |
464 | //@} |
465 | |
466 | /** Enable sending RTCP feedback messages according to RFC 4585 */ |
467 | int feedback; |
468 | |
469 | /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */ |
470 | uint32_t ssrc; |
471 | |
472 | char crypto_suite[40]; |
473 | char crypto_params[100]; |
474 | } RTSPStream; |
475 | |
476 | void ff_rtsp_parse_line(AVFormatContext *s, |
477 | RTSPMessageHeader *reply, const char *buf, |
478 | RTSPState *rt, const char *method); |
479 | |
480 | /** |
481 | * Send a command to the RTSP server without waiting for the reply. |
482 | * |
483 | * @see rtsp_send_cmd_with_content_async |
484 | */ |
485 | int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, |
486 | const char *url, const char *headers); |
487 | |
488 | /** |
489 | * Send a command to the RTSP server and wait for the reply. |
490 | * |
491 | * @param s RTSP (de)muxer context |
492 | * @param method the method for the request |
493 | * @param url the target url for the request |
494 | * @param headers extra header lines to include in the request |
495 | * @param reply pointer where the RTSP message header will be stored |
496 | * @param content_ptr pointer where the RTSP message body, if any, will |
497 | * be stored (length is in reply) |
498 | * @param send_content if non-null, the data to send as request body content |
499 | * @param send_content_length the length of the send_content data, or 0 if |
500 | * send_content is null |
501 | * |
502 | * @return zero if success, nonzero otherwise |
503 | */ |
504 | int ff_rtsp_send_cmd_with_content(AVFormatContext *s, |
505 | const char *method, const char *url, |
506 | const char *headers, |
507 | RTSPMessageHeader *reply, |
508 | unsigned char **content_ptr, |
509 | const unsigned char *send_content, |
510 | int send_content_length); |
511 | |
512 | /** |
513 | * Send a command to the RTSP server and wait for the reply. |
514 | * |
515 | * @see rtsp_send_cmd_with_content |
516 | */ |
517 | int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, |
518 | const char *url, const char *headers, |
519 | RTSPMessageHeader *reply, unsigned char **content_ptr); |
520 | |
521 | /** |
522 | * Read a RTSP message from the server, or prepare to read data |
523 | * packets if we're reading data interleaved over the TCP/RTSP |
524 | * connection as well. |
525 | * |
526 | * @param s RTSP (de)muxer context |
527 | * @param reply pointer where the RTSP message header will be stored |
528 | * @param content_ptr pointer where the RTSP message body, if any, will |
529 | * be stored (length is in reply) |
530 | * @param return_on_interleaved_data whether the function may return if we |
531 | * encounter a data marker ('$'), which precedes data |
532 | * packets over interleaved TCP/RTSP connections. If this |
533 | * is set, this function will return 1 after encountering |
534 | * a '$'. If it is not set, the function will skip any |
535 | * data packets (if they are encountered), until a reply |
536 | * has been fully parsed. If no more data is available |
537 | * without parsing a reply, it will return an error. |
538 | * @param method the RTSP method this is a reply to. This affects how |
539 | * some response headers are acted upon. May be NULL. |
540 | * |
541 | * @return 1 if a data packets is ready to be received, -1 on error, |
542 | * and 0 on success. |
543 | */ |
544 | int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, |
545 | unsigned char **content_ptr, |
546 | int return_on_interleaved_data, const char *method); |
547 | |
548 | /** |
549 | * Skip a RTP/TCP interleaved packet. |
550 | */ |
551 | void ff_rtsp_skip_packet(AVFormatContext *s); |
552 | |
553 | /** |
554 | * Connect to the RTSP server and set up the individual media streams. |
555 | * This can be used for both muxers and demuxers. |
556 | * |
557 | * @param s RTSP (de)muxer context |
558 | * |
559 | * @return 0 on success, < 0 on error. Cleans up all allocations done |
560 | * within the function on error. |
561 | */ |
562 | int ff_rtsp_connect(AVFormatContext *s); |
563 | |
564 | /** |
565 | * Close and free all streams within the RTSP (de)muxer |
566 | * |
567 | * @param s RTSP (de)muxer context |
568 | */ |
569 | void ff_rtsp_close_streams(AVFormatContext *s); |
570 | |
571 | /** |
572 | * Close all connection handles within the RTSP (de)muxer |
573 | * |
574 | * @param s RTSP (de)muxer context |
575 | */ |
576 | void ff_rtsp_close_connections(AVFormatContext *s); |
577 | |
578 | /** |
579 | * Get the description of the stream and set up the RTSPStream child |
580 | * objects. |
581 | */ |
582 | int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply); |
583 | |
584 | /** |
585 | * Announce the stream to the server and set up the RTSPStream child |
586 | * objects for each media stream. |
587 | */ |
588 | int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr); |
589 | |
590 | /** |
591 | * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in |
592 | * listen mode. |
593 | */ |
594 | int ff_rtsp_parse_streaming_commands(AVFormatContext *s); |
595 | |
596 | /** |
597 | * Parse an SDP description of streams by populating an RTSPState struct |
598 | * within the AVFormatContext; also allocate the RTP streams and the |
599 | * pollfd array used for UDP streams. |
600 | */ |
601 | int ff_sdp_parse(AVFormatContext *s, const char *content); |
602 | |
603 | /** |
604 | * Receive one RTP packet from an TCP interleaved RTSP stream. |
605 | */ |
606 | int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
607 | uint8_t *buf, int buf_size); |
608 | |
609 | /** |
610 | * Send buffered packets over TCP. |
611 | */ |
612 | int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st); |
613 | |
614 | /** |
615 | * Receive one packet from the RTSPStreams set up in the AVFormatContext |
616 | * (which should contain a RTSPState struct as priv_data). |
617 | */ |
618 | int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt); |
619 | |
620 | /** |
621 | * Do the SETUP requests for each stream for the chosen |
622 | * lower transport mode. |
623 | * @return 0 on success, <0 on error, 1 if protocol is unavailable |
624 | */ |
625 | int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, |
626 | int lower_transport, const char *real_challenge); |
627 | |
628 | /** |
629 | * Undo the effect of ff_rtsp_make_setup_request, close the |
630 | * transport_priv and rtp_handle fields. |
631 | */ |
632 | void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets); |
633 | |
634 | /** |
635 | * Open RTSP transport context. |
636 | */ |
637 | int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st); |
638 | |
639 | extern const AVOption ff_rtsp_options[]; |
640 | |
641 | #endif /* AVFORMAT_RTSP_H */ |
642 |