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