a738a3d4349336981e349bb4caa948496d6d19ab
[libav.git] / libavformat / rtsp.h
1 /*
2 * RTSP definitions
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
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.
11 *
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.
16 *
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
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 };
46
47 /**
48 * Packet profile of the data that we will be receiving. Real servers
49 * commonly send RDT (although they can sometimes send RTP as well),
50 * whereas most others will send RTP.
51 */
52 enum RTSPTransport {
53 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
54 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
55 RTSP_TRANSPORT_NB
56 };
57
58 /**
59 * Transport mode for the RTSP data. This may be plain, or
60 * tunneled, which is done over HTTP.
61 */
62 enum RTSPControlTransport {
63 RTSP_MODE_PLAIN, /**< Normal RTSP */
64 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
65 };
66
67 #define RTSP_DEFAULT_PORT 554
68 #define RTSP_MAX_TRANSPORTS 8
69 #define RTSP_TCP_MAX_PACKET_SIZE 1472
70 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
71 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
72 #define RTSP_RTP_PORT_MIN 5000
73 #define RTSP_RTP_PORT_MAX 10000
74
75 /**
76 * This describes a single item in the "Transport:" line of one stream as
77 * negotiated by the SETUP RTSP command. Multiple transports are comma-
78 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
79 * client_port=1000-1001;server_port=1800-1801") and described in separate
80 * RTSPTransportFields.
81 */
82 typedef struct RTSPTransportField {
83 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
84 * with a '$', stream length and stream ID. If the stream ID is within
85 * the range of this interleaved_min-max, then the packet belongs to
86 * this stream. */
87 int interleaved_min, interleaved_max;
88
89 /** UDP multicast port range; the ports to which we should connect to
90 * receive multicast UDP data. */
91 int port_min, port_max;
92
93 /** UDP client ports; these should be the local ports of the UDP RTP
94 * (and RTCP) sockets over which we receive RTP/RTCP data. */
95 int client_port_min, client_port_max;
96
97 /** UDP unicast server port range; the ports to which we should connect
98 * to receive unicast UDP RTP/RTCP data. */
99 int server_port_min, server_port_max;
100
101 /** time-to-live value (required for multicast); the amount of HOPs that
102 * packets will be allowed to make before being discarded. */
103 int ttl;
104
105 /** transport set to record data */
106 int mode_record;
107
108 struct sockaddr_storage destination; /**< destination IP address */
109 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
110
111 /** data/packet transport protocol; e.g. RTP or RDT */
112 enum RTSPTransport transport;
113
114 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
115 enum RTSPLowerTransport lower_transport;
116 } RTSPTransportField;
117
118 /**
119 * This describes the server response to each RTSP command.
120 */
121 typedef struct RTSPMessageHeader {
122 /** length of the data following this header */
123 int content_length;
124
125 enum RTSPStatusCode status_code; /**< response code from server */
126
127 /** number of items in the 'transports' variable below */
128 int nb_transports;
129
130 /** Time range of the streams that the server will stream. In
131 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
132 int64_t range_start, range_end;
133
134 /** describes the complete "Transport:" line of the server in response
135 * to a SETUP RTSP command by the client */
136 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
137
138 int seq; /**< sequence number */
139
140 /** the "Session:" field. This value is initially set by the server and
141 * should be re-transmitted by the client in every RTSP command. */
142 char session_id[512];
143
144 /** the "Location:" field. This value is used to handle redirection.
145 */
146 char location[4096];
147
148 /** the "RealChallenge1:" field from the server */
149 char real_challenge[64];
150
151 /** the "Server: field, which can be used to identify some special-case
152 * servers that are not 100% standards-compliant. We use this to identify
153 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
154 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
155 * use something like "Helix [..] Server Version v.e.r.sion (platform)
156 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
157 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
158 char server[64];
159
160 /** The "timeout" comes as part of the server response to the "SETUP"
161 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
162 * time, in seconds, that the server will go without traffic over the
163 * RTSP/TCP connection before it closes the connection. To prevent
164 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
165 * than this value. */
166 int timeout;
167
168 /** The "Notice" or "X-Notice" field value. See
169 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
170 * for a complete list of supported values. */
171 int notice;
172
173 /** The "reason" is meant to specify better the meaning of the error code
174 * returned
175 */
176 char reason[256];
177
178 /**
179 * Content type header
180 */
181 char content_type[64];
182 } RTSPMessageHeader;
183
184 /**
185 * Client state, i.e. whether we are currently receiving data (PLAYING) or
186 * setup-but-not-receiving (PAUSED). State can be changed in applications
187 * by calling av_read_play/pause().
188 */
189 enum RTSPClientState {
190 RTSP_STATE_IDLE, /**< not initialized */
191 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
192 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
193 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
194 };
195
196 /**
197 * Identify particular servers that require special handling, such as
198 * standards-incompliant "Transport:" lines in the SETUP request.
199 */
200 enum RTSPServerType {
201 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
202 RTSP_SERVER_REAL, /**< Realmedia-style server */
203 RTSP_SERVER_WMS, /**< Windows Media server */
204 RTSP_SERVER_NB
205 };
206
207 /**
208 * Private data for the RTSP demuxer.
209 *
210 * @todo Use AVIOContext instead of URLContext
211 */
212 typedef struct RTSPState {
213 const AVClass *class; /**< Class for private options. */
214 URLContext *rtsp_hd; /* RTSP TCP connection handle */
215
216 /** number of items in the 'rtsp_streams' variable */
217 int nb_rtsp_streams;
218
219 struct RTSPStream **rtsp_streams; /**< streams in this session */
220
221 /** indicator of whether we are currently receiving data from the
222 * server. Basically this isn't more than a simple cache of the
223 * last PLAY/PAUSE command sent to the server, to make sure we don't
224 * send 2x the same unexpectedly or commands in the wrong state. */
225 enum RTSPClientState state;
226
227 /** the seek value requested when calling av_seek_frame(). This value
228 * is subsequently used as part of the "Range" parameter when emitting
229 * the RTSP PLAY command. If we are currently playing, this command is
230 * called instantly. If we are currently paused, this command is called
231 * whenever we resume playback. Either way, the value is only used once,
232 * see rtsp_read_play() and rtsp_read_seek(). */
233 int64_t seek_timestamp;
234
235 int seq; /**< RTSP command sequence number */
236
237 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
238 * identifier that the client should re-transmit in each RTSP command */
239 char session_id[512];
240
241 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
242 * the server will go without traffic on the RTSP/TCP line before it
243 * closes the connection. */
244 int timeout;
245
246 /** timestamp of the last RTSP command that we sent to the RTSP server.
247 * This is used to calculate when to send dummy commands to keep the
248 * connection alive, in conjunction with timeout. */
249 int64_t last_cmd_time;
250
251 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
252 enum RTSPTransport transport;
253
254 /** the negotiated network layer transport protocol; e.g. TCP or UDP
255 * uni-/multicast */
256 enum RTSPLowerTransport lower_transport;
257
258 /** brand of server that we're talking to; e.g. WMS, REAL or other.
259 * Detected based on the value of RTSPMessageHeader->server or the presence
260 * of RTSPMessageHeader->real_challenge */
261 enum RTSPServerType server_type;
262
263 /** the "RealChallenge1:" field from the server */
264 char real_challenge[64];
265
266 /** plaintext authorization line (username:password) */
267 char auth[128];
268
269 /** authentication state */
270 HTTPAuthState auth_state;
271
272 /** The last reply of the server to a RTSP command */
273 char last_reply[2048]; /* XXX: allocate ? */
274
275 /** RTSPStream->transport_priv of the last stream that we read a
276 * packet from */
277 void *cur_transport_priv;
278
279 /** The following are used for Real stream selection */
280 //@{
281 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
282 int need_subscription;
283
284 /** stream setup during the last frame read. This is used to detect if
285 * we need to subscribe or unsubscribe to any new streams. */
286 enum AVDiscard *real_setup_cache;
287
288 /** current stream setup. This is a temporary buffer used to compare
289 * current setup to previous frame setup. */
290 enum AVDiscard *real_setup;
291
292 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
293 * this is used to send the same "Unsubscribe:" if stream setup changed,
294 * before sending a new "Subscribe:" command. */
295 char last_subscription[1024];
296 //@}
297
298 /** The following are used for RTP/ASF streams */
299 //@{
300 /** ASF demuxer context for the embedded ASF stream from WMS servers */
301 AVFormatContext *asf_ctx;
302
303 /** cache for position of the asf demuxer, since we load a new
304 * data packet in the bytecontext for each incoming RTSP packet. */
305 uint64_t asf_pb_pos;
306 //@}
307
308 /** some MS RTSP streams contain a URL in the SDP that we need to use
309 * for all subsequent RTSP requests, rather than the input URI; in
310 * other cases, this is a copy of AVFormatContext->filename. */
311 char control_uri[1024];
312
313 /** Additional output handle, used when input and output are done
314 * separately, eg for HTTP tunneling. */
315 URLContext *rtsp_hd_out;
316
317 /** RTSP transport mode, such as plain or tunneled. */
318 enum RTSPControlTransport control_transport;
319
320 /* Number of RTCP BYE packets the RTSP session has received.
321 * An EOF is propagated back if nb_byes == nb_streams.
322 * This is reset after a seek. */
323 int nb_byes;
324
325 /** Reusable buffer for receiving packets */
326 uint8_t* recvbuf;
327
328 /**
329 * A mask with all requested transport methods
330 */
331 int lower_transport_mask;
332
333 /**
334 * The number of returned packets
335 */
336 uint64_t packets;
337
338 /**
339 * Polling array for udp
340 */
341 struct pollfd *p;
342
343 /**
344 * Whether the server supports the GET_PARAMETER method.
345 */
346 int get_parameter_supported;
347
348 /**
349 * Do not begin to play the stream immediately.
350 */
351 int initial_pause;
352
353 /**
354 * Option flags for the chained RTP muxer.
355 */
356 int rtp_muxer_flags;
357
358 /** Whether the server accepts the x-Dynamic-Rate header */
359 int accept_dynamic_rate;
360
361 /**
362 * Various option flags for the RTSP muxer/demuxer.
363 */
364 int rtsp_flags;
365
366 /**
367 * Mask of all requested media types
368 */
369 int media_type_mask;
370
371 /**
372 * Minimum and maximum local UDP ports.
373 */
374 int rtp_port_min, rtp_port_max;
375
376 /**
377 * Timeout to wait for incoming connections.
378 */
379 int initial_timeout;
380 } RTSPState;
381
382 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
383 receive packets only from the right
384 source address and port. */
385 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
386
387 /**
388 * Describe a single stream, as identified by a single m= line block in the
389 * SDP content. In the case of RDT, one RTSPStream can represent multiple
390 * AVStreams. In this case, each AVStream in this set has similar content
391 * (but different codec/bitrate).
392 */
393 typedef struct RTSPStream {
394 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
395 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
396
397 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
398 int stream_index;
399
400 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
401 * for the selected transport. Only used for TCP. */
402 int interleaved_min, interleaved_max;
403
404 char control_url[1024]; /**< url for this stream (from SDP) */
405
406 /** The following are used only in SDP, not RTSP */
407 //@{
408 int sdp_port; /**< port (from SDP content) */
409 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
410 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
411 int sdp_payload_type; /**< payload type */
412 //@}
413
414 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
415 //@{
416 /** handler structure */
417 RTPDynamicProtocolHandler *dynamic_handler;
418
419 /** private data associated with the dynamic protocol */
420 PayloadContext *dynamic_protocol_context;
421 //@}
422 } RTSPStream;
423
424 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
425 RTSPState *rt, const char *method);
426
427 /**
428 * Send a command to the RTSP server without waiting for the reply.
429 *
430 * @see rtsp_send_cmd_with_content_async
431 */
432 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
433 const char *url, const char *headers);
434
435 /**
436 * Send a command to the RTSP server and wait for the reply.
437 *
438 * @param s RTSP (de)muxer context
439 * @param method the method for the request
440 * @param url the target url for the request
441 * @param headers extra header lines to include in the request
442 * @param reply pointer where the RTSP message header will be stored
443 * @param content_ptr pointer where the RTSP message body, if any, will
444 * be stored (length is in reply)
445 * @param send_content if non-null, the data to send as request body content
446 * @param send_content_length the length of the send_content data, or 0 if
447 * send_content is null
448 *
449 * @return zero if success, nonzero otherwise
450 */
451 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
452 const char *method, const char *url,
453 const char *headers,
454 RTSPMessageHeader *reply,
455 unsigned char **content_ptr,
456 const unsigned char *send_content,
457 int send_content_length);
458
459 /**
460 * Send a command to the RTSP server and wait for the reply.
461 *
462 * @see rtsp_send_cmd_with_content
463 */
464 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
465 const char *url, const char *headers,
466 RTSPMessageHeader *reply, unsigned char **content_ptr);
467
468 /**
469 * Read a RTSP message from the server, or prepare to read data
470 * packets if we're reading data interleaved over the TCP/RTSP
471 * connection as well.
472 *
473 * @param s RTSP (de)muxer context
474 * @param reply pointer where the RTSP message header will be stored
475 * @param content_ptr pointer where the RTSP message body, if any, will
476 * be stored (length is in reply)
477 * @param return_on_interleaved_data whether the function may return if we
478 * encounter a data marker ('$'), which precedes data
479 * packets over interleaved TCP/RTSP connections. If this
480 * is set, this function will return 1 after encountering
481 * a '$'. If it is not set, the function will skip any
482 * data packets (if they are encountered), until a reply
483 * has been fully parsed. If no more data is available
484 * without parsing a reply, it will return an error.
485 * @param method the RTSP method this is a reply to. This affects how
486 * some response headers are acted upon. May be NULL.
487 *
488 * @return 1 if a data packets is ready to be received, -1 on error,
489 * and 0 on success.
490 */
491 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
492 unsigned char **content_ptr,
493 int return_on_interleaved_data, const char *method);
494
495 /**
496 * Skip a RTP/TCP interleaved packet.
497 */
498 void ff_rtsp_skip_packet(AVFormatContext *s);
499
500 /**
501 * Connect to the RTSP server and set up the individual media streams.
502 * This can be used for both muxers and demuxers.
503 *
504 * @param s RTSP (de)muxer context
505 *
506 * @return 0 on success, < 0 on error. Cleans up all allocations done
507 * within the function on error.
508 */
509 int ff_rtsp_connect(AVFormatContext *s);
510
511 /**
512 * Close and free all streams within the RTSP (de)muxer
513 *
514 * @param s RTSP (de)muxer context
515 */
516 void ff_rtsp_close_streams(AVFormatContext *s);
517
518 /**
519 * Close all connection handles within the RTSP (de)muxer
520 *
521 * @param s RTSP (de)muxer context
522 */
523 void ff_rtsp_close_connections(AVFormatContext *s);
524
525 /**
526 * Get the description of the stream and set up the RTSPStream child
527 * objects.
528 */
529 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
530
531 /**
532 * Announce the stream to the server and set up the RTSPStream child
533 * objects for each media stream.
534 */
535 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
536
537 /**
538 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
539 * listen mode.
540 */
541 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
542
543 /**
544 * Parse an SDP description of streams by populating an RTSPState struct
545 * within the AVFormatContext; also allocate the RTP streams and the
546 * pollfd array used for UDP streams.
547 */
548 int ff_sdp_parse(AVFormatContext *s, const char *content);
549
550 /**
551 * Receive one RTP packet from an TCP interleaved RTSP stream.
552 */
553 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
554 uint8_t *buf, int buf_size);
555
556 /**
557 * Receive one packet from the RTSPStreams set up in the AVFormatContext
558 * (which should contain a RTSPState struct as priv_data).
559 */
560 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
561
562 /**
563 * Do the SETUP requests for each stream for the chosen
564 * lower transport mode.
565 * @return 0 on success, <0 on error, 1 if protocol is unavailable
566 */
567 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
568 int lower_transport, const char *real_challenge);
569
570 /**
571 * Undo the effect of ff_rtsp_make_setup_request, close the
572 * transport_priv and rtp_handle fields.
573 */
574 void ff_rtsp_undo_setup(AVFormatContext *s);
575
576 /**
577 * Open RTSP transport context.
578 */
579 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
580
581 extern const AVOption ff_rtsp_options[];
582
583 #endif /* AVFORMAT_RTSP_H */