62bd3a2eccc9957ddd925e86115c3806e82f882c
[libav.git] / libavformat / rtsp.h
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 /**
32 * Network layer over which RTP/etc packet data will be transported.
33 */
34 enum RTSPLowerTransport {
35 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
36 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
37 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
38 RTSP_LOWER_TRANSPORT_NB
39 };
40
41 /**
42 * Packet profile of the data that we will be receiving. Real servers
43 * commonly send RDT (although they can sometimes send RTP as well),
44 * whereas most others will send RTP.
45 */
46 enum RTSPTransport {
47 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
48 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
49 RTSP_TRANSPORT_NB
50 };
51
52 /**
53 * Transport mode for the RTSP data. This may be plain, or
54 * tunneled, which is done over HTTP.
55 */
56 enum RTSPControlTransport {
57 RTSP_MODE_PLAIN, /**< Normal RTSP */
58 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
59 };
60
61 #define RTSP_DEFAULT_PORT 554
62 #define RTSP_MAX_TRANSPORTS 8
63 #define RTSP_TCP_MAX_PACKET_SIZE 1472
64 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
65 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
66 #define RTSP_RTP_PORT_MIN 5000
67 #define RTSP_RTP_PORT_MAX 10000
68
69 /**
70 * This describes a single item in the "Transport:" line of one stream as
71 * negotiated by the SETUP RTSP command. Multiple transports are comma-
72 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
73 * client_port=1000-1001;server_port=1800-1801") and described in separate
74 * RTSPTransportFields.
75 */
76 typedef struct RTSPTransportField {
77 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
78 * with a '$', stream length and stream ID. If the stream ID is within
79 * the range of this interleaved_min-max, then the packet belongs to
80 * this stream. */
81 int interleaved_min, interleaved_max;
82
83 /** UDP multicast port range; the ports to which we should connect to
84 * receive multicast UDP data. */
85 int port_min, port_max;
86
87 /** UDP client ports; these should be the local ports of the UDP RTP
88 * (and RTCP) sockets over which we receive RTP/RTCP data. */
89 int client_port_min, client_port_max;
90
91 /** UDP unicast server port range; the ports to which we should connect
92 * to receive unicast UDP RTP/RTCP data. */
93 int server_port_min, server_port_max;
94
95 /** time-to-live value (required for multicast); the amount of HOPs that
96 * packets will be allowed to make before being discarded. */
97 int ttl;
98
99 struct sockaddr_storage destination; /**< destination IP address */
100 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
101
102 /** data/packet transport protocol; e.g. RTP or RDT */
103 enum RTSPTransport transport;
104
105 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
106 enum RTSPLowerTransport lower_transport;
107 } RTSPTransportField;
108
109 /**
110 * This describes the server response to each RTSP command.
111 */
112 typedef struct RTSPMessageHeader {
113 /** length of the data following this header */
114 int content_length;
115
116 enum RTSPStatusCode status_code; /**< response code from server */
117
118 /** number of items in the 'transports' variable below */
119 int nb_transports;
120
121 /** Time range of the streams that the server will stream. In
122 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
123 int64_t range_start, range_end;
124
125 /** describes the complete "Transport:" line of the server in response
126 * to a SETUP RTSP command by the client */
127 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
128
129 int seq; /**< sequence number */
130
131 /** the "Session:" field. This value is initially set by the server and
132 * should be re-transmitted by the client in every RTSP command. */
133 char session_id[512];
134
135 /** the "Location:" field. This value is used to handle redirection.
136 */
137 char location[4096];
138
139 /** the "RealChallenge1:" field from the server */
140 char real_challenge[64];
141
142 /** the "Server: field, which can be used to identify some special-case
143 * servers that are not 100% standards-compliant. We use this to identify
144 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
145 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
146 * use something like "Helix [..] Server Version v.e.r.sion (platform)
147 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
148 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
149 char server[64];
150
151 /** The "timeout" comes as part of the server response to the "SETUP"
152 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
153 * time, in seconds, that the server will go without traffic over the
154 * RTSP/TCP connection before it closes the connection. To prevent
155 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
156 * than this value. */
157 int timeout;
158
159 /** The "Notice" or "X-Notice" field value. See
160 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
161 * for a complete list of supported values. */
162 int notice;
163
164 /** The "reason" is meant to specify better the meaning of the error code
165 * returned
166 */
167 char reason[256];
168 } RTSPMessageHeader;
169
170 /**
171 * Client state, i.e. whether we are currently receiving data (PLAYING) or
172 * setup-but-not-receiving (PAUSED). State can be changed in applications
173 * by calling av_read_play/pause().
174 */
175 enum RTSPClientState {
176 RTSP_STATE_IDLE, /**< not initialized */
177 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
178 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
179 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
180 };
181
182 /**
183 * Identifies particular servers that require special handling, such as
184 * standards-incompliant "Transport:" lines in the SETUP request.
185 */
186 enum RTSPServerType {
187 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
188 RTSP_SERVER_REAL, /**< Realmedia-style server */
189 RTSP_SERVER_WMS, /**< Windows Media server */
190 RTSP_SERVER_NB
191 };
192
193 /**
194 * Private data for the RTSP demuxer.
195 *
196 * @todo Use ByteIOContext instead of URLContext
197 */
198 typedef struct RTSPState {
199 URLContext *rtsp_hd; /* RTSP TCP connection handle */
200
201 /** number of items in the 'rtsp_streams' variable */
202 int nb_rtsp_streams;
203
204 struct RTSPStream **rtsp_streams; /**< streams in this session */
205
206 /** indicator of whether we are currently receiving data from the
207 * server. Basically this isn't more than a simple cache of the
208 * last PLAY/PAUSE command sent to the server, to make sure we don't
209 * send 2x the same unexpectedly or commands in the wrong state. */
210 enum RTSPClientState state;
211
212 /** the seek value requested when calling av_seek_frame(). This value
213 * is subsequently used as part of the "Range" parameter when emitting
214 * the RTSP PLAY command. If we are currently playing, this command is
215 * called instantly. If we are currently paused, this command is called
216 * whenever we resume playback. Either way, the value is only used once,
217 * see rtsp_read_play() and rtsp_read_seek(). */
218 int64_t seek_timestamp;
219
220 /* XXX: currently we use unbuffered input */
221 // ByteIOContext rtsp_gb;
222
223 int seq; /**< RTSP command sequence number */
224
225 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
226 * identifier that the client should re-transmit in each RTSP command */
227 char session_id[512];
228
229 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
230 * the server will go without traffic on the RTSP/TCP line before it
231 * closes the connection. */
232 int timeout;
233
234 /** timestamp of the last RTSP command that we sent to the RTSP server.
235 * This is used to calculate when to send dummy commands to keep the
236 * connection alive, in conjunction with timeout. */
237 int64_t last_cmd_time;
238
239 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
240 enum RTSPTransport transport;
241
242 /** the negotiated network layer transport protocol; e.g. TCP or UDP
243 * uni-/multicast */
244 enum RTSPLowerTransport lower_transport;
245
246 /** brand of server that we're talking to; e.g. WMS, REAL or other.
247 * Detected based on the value of RTSPMessageHeader->server or the presence
248 * of RTSPMessageHeader->real_challenge */
249 enum RTSPServerType server_type;
250
251 /** the "RealChallenge1:" field from the server */
252 char real_challenge[64];
253
254 /** plaintext authorization line (username:password) */
255 char auth[128];
256
257 /** authentication state */
258 HTTPAuthState auth_state;
259
260 /** The last reply of the server to a RTSP command */
261 char last_reply[2048]; /* XXX: allocate ? */
262
263 /** RTSPStream->transport_priv of the last stream that we read a
264 * packet from */
265 void *cur_transport_priv;
266
267 /** The following are used for Real stream selection */
268 //@{
269 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
270 int need_subscription;
271
272 /** stream setup during the last frame read. This is used to detect if
273 * we need to subscribe or unsubscribe to any new streams. */
274 enum AVDiscard *real_setup_cache;
275
276 /** current stream setup. This is a temporary buffer used to compare
277 * current setup to previous frame setup. */
278 enum AVDiscard *real_setup;
279
280 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
281 * this is used to send the same "Unsubscribe:" if stream setup changed,
282 * before sending a new "Subscribe:" command. */
283 char last_subscription[1024];
284 //@}
285
286 /** The following are used for RTP/ASF streams */
287 //@{
288 /** ASF demuxer context for the embedded ASF stream from WMS servers */
289 AVFormatContext *asf_ctx;
290
291 /** cache for position of the asf demuxer, since we load a new
292 * data packet in the bytecontext for each incoming RTSP packet. */
293 uint64_t asf_pb_pos;
294 //@}
295
296 /** some MS RTSP streams contain a URL in the SDP that we need to use
297 * for all subsequent RTSP requests, rather than the input URI; in
298 * other cases, this is a copy of AVFormatContext->filename. */
299 char control_uri[1024];
300
301 /** Additional output handle, used when input and output are done
302 * separately, eg for HTTP tunneling. */
303 URLContext *rtsp_hd_out;
304
305 /** RTSP transport mode, such as plain or tunneled. */
306 enum RTSPControlTransport control_transport;
307
308 /* Number of RTCP BYE packets the RTSP session has received.
309 * An EOF is propagated back if nb_byes == nb_streams.
310 * This is reset after a seek. */
311 int nb_byes;
312
313 /** Reusable buffer for receiving packets */
314 uint8_t* recvbuf;
315
316 /** Filter incoming UDP packets - receive packets only from the right
317 * source address and port. */
318 int filter_source;
319
320 /**
321 * A mask with all requested transport methods
322 */
323 int lower_transport_mask;
324
325 /**
326 * The number of returned packets
327 */
328 uint64_t packets;
329 } RTSPState;
330
331 /**
332 * Describes a single stream, as identified by a single m= line block in the
333 * SDP content. In the case of RDT, one RTSPStream can represent multiple
334 * AVStreams. In this case, each AVStream in this set has similar content
335 * (but different codec/bitrate).
336 */
337 typedef struct RTSPStream {
338 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
339 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
340
341 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
342 int stream_index;
343
344 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
345 * for the selected transport. Only used for TCP. */
346 int interleaved_min, interleaved_max;
347
348 char control_url[1024]; /**< url for this stream (from SDP) */
349
350 /** The following are used only in SDP, not RTSP */
351 //@{
352 int sdp_port; /**< port (from SDP content) */
353 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
354 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
355 int sdp_payload_type; /**< payload type */
356 //@}
357
358 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
359 //@{
360 /** handler structure */
361 RTPDynamicProtocolHandler *dynamic_handler;
362
363 /** private data associated with the dynamic protocol */
364 PayloadContext *dynamic_protocol_context;
365 //@}
366 } RTSPStream;
367
368 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
369 RTSPState *rt, const char *method);
370
371 extern int rtsp_rtp_port_min;
372 extern int rtsp_rtp_port_max;
373
374 /**
375 * Send a command to the RTSP server without waiting for the reply.
376 *
377 * @param s RTSP (de)muxer context
378 * @param method the method for the request
379 * @param url the target url for the request
380 * @param headers extra header lines to include in the request
381 * @param send_content if non-null, the data to send as request body content
382 * @param send_content_length the length of the send_content data, or 0 if
383 * send_content is null
384 *
385 * @return zero if success, nonzero otherwise
386 */
387 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
388 const char *method, const char *url,
389 const char *headers,
390 const unsigned char *send_content,
391 int send_content_length);
392 /**
393 * Send a command to the RTSP server without waiting for the reply.
394 *
395 * @see rtsp_send_cmd_with_content_async
396 */
397 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
398 const char *url, const char *headers);
399
400 /**
401 * Send a command to the RTSP server and wait for the reply.
402 *
403 * @param s RTSP (de)muxer context
404 * @param method the method for the request
405 * @param url the target url for the request
406 * @param headers extra header lines to include in the request
407 * @param reply pointer where the RTSP message header will be stored
408 * @param content_ptr pointer where the RTSP message body, if any, will
409 * be stored (length is in reply)
410 * @param send_content if non-null, the data to send as request body content
411 * @param send_content_length the length of the send_content data, or 0 if
412 * send_content is null
413 *
414 * @return zero if success, nonzero otherwise
415 */
416 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
417 const char *method, const char *url,
418 const char *headers,
419 RTSPMessageHeader *reply,
420 unsigned char **content_ptr,
421 const unsigned char *send_content,
422 int send_content_length);
423
424 /**
425 * Send a command to the RTSP server and wait for the reply.
426 *
427 * @see rtsp_send_cmd_with_content
428 */
429 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
430 const char *url, const char *headers,
431 RTSPMessageHeader *reply, unsigned char **content_ptr);
432
433 /**
434 * Read a RTSP message from the server, or prepare to read data
435 * packets if we're reading data interleaved over the TCP/RTSP
436 * connection as well.
437 *
438 * @param s RTSP (de)muxer context
439 * @param reply pointer where the RTSP message header will be stored
440 * @param content_ptr pointer where the RTSP message body, if any, will
441 * be stored (length is in reply)
442 * @param return_on_interleaved_data whether the function may return if we
443 * encounter a data marker ('$'), which precedes data
444 * packets over interleaved TCP/RTSP connections. If this
445 * is set, this function will return 1 after encountering
446 * a '$'. If it is not set, the function will skip any
447 * data packets (if they are encountered), until a reply
448 * has been fully parsed. If no more data is available
449 * without parsing a reply, it will return an error.
450 * @param method the RTSP method this is a reply to. This affects how
451 * some response headers are acted upon. May be NULL.
452 *
453 * @return 1 if a data packets is ready to be received, -1 on error,
454 * and 0 on success.
455 */
456 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
457 unsigned char **content_ptr,
458 int return_on_interleaved_data, const char *method);
459
460 /**
461 * Skip a RTP/TCP interleaved packet.
462 */
463 void ff_rtsp_skip_packet(AVFormatContext *s);
464
465 /**
466 * Connect to the RTSP server and set up the individual media streams.
467 * This can be used for both muxers and demuxers.
468 *
469 * @param s RTSP (de)muxer context
470 *
471 * @return 0 on success, < 0 on error. Cleans up all allocations done
472 * within the function on error.
473 */
474 int ff_rtsp_connect(AVFormatContext *s);
475
476 /**
477 * Close and free all streams within the RTSP (de)muxer
478 *
479 * @param s RTSP (de)muxer context
480 */
481 void ff_rtsp_close_streams(AVFormatContext *s);
482
483 /**
484 * Close all connection handles within the RTSP (de)muxer
485 *
486 * @param rt RTSP (de)muxer context
487 */
488 void ff_rtsp_close_connections(AVFormatContext *rt);
489
490 /**
491 * Get the description of the stream and set up the RTSPStream child
492 * objects.
493 */
494 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
495
496 /**
497 * Announce the stream to the server and set up the RTSPStream child
498 * objects for each media stream.
499 */
500 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
501
502 /**
503 * Parse a SDP description of streams by populating an RTSPState struct
504 * within the AVFormatContext.
505 */
506 int ff_sdp_parse(AVFormatContext *s, const char *content);
507
508 /**
509 * Receive one RTP packet from an TCP interleaved RTSP stream.
510 */
511 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
512 uint8_t *buf, int buf_size);
513
514 /**
515 * Receive one packet from the RTSPStreams set up in the AVFormatContext
516 * (which should contain a RTSPState struct as priv_data).
517 */
518 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
519
520 /**
521 * Do the SETUP requests for each stream for the chosen
522 * lower transport mode.
523 */
524 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
525 int lower_transport, const char *real_challenge);
526
527 /**
528 * Undo the effect of ff_rtsp_make_setup_request, close the
529 * transport_priv and rtp_handle fields.
530 */
531 void ff_rtsp_undo_setup(AVFormatContext *s);
532
533 #endif /* AVFORMAT_RTSP_H */