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