RTSP: Add a second URLContext for outgoing messages
[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 #define RTSP_DEFAULT_PORT 554
53 #define RTSP_MAX_TRANSPORTS 8
54 #define RTSP_TCP_MAX_PACKET_SIZE 1472
55 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
56 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
57 #define RTSP_RTP_PORT_MIN 5000
58 #define RTSP_RTP_PORT_MAX 10000
59
60 /**
61 * This describes a single item in the "Transport:" line of one stream as
62 * negotiated by the SETUP RTSP command. Multiple transports are comma-
63 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
64 * client_port=1000-1001;server_port=1800-1801") and described in separate
65 * RTSPTransportFields.
66 */
67 typedef struct RTSPTransportField {
68 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
69 * with a '$', stream length and stream ID. If the stream ID is within
70 * the range of this interleaved_min-max, then the packet belongs to
71 * this stream. */
72 int interleaved_min, interleaved_max;
73
74 /** UDP multicast port range; the ports to which we should connect to
75 * receive multicast UDP data. */
76 int port_min, port_max;
77
78 /** UDP client ports; these should be the local ports of the UDP RTP
79 * (and RTCP) sockets over which we receive RTP/RTCP data. */
80 int client_port_min, client_port_max;
81
82 /** UDP unicast server port range; the ports to which we should connect
83 * to receive unicast UDP RTP/RTCP data. */
84 int server_port_min, server_port_max;
85
86 /** time-to-live value (required for multicast); the amount of HOPs that
87 * packets will be allowed to make before being discarded. */
88 int ttl;
89
90 uint32_t destination; /**< destination IP address */
91
92 /** data/packet transport protocol; e.g. RTP or RDT */
93 enum RTSPTransport transport;
94
95 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
96 enum RTSPLowerTransport lower_transport;
97 } RTSPTransportField;
98
99 /**
100 * This describes the server response to each RTSP command.
101 */
102 typedef struct RTSPMessageHeader {
103 /** length of the data following this header */
104 int content_length;
105
106 enum RTSPStatusCode status_code; /**< response code from server */
107
108 /** number of items in the 'transports' variable below */
109 int nb_transports;
110
111 /** Time range of the streams that the server will stream. In
112 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
113 int64_t range_start, range_end;
114
115 /** describes the complete "Transport:" line of the server in response
116 * to a SETUP RTSP command by the client */
117 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
118
119 int seq; /**< sequence number */
120
121 /** the "Session:" field. This value is initially set by the server and
122 * should be re-transmitted by the client in every RTSP command. */
123 char session_id[512];
124
125 /** the "Location:" field. This value is used to handle redirection.
126 */
127 char location[4096];
128
129 /** the "RealChallenge1:" field from the server */
130 char real_challenge[64];
131
132 /** the "Server: field, which can be used to identify some special-case
133 * servers that are not 100% standards-compliant. We use this to identify
134 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
135 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
136 * use something like "Helix [..] Server Version v.e.r.sion (platform)
137 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
138 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
139 char server[64];
140
141 /** The "timeout" comes as part of the server response to the "SETUP"
142 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
143 * time, in seconds, that the server will go without traffic over the
144 * RTSP/TCP connection before it closes the connection. To prevent
145 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
146 * than this value. */
147 int timeout;
148
149 /** The "Notice" or "X-Notice" field value. See
150 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
151 * for a complete list of supported values. */
152 int notice;
153 } RTSPMessageHeader;
154
155 /**
156 * Client state, i.e. whether we are currently receiving data (PLAYING) or
157 * setup-but-not-receiving (PAUSED). State can be changed in applications
158 * by calling av_read_play/pause().
159 */
160 enum RTSPClientState {
161 RTSP_STATE_IDLE, /**< not initialized */
162 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
163 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
164 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
165 };
166
167 /**
168 * Identifies particular servers that require special handling, such as
169 * standards-incompliant "Transport:" lines in the SETUP request.
170 */
171 enum RTSPServerType {
172 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
173 RTSP_SERVER_REAL, /**< Realmedia-style server */
174 RTSP_SERVER_WMS, /**< Windows Media server */
175 RTSP_SERVER_NB
176 };
177
178 /**
179 * Private data for the RTSP demuxer.
180 *
181 * @todo Use ByteIOContext instead of URLContext
182 */
183 typedef struct RTSPState {
184 URLContext *rtsp_hd; /* RTSP TCP connexion handle */
185
186 /** number of items in the 'rtsp_streams' variable */
187 int nb_rtsp_streams;
188
189 struct RTSPStream **rtsp_streams; /**< streams in this session */
190
191 /** indicator of whether we are currently receiving data from the
192 * server. Basically this isn't more than a simple cache of the
193 * last PLAY/PAUSE command sent to the server, to make sure we don't
194 * send 2x the same unexpectedly or commands in the wrong state. */
195 enum RTSPClientState state;
196
197 /** the seek value requested when calling av_seek_frame(). This value
198 * is subsequently used as part of the "Range" parameter when emitting
199 * the RTSP PLAY command. If we are currently playing, this command is
200 * called instantly. If we are currently paused, this command is called
201 * whenever we resume playback. Either way, the value is only used once,
202 * see rtsp_read_play() and rtsp_read_seek(). */
203 int64_t seek_timestamp;
204
205 /* XXX: currently we use unbuffered input */
206 // ByteIOContext rtsp_gb;
207
208 int seq; /**< RTSP command sequence number */
209
210 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
211 * identifier that the client should re-transmit in each RTSP command */
212 char session_id[512];
213
214 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
215 * the server will go without traffic on the RTSP/TCP line before it
216 * closes the connection. */
217 int timeout;
218
219 /** timestamp of the last RTSP command that we sent to the RTSP server.
220 * This is used to calculate when to send dummy commands to keep the
221 * connection alive, in conjunction with timeout. */
222 int64_t last_cmd_time;
223
224 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
225 enum RTSPTransport transport;
226
227 /** the negotiated network layer transport protocol; e.g. TCP or UDP
228 * uni-/multicast */
229 enum RTSPLowerTransport lower_transport;
230
231 /** brand of server that we're talking to; e.g. WMS, REAL or other.
232 * Detected based on the value of RTSPMessageHeader->server or the presence
233 * of RTSPMessageHeader->real_challenge */
234 enum RTSPServerType server_type;
235
236 /** plaintext authorization line (username:password) */
237 char auth[128];
238
239 /** authentication state */
240 HTTPAuthState auth_state;
241
242 /** The last reply of the server to a RTSP command */
243 char last_reply[2048]; /* XXX: allocate ? */
244
245 /** RTSPStream->transport_priv of the last stream that we read a
246 * packet from */
247 void *cur_transport_priv;
248
249 /** The following are used for Real stream selection */
250 //@{
251 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
252 int need_subscription;
253
254 /** stream setup during the last frame read. This is used to detect if
255 * we need to subscribe or unsubscribe to any new streams. */
256 enum AVDiscard real_setup_cache[MAX_STREAMS];
257
258 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
259 * this is used to send the same "Unsubscribe:" if stream setup changed,
260 * before sending a new "Subscribe:" command. */
261 char last_subscription[1024];
262 //@}
263
264 /** The following are used for RTP/ASF streams */
265 //@{
266 /** ASF demuxer context for the embedded ASF stream from WMS servers */
267 AVFormatContext *asf_ctx;
268
269 /** cache for position of the asf demuxer, since we load a new
270 * data packet in the bytecontext for each incoming RTSP packet. */
271 uint64_t asf_pb_pos;
272 //@}
273
274 /** some MS RTSP streams contain a URL in the SDP that we need to use
275 * for all subsequent RTSP requests, rather than the input URI; in
276 * other cases, this is a copy of AVFormatContext->filename. */
277 char control_uri[1024];
278
279 /** The synchronized start time of the output streams. */
280 int64_t start_time;
281
282 /** Additional output handle, used when input and output are done
283 * separately, eg for HTTP tunneling. */
284 URLContext *rtsp_hd_out;
285 } RTSPState;
286
287 /**
288 * Describes a single stream, as identified by a single m= line block in the
289 * SDP content. In the case of RDT, one RTSPStream can represent multiple
290 * AVStreams. In this case, each AVStream in this set has similar content
291 * (but different codec/bitrate).
292 */
293 typedef struct RTSPStream {
294 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
295 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
296
297 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
298 int stream_index;
299
300 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
301 * for the selected transport. Only used for TCP. */
302 int interleaved_min, interleaved_max;
303
304 char control_url[1024]; /**< url for this stream (from SDP) */
305
306 /** The following are used only in SDP, not RTSP */
307 //@{
308 int sdp_port; /**< port (from SDP content) */
309 struct in_addr sdp_ip; /**< IP address (from SDP content) */
310 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
311 int sdp_payload_type; /**< payload type */
312 //@}
313
314 /** rtp payload parsing infos from SDP (i.e. mapping between private
315 * payload IDs and media-types (string), so that we can derive what
316 * type of payload we're dealing with (and how to parse it). */
317 RTPPayloadData rtp_payload_data;
318
319 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
320 //@{
321 /** handler structure */
322 RTPDynamicProtocolHandler *dynamic_handler;
323
324 /** private data associated with the dynamic protocol */
325 PayloadContext *dynamic_protocol_context;
326 //@}
327 } RTSPStream;
328
329 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
330 HTTPAuthState *auth_state);
331
332 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
333 extern int rtsp_default_protocols;
334 #endif
335 extern int rtsp_rtp_port_min;
336 extern int rtsp_rtp_port_max;
337
338 /**
339 * Send a command to the RTSP server without waiting for the reply.
340 *
341 * @param s RTSP (de)muxer context
342 * @param method the method for the request
343 * @param url the target url for the request
344 * @param headers extra header lines to include in the request
345 * @param send_content if non-null, the data to send as request body content
346 * @param send_content_length the length of the send_content data, or 0 if
347 * send_content is null
348 */
349 void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
350 const char *method, const char *url,
351 const char *headers,
352 const unsigned char *send_content,
353 int send_content_length);
354 /**
355 * Send a command to the RTSP server without waiting for the reply.
356 *
357 * @see rtsp_send_cmd_with_content_async
358 */
359 void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
360 const char *url, const char *headers);
361
362 /**
363 * Send a command to the RTSP server and wait for the reply.
364 *
365 * @param s RTSP (de)muxer context
366 * @param method the method for the request
367 * @param url the target url for the request
368 * @param headers extra header lines to include in the request
369 * @param reply pointer where the RTSP message header will be stored
370 * @param content_ptr pointer where the RTSP message body, if any, will
371 * be stored (length is in reply)
372 * @param send_content if non-null, the data to send as request body content
373 * @param send_content_length the length of the send_content data, or 0 if
374 * send_content is null
375 */
376 void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
377 const char *method, const char *url,
378 const char *headers,
379 RTSPMessageHeader *reply,
380 unsigned char **content_ptr,
381 const unsigned char *send_content,
382 int send_content_length);
383
384 /**
385 * Send a command to the RTSP server and wait for the reply.
386 *
387 * @see rtsp_send_cmd_with_content
388 */
389 void ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
390 const char *url, const char *headers,
391 RTSPMessageHeader *reply, unsigned char **content_ptr);
392
393 /**
394 * Read a RTSP message from the server, or prepare to read data
395 * packets if we're reading data interleaved over the TCP/RTSP
396 * connection as well.
397 *
398 * @param s RTSP (de)muxer context
399 * @param reply pointer where the RTSP message header will be stored
400 * @param content_ptr pointer where the RTSP message body, if any, will
401 * be stored (length is in reply)
402 * @param return_on_interleaved_data whether the function may return if we
403 * encounter a data marker ('$'), which precedes data
404 * packets over interleaved TCP/RTSP connections. If this
405 * is set, this function will return 1 after encountering
406 * a '$'. If it is not set, the function will skip any
407 * data packets (if they are encountered), until a reply
408 * has been fully parsed. If no more data is available
409 * without parsing a reply, it will return an error.
410 *
411 * @return 1 if a data packets is ready to be received, -1 on error,
412 * and 0 on success.
413 */
414 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
415 unsigned char **content_ptr,
416 int return_on_interleaved_data);
417
418 /**
419 * Skip a RTP/TCP interleaved packet.
420 */
421 void ff_rtsp_skip_packet(AVFormatContext *s);
422
423 /**
424 * Connect to the RTSP server and set up the individual media streams.
425 * This can be used for both muxers and demuxers.
426 *
427 * @param s RTSP (de)muxer context
428 *
429 * @return 0 on success, < 0 on error. Cleans up all allocations done
430 * within the function on error.
431 */
432 int ff_rtsp_connect(AVFormatContext *s);
433
434 /**
435 * Close and free all streams within the RTSP (de)muxer
436 *
437 * @param s RTSP (de)muxer context
438 */
439 void ff_rtsp_close_streams(AVFormatContext *s);
440
441 /**
442 * Close all connection handles within the RTSP (de)muxer
443 *
444 * @param rt RTSP (de)muxer context
445 */
446 void ff_rtsp_close_connections(AVFormatContext *rt);
447
448 #endif /* AVFORMAT_RTSP_H */