Statistics
| Branch: | Revision:

ffmpeg / libavformat / rtsp.h @ abbc1d27

History | View | Annotate | Download (19 KB)

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
    /** plaintext authorization line (username:password) */
252
    char auth[128];
253

    
254
    /** authentication state */
255
    HTTPAuthState auth_state;
256

    
257
    /** The last reply of the server to a RTSP command */
258
    char last_reply[2048]; /* XXX: allocate ? */
259

    
260
    /** RTSPStream->transport_priv of the last stream that we read a
261
     * packet from */
262
    void *cur_transport_priv;
263

    
264
    /** The following are used for Real stream selection */
265
    //@{
266
    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
267
    int need_subscription;
268

    
269
    /** stream setup during the last frame read. This is used to detect if
270
     * we need to subscribe or unsubscribe to any new streams. */
271
    enum AVDiscard *real_setup_cache;
272

    
273
    /** current stream setup. This is a temporary buffer used to compare
274
     * current setup to previous frame setup. */
275
    enum AVDiscard *real_setup;
276

    
277
    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
278
     * this is used to send the same "Unsubscribe:" if stream setup changed,
279
     * before sending a new "Subscribe:" command. */
280
    char last_subscription[1024];
281
    //@}
282

    
283
    /** The following are used for RTP/ASF streams */
284
    //@{
285
    /** ASF demuxer context for the embedded ASF stream from WMS servers */
286
    AVFormatContext *asf_ctx;
287

    
288
    /** cache for position of the asf demuxer, since we load a new
289
     * data packet in the bytecontext for each incoming RTSP packet. */
290
    uint64_t asf_pb_pos;
291
    //@}
292

    
293
    /** some MS RTSP streams contain a URL in the SDP that we need to use
294
     * for all subsequent RTSP requests, rather than the input URI; in
295
     * other cases, this is a copy of AVFormatContext->filename. */
296
    char control_uri[1024];
297

    
298
    /** Additional output handle, used when input and output are done
299
     * separately, eg for HTTP tunneling. */
300
    URLContext *rtsp_hd_out;
301

    
302
    /** RTSP transport mode, such as plain or tunneled. */
303
    enum RTSPControlTransport control_transport;
304

    
305
    /* Number of RTCP BYE packets the RTSP session has received.
306
     * An EOF is propagated back if nb_byes == nb_streams.
307
     * This is reset after a seek. */
308
    int nb_byes;
309

    
310
    /** Reusable buffer for receiving packets */
311
    uint8_t* recvbuf;
312

    
313
    /** Filter incoming UDP packets - receive packets only from the right
314
     * source address and port. */
315
    int filter_source;
316
} RTSPState;
317

    
318
/**
319
 * Describes a single stream, as identified by a single m= line block in the
320
 * SDP content. In the case of RDT, one RTSPStream can represent multiple
321
 * AVStreams. In this case, each AVStream in this set has similar content
322
 * (but different codec/bitrate).
323
 */
324
typedef struct RTSPStream {
325
    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
326
    void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
327

    
328
    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
329
    int stream_index;
330

    
331
    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
332
     * for the selected transport. Only used for TCP. */
333
    int interleaved_min, interleaved_max;
334

    
335
    char control_url[1024];   /**< url for this stream (from SDP) */
336

    
337
    /** The following are used only in SDP, not RTSP */
338
    //@{
339
    int sdp_port;             /**< port (from SDP content) */
340
    struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
341
    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
342
    int sdp_payload_type;     /**< payload type */
343
    //@}
344

    
345
    /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
346
    //@{
347
    /** handler structure */
348
    RTPDynamicProtocolHandler *dynamic_handler;
349

    
350
    /** private data associated with the dynamic protocol */
351
    PayloadContext *dynamic_protocol_context;
352
    //@}
353
} RTSPStream;
354

    
355
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
356
                        RTSPState *rt, const char *method);
357

    
358
extern int rtsp_rtp_port_min;
359
extern int rtsp_rtp_port_max;
360

    
361
/**
362
 * Send a command to the RTSP server without waiting for the reply.
363
 *
364
 * @param s RTSP (de)muxer context
365
 * @param method the method for the request
366
 * @param url the target url for the request
367
 * @param headers extra header lines to include in the request
368
 * @param send_content if non-null, the data to send as request body content
369
 * @param send_content_length the length of the send_content data, or 0 if
370
 *                            send_content is null
371
 *
372
 * @return zero if success, nonzero otherwise
373
 */
374
int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
375
                                        const char *method, const char *url,
376
                                        const char *headers,
377
                                        const unsigned char *send_content,
378
                                        int send_content_length);
379
/**
380
 * Send a command to the RTSP server without waiting for the reply.
381
 *
382
 * @see rtsp_send_cmd_with_content_async
383
 */
384
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
385
                           const char *url, const char *headers);
386

    
387
/**
388
 * Send a command to the RTSP server and wait for the reply.
389
 *
390
 * @param s RTSP (de)muxer context
391
 * @param method the method for the request
392
 * @param url the target url for the request
393
 * @param headers extra header lines to include in the request
394
 * @param reply pointer where the RTSP message header will be stored
395
 * @param content_ptr pointer where the RTSP message body, if any, will
396
 *                    be stored (length is in reply)
397
 * @param send_content if non-null, the data to send as request body content
398
 * @param send_content_length the length of the send_content data, or 0 if
399
 *                            send_content is null
400
 *
401
 * @return zero if success, nonzero otherwise
402
 */
403
int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
404
                                  const char *method, const char *url,
405
                                  const char *headers,
406
                                  RTSPMessageHeader *reply,
407
                                  unsigned char **content_ptr,
408
                                  const unsigned char *send_content,
409
                                  int send_content_length);
410

    
411
/**
412
 * Send a command to the RTSP server and wait for the reply.
413
 *
414
 * @see rtsp_send_cmd_with_content
415
 */
416
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
417
                     const char *url, const char *headers,
418
                     RTSPMessageHeader *reply, unsigned char **content_ptr);
419

    
420
/**
421
 * Read a RTSP message from the server, or prepare to read data
422
 * packets if we're reading data interleaved over the TCP/RTSP
423
 * connection as well.
424
 *
425
 * @param s RTSP (de)muxer context
426
 * @param reply pointer where the RTSP message header will be stored
427
 * @param content_ptr pointer where the RTSP message body, if any, will
428
 *                    be stored (length is in reply)
429
 * @param return_on_interleaved_data whether the function may return if we
430
 *                   encounter a data marker ('$'), which precedes data
431
 *                   packets over interleaved TCP/RTSP connections. If this
432
 *                   is set, this function will return 1 after encountering
433
 *                   a '$'. If it is not set, the function will skip any
434
 *                   data packets (if they are encountered), until a reply
435
 *                   has been fully parsed. If no more data is available
436
 *                   without parsing a reply, it will return an error.
437
 * @param method the RTSP method this is a reply to. This affects how
438
 *               some response headers are acted upon. May be NULL.
439
 *
440
 * @return 1 if a data packets is ready to be received, -1 on error,
441
 *          and 0 on success.
442
 */
443
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
444
                       unsigned char **content_ptr,
445
                       int return_on_interleaved_data, const char *method);
446

    
447
/**
448
 * Skip a RTP/TCP interleaved packet.
449
 */
450
void ff_rtsp_skip_packet(AVFormatContext *s);
451

    
452
/**
453
 * Connect to the RTSP server and set up the individual media streams.
454
 * This can be used for both muxers and demuxers.
455
 *
456
 * @param s RTSP (de)muxer context
457
 *
458
 * @return 0 on success, < 0 on error. Cleans up all allocations done
459
 *          within the function on error.
460
 */
461
int ff_rtsp_connect(AVFormatContext *s);
462

    
463
/**
464
 * Close and free all streams within the RTSP (de)muxer
465
 *
466
 * @param s RTSP (de)muxer context
467
 */
468
void ff_rtsp_close_streams(AVFormatContext *s);
469

    
470
/**
471
 * Close all connection handles within the RTSP (de)muxer
472
 *
473
 * @param rt RTSP (de)muxer context
474
 */
475
void ff_rtsp_close_connections(AVFormatContext *rt);
476

    
477
/**
478
 * Get the description of the stream and set up the RTSPStream child
479
 * objects.
480
 */
481
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
482

    
483
/**
484
 * Announce the stream to the server and set up the RTSPStream child
485
 * objects for each media stream.
486
 */
487
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
488

    
489
/**
490
 * Parse a SDP description of streams by populating an RTSPState struct
491
 * within the AVFormatContext.
492
 */
493
int ff_sdp_parse(AVFormatContext *s, const char *content);
494

    
495
/**
496
 * Receive one RTP packet from an TCP interleaved RTSP stream.
497
 */
498
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
499
                            uint8_t *buf, int buf_size);
500

    
501
/**
502
 * Receive one packet from the RTSPStreams set up in the AVFormatContext
503
 * (which should contain a RTSPState struct as priv_data).
504
 */
505
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
506

    
507
/**
508
 * Do the SETUP requests for each stream for the chosen
509
 * lower transport mode.
510
 */
511
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
512
                               int lower_transport, const char *real_challenge);
513

    
514
/**
515
 * Undo the effect of ff_rtsp_make_setup_request, close the
516
 * transport_priv and rtp_handle fields.
517
 */
518
void ff_rtsp_undo_setup(AVFormatContext *s);
519

    
520
#endif /* AVFORMAT_RTSP_H */