Statistics
| Branch: | Revision:

ffmpeg / libavformat / rtsp.h @ bf7e799c

History | View | Annotate | Download (11.3 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 FFMPEG_RTSP_H
22
#define FFMPEG_RTSP_H
23

    
24
#include <stdint.h>
25
#include "avformat.h"
26
#include "rtspcodes.h"
27
#include "rtpdec.h"
28
#include "network.h"
29

    
30
/**
31
 * Network layer over which RTP/etc packet data will be transported.
32
 */
33
enum RTSPLowerTransport {
34
    RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
35
    RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
36
    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
37
    RTSP_LOWER_TRANSPORT_NB
38
};
39

    
40
/**
41
 * Packet profile of the data that we will be receiving. Real servers
42
 * commonly send RDT (although they can sometimes send RTP as well),
43
 * whereas most others will send RTP.
44
 */
45
enum RTSPTransport {
46
    RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
47
    RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
48
    RTSP_TRANSPORT_NB
49
};
50

    
51
#define RTSP_DEFAULT_PORT   554
52
#define RTSP_MAX_TRANSPORTS 8
53
#define RTSP_TCP_MAX_PACKET_SIZE 1472
54
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
55
#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
56
#define RTSP_RTP_PORT_MIN 5000
57
#define RTSP_RTP_PORT_MAX 10000
58

    
59
/**
60
 * This describes a single item in the "Transport:" line of one stream as
61
 * negotiated by the SETUP RTSP command. Multiple transports are comma-
62
 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
63
 * client_port=1000-1001;server_port=1800-1801") and described in separate
64
 * RTSPTransportFields.
65
 */
66
typedef struct RTSPTransportField {
67
    /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
68
     * with a '$', stream length and stream ID. If the stream ID is within
69
     * the range of this interleaved_min-max, then the packet belongs to
70
     * this stream. */
71
    int interleaved_min, interleaved_max;
72

    
73
    /** UDP multicast port range; the ports to which we should connect to
74
     * receive multicast UDP data. */
75
    int port_min, port_max;
76

    
77
    /** UDP client ports; these should be the local ports of the UDP RTP
78
     * (and RTCP) sockets over which we receive RTP/RTCP data. */
79
    int client_port_min, client_port_max;
80

    
81
    /** UDP unicast server port range; the ports to which we should connect
82
     * to receive unicast UDP RTP/RTCP data. */
83
    int server_port_min, server_port_max;
84

    
85
    /** time-to-live value (required for multicast); the amount of HOPs that
86
     * packets will be allowed to make before being discarded. */
87
    int ttl;
88

    
89
    uint32_t destination; /**< destination IP address */
90

    
91
    /** data/packet transport protocol; e.g. RTP or RDT */
92
    enum RTSPTransport transport;
93

    
94
    /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
95
    enum RTSPLowerTransport lower_transport;
96
} RTSPTransportField;
97

    
98
/**
99
 * This describes the server response to each RTSP command.
100
 */
101
typedef struct RTSPMessageHeader {
102
    /** length of the data following this header */
103
    int content_length;
104

    
105
    enum RTSPStatusCode status_code; /**< response code from server */
106

    
107
    /** number of items in the 'transports' variable below */
108
    int nb_transports;
109

    
110
    /** Time range of the streams that the server will stream. In
111
     * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
112
    int64_t range_start, range_end;
113

    
114
    /** describes the complete "Transport:" line of the server in response
115
     * to a SETUP RTSP command by the client */
116
    RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
117

    
118
    int seq;                         /**< sequence number */
119

    
120
    /** the "Session:" field. This value is initially set by the server and
121
     * should be re-transmitted by the client in every RTSP command. */
122
    char session_id[512];
123

    
124
    /** the "RealChallenge1:" field from the server */
125
    char real_challenge[64];
126

    
127
    /** the "Server: field, which can be used to identify some special-case
128
     * servers that are not 100% standards-compliant. We use this to identify
129
     * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
130
     * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
131
     * use something like "Helix [..] Server Version v.e.r.sion (platform)
132
     * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
133
     * where platform is the output of $uname -msr | sed 's/ /-/g'. */
134
    char server[64];
135

    
136
    /** The "timeout" comes as part of the server response to the "SETUP"
137
     * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
138
     * time, in seconds, that the server will go without traffic over the
139
     * RTSP/TCP connection before it closes the connection. To prevent
140
     * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
141
     * than this value. */
142
    int timeout;
143
} RTSPMessageHeader;
144

    
145
/**
146
 * Client state, i.e. whether we are currently receiving data (PLAYING) or
147
 * setup-but-not-receiving (PAUSED). State can be changed in applications
148
 * by calling av_read_play/pause().
149
 */
150
enum RTSPClientState {
151
    RTSP_STATE_IDLE,    /**< not initialized */
152
    RTSP_STATE_PLAYING, /**< initialized and receiving data */
153
    RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
154
};
155

    
156
/**
157
 * Identifies particular servers that require special handling, such as
158
 * standards-incompliant "Transport:" lines in the SETUP request.
159
 */
160
enum RTSPServerType {
161
    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
162
    RTSP_SERVER_REAL, /**< Realmedia-style server */
163
    RTSP_SERVER_WMS,  /**< Windows Media server */
164
    RTSP_SERVER_NB
165
};
166

    
167
/**
168
 * Private data for the RTSP demuxer.
169
 *
170
 * @todo Use ByteIOContext instead of URLContext
171
 */
172
typedef struct RTSPState {
173
    URLContext *rtsp_hd; /* RTSP TCP connexion handle */
174

    
175
    /** number of items in the 'rtsp_streams' variable */
176
    int nb_rtsp_streams;
177

    
178
    struct RTSPStream **rtsp_streams; /**< streams in this session */
179

    
180
    /** indicator of whether we are currently receiving data from the
181
     * server. Basically this isn't more than a simple cache of the
182
     * last PLAY/PAUSE command sent to the server, to make sure we don't
183
     * send 2x the same unexpectedly or commands in the wrong state. */
184
    enum RTSPClientState state;
185

    
186
    /** the seek value requested when calling av_seek_frame(). This value
187
     * is subsequently used as part of the "Range" parameter when emitting
188
     * the RTSP PLAY command. If we are currently playing, this command is
189
     * called instantly. If we are currently paused, this command is called
190
     * whenever we resume playback. Either way, the value is only used once,
191
     * see rtsp_read_play() and rtsp_read_seek(). */
192
    int64_t seek_timestamp;
193

    
194
    /* XXX: currently we use unbuffered input */
195
    //    ByteIOContext rtsp_gb;
196

    
197
    int seq;                          /**< RTSP command sequence number */
198

    
199
    /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
200
     * identifier that the client should re-transmit in each RTSP command */
201
    char session_id[512];
202

    
203
    /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
204
     * the server will go without traffic on the RTSP/TCP line before it
205
     * closes the connection. */
206
    int timeout;
207

    
208
    /** timestamp of the last RTSP command that we sent to the RTSP server.
209
     * This is used to calculate when to send dummy commands to keep the
210
     * connection alive, in conjunction with timeout. */
211
    int64_t last_cmd_time;
212

    
213
    /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
214
    enum RTSPTransport transport;
215

    
216
    /** the negotiated network layer transport protocol; e.g. TCP or UDP
217
     * uni-/multicast */
218
    enum RTSPLowerTransport lower_transport;
219

    
220
    /** brand of server that we're talking to; e.g. WMS, REAL or other.
221
     * Detected based on the value of RTSPMessageHeader->server or the presence
222
     * of RTSPMessageHeader->real_challenge */
223
    enum RTSPServerType server_type;
224

    
225
    /** The last reply of the server to a RTSP command */
226
    char last_reply[2048]; /* XXX: allocate ? */
227

    
228
    /** RTSPStream->transport_priv of the last stream that we read a
229
     * packet from */
230
    void *cur_transport_priv;
231

    
232
    /** The following are used for Real stream selection */
233
    //@{
234
    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
235
    int need_subscription;
236

    
237
    /** stream setup during the last frame read. This is used to detect if
238
     * we need to subscribe or unsubscribe to any new streams. */
239
    enum AVDiscard real_setup_cache[MAX_STREAMS];
240

    
241
    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
242
     * this is used to send the same "Unsubscribe:" if stream setup changed,
243
     * before sending a new "Subscribe:" command. */
244
    char last_subscription[1024];
245
    //@}
246

    
247
    /** The following are used for RTP/ASF streams */
248
    //@{
249
    /** ASF demuxer context for the embedded ASF stream from WMS servers */
250
    AVFormatContext *asf_ctx;
251
    //@}
252
} RTSPState;
253

    
254
/**
255
 * Describes a single stream, as identified by a single m= line block in the
256
 * SDP content. In the case of RDT, one RTSPStream can represent multiple
257
 * AVStreams. In this case, each AVStream in this set has similar content
258
 * (but different codec/bitrate).
259
 */
260
typedef struct RTSPStream {
261
    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
262
    void *transport_priv; /**< RTP/RDT parse context */
263

    
264
    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
265
    int stream_index;
266

    
267
    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
268
     * for the selected transport. Only used for TCP. */
269
    int interleaved_min, interleaved_max;
270

    
271
    char control_url[1024];   /**< url for this stream (from SDP) */
272

    
273
    /** The following are used only in SDP, not RTSP */
274
    //@{
275
    int sdp_port;             /**< port (from SDP content) */
276
    struct in_addr sdp_ip;    /**< IP address (from SDP content) */
277
    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
278
    int sdp_payload_type;     /**< payload type */
279
    //@}
280

    
281
    /** rtp payload parsing infos from SDP (i.e. mapping between private
282
     * payload IDs and media-types (string), so that we can derive what
283
     * type of payload we're dealing with (and how to parse it). */
284
    RTPPayloadData rtp_payload_data;
285

    
286
    /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
287
    //@{
288
    /** handler structure */
289
    RTPDynamicProtocolHandler *dynamic_handler;
290

    
291
    /** private data associated with the dynamic protocol */
292
    PayloadContext *dynamic_protocol_context;
293
    //@}
294
} RTSPStream;
295

    
296
int rtsp_init(void);
297
void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf);
298

    
299
#if LIBAVFORMAT_VERSION_INT < (53 << 16)
300
extern int rtsp_default_protocols;
301
#endif
302
extern int rtsp_rtp_port_min;
303
extern int rtsp_rtp_port_max;
304

    
305
int rtsp_pause(AVFormatContext *s);
306
int rtsp_resume(AVFormatContext *s);
307

    
308
#endif /* FFMPEG_RTSP_H */