Remove '\p', '\c' and '\e' doxygen markup from doxy, as it should
[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 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 */