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 */
33enum 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 */
45enum 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 */
66typedef 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 */
101typedef 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} RTSPMessageHeader;
136
137/**
138 * Client state, i.e. whether we are currently receiving data (PLAYING) or
139 * setup-but-not-receiving (PAUSED). State can be changed in applications
140 * by calling av_read_play/pause().
141 */
142enum RTSPClientState {
143    RTSP_STATE_IDLE,    /**< not initialized */
144    RTSP_STATE_PLAYING, /**< initialized and receiving data */
145    RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
146};
147
148/**
149 * Identifies particular servers that require special handling, such as
150 * standards-incompliant "Transport:" lines in the SETUP request.
151 */
152enum RTSPServerType {
153    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
154    RTSP_SERVER_REAL, /**< Realmedia-style server */
155    RTSP_SERVER_WMS,  /**< Windows Media server */
156    RTSP_SERVER_NB
157};
158
159/**
160 * Private data for the RTSP demuxer.
161 */
162typedef struct RTSPState {
163    URLContext *rtsp_hd; /* RTSP TCP connexion handle */
164
165    /** number of items in the 'rtsp_streams' variable */
166    int nb_rtsp_streams;
167
168    struct RTSPStream **rtsp_streams; /**< streams in this session */
169
170    /** indicator of whether we are currently receiving data from the
171     * server. Basically this isn't more than a simple cache of the
172     * last PLAY/PAUSE command sent to the server, to make sure we don't
173     * send 2x the same unexpectedly or commands in the wrong state. */
174    enum RTSPClientState state;
175
176    /** the seek value requested when calling av_seek_frame(). This value
177     * is subsequently used as part of the "Range" parameter when emitting
178     * the RTSP PLAY command. If we are currently playing, this command is
179     * called instantly. If we are currently paused, this command is called
180     * whenever we resume playback. Either way, the value is only used once,
181     * see rtsp_read_play() and rtsp_read_seek(). */
182    int64_t seek_timestamp;
183
184    /* XXX: currently we use unbuffered input */
185    //    ByteIOContext rtsp_gb;
186
187    int seq;                          /**< RTSP command sequence number */
188
189    /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
190     * identifier that the client should re-transmit in each RTSP command */
191    char session_id[512];
192
193    /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
194    enum RTSPTransport transport;
195
196    /** the negotiated network layer transport protocol; e.g. TCP or UDP
197     * uni-/multicast */
198    enum RTSPLowerTransport lower_transport;
199
200    /** brand of server that we're talking to; e.g. WMS, REAL or other.
201     * Detected based on the value of RTSPMessageHeader->server or the presence
202     * of RTSPMessageHeader->real_challenge */
203    enum RTSPServerType server_type;
204
205    /** The last reply of the server to a RTSP command */
206    char last_reply[2048]; /* XXX: allocate ? */
207
208    /** RTSPStream->transport_priv of the last stream that we read a
209     * packet from */
210    void *cur_transport_priv;
211
212    /** The following are used for Real stream selection */
213    //@{
214    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
215    int need_subscription;
216
217    /** stream setup during the last frame read. This is used to detect if
218     * we need to subscribe or unsubscribe to any new streams. */
219    enum AVDiscard real_setup_cache[MAX_STREAMS];
220
221    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
222     * this is used to send the same "Unsubscribe:" if stream setup changed,
223     * before sending a new "Subscribe:" command. */
224    char last_subscription[1024];
225    //@}
226} RTSPState;
227
228/**
229 * Describes a single stream, as identified by a single m= line block in the
230 * SDP content. In the case of RDT, one RTSPStream can represent multiple
231 * AVStreams. In this case, each AVStream in this set has similar content
232 * (but different codec/bitrate).
233 */
234typedef struct RTSPStream {
235    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
236    void *transport_priv; /**< RTP/RDT parse context */
237
238    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
239    int stream_index;
240
241    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
242     * for the selected transport. Only used for TCP. */
243    int interleaved_min, interleaved_max;
244
245    char control_url[1024];   /**< url for this stream (from SDP) */
246
247    /** The following are used only in SDP, not RTSP */
248    //@{
249    int sdp_port;             /**< port (from SDP content) */
250    struct in_addr sdp_ip;    /**< IP address (from SDP content) */
251    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
252    int sdp_payload_type;     /**< payload type */
253    //@}
254
255    /** rtp payload parsing infos from SDP (i.e. mapping between private
256     * payload IDs and media-types (string), so that we can derive what
257     * type of payload we're dealing with (and how to parse it). */
258    RTPPayloadData rtp_payload_data;
259
260    /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
261    //@{
262    /** handler structure */
263    RTPDynamicProtocolHandler *dynamic_handler;
264
265    /** private data associated with the dynamic protocol */
266    PayloadContext *dynamic_protocol_context;
267    //@}
268} RTSPStream;
269
270int rtsp_init(void);
271void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf);
272
273#if LIBAVFORMAT_VERSION_INT < (53 << 16)
274extern int rtsp_default_protocols;
275#endif
276extern int rtsp_rtp_port_min;
277extern int rtsp_rtp_port_max;
278
279int rtsp_pause(AVFormatContext *s);
280int rtsp_resume(AVFormatContext *s);
281
282#endif /* FFMPEG_RTSP_H */
283