summaryrefslogtreecommitdiff
path: root/libavformat/rtsp.h
blob: ce010eca206889a95068f0f2186ef360c7eb1f7b (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
/*
 * RTSP definitions
 * Copyright (c) 2002 Fabrice Bellard
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
#ifndef FFMPEG_RTSP_H
#define FFMPEG_RTSP_H

#include <stdint.h>
#include "avformat.h"
#include "rtspcodes.h"
#include "rtp.h"
#include "network.h"

enum RTSPLowerTransport {
    RTSP_LOWER_TRANSPORT_UDP = 0,
    RTSP_LOWER_TRANSPORT_TCP = 1,
    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
    /**
     * This is not part of public API and shouldn't be used outside of ffmpeg.
     */
    RTSP_LOWER_TRANSPORT_LAST
};

#define RTSP_DEFAULT_PORT   554
#define RTSP_MAX_TRANSPORTS 8
#define RTSP_TCP_MAX_PACKET_SIZE 1472
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
#define RTSP_RTP_PORT_MIN 5000
#define RTSP_RTP_PORT_MAX 10000

typedef struct RTSPTransportField {
    int interleaved_min, interleaved_max;  /**< interleave ids, if TCP transport */
    int port_min, port_max; /**< RTP ports */
    int client_port_min, client_port_max; /**< RTP ports */
    int server_port_min, server_port_max; /**< RTP ports */
    int ttl; /**< ttl value */
    uint32_t destination; /**< destination IP address */
    int transport;
    enum RTSPLowerTransport lower_transport;
} RTSPTransportField;

typedef struct RTSPHeader {
    int content_length;
    enum RTSPStatusCode status_code; /**< response code from server */
    int nb_transports;
    /** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
    int64_t range_start, range_end;
    RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
    int seq; /**< sequence number */
    char session_id[512];
    char real_challenge[64]; /**< the RealChallenge1 field from the server */
    char server[64];
} RTSPHeader;

enum RTSPClientState {
    RTSP_STATE_IDLE,
    RTSP_STATE_PLAYING,
    RTSP_STATE_PAUSED,
};

enum RTSPServerType {
    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
    RTSP_SERVER_REAL, /**< Realmedia-style server */
    RTSP_SERVER_WMS,  /**< Windows Media server */
    RTSP_SERVER_LAST
};

enum RTSPTransport {
    RTSP_TRANSPORT_RTP,
    RTSP_TRANSPORT_RDT,
    RTSP_TRANSPORT_LAST
};

typedef struct RTSPState {
    URLContext *rtsp_hd; /* RTSP TCP connexion handle */
    int nb_rtsp_streams;
    struct RTSPStream **rtsp_streams;

    enum RTSPClientState state;
    int64_t seek_timestamp;

    /* XXX: currently we use unbuffered input */
    //    ByteIOContext rtsp_gb;
    int seq;        /* RTSP command sequence number */
    char session_id[512];
    enum RTSPTransport transport;
    enum RTSPLowerTransport lower_transport;
    enum RTSPServerType server_type;
    char last_reply[2048]; /* XXX: allocate ? */
    void *cur_tx;
    int need_subscription;
    enum AVDiscard real_setup_cache[MAX_STREAMS];
    char last_subscription[1024];
} RTSPState;

typedef struct RTSPStream {
    URLContext *rtp_handle; /* RTP stream handle */
    void *tx_ctx; /* RTP/RDT parse context */

    int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
    int interleaved_min, interleaved_max;  /* interleave ids, if TCP transport */
    char control_url[1024]; /* url for this stream (from SDP) */

    int sdp_port; /* port (from SDP content - not used in RTSP) */
    struct in_addr sdp_ip; /* IP address  (from SDP content - not used in RTSP) */
    int sdp_ttl;  /* IP TTL (from SDP content - not used in RTSP) */
    int sdp_payload_type; /* payload type - only used in SDP */
    RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */

    RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
    PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
} RTSPStream;

/** the callback can be used to extend the connection setup/teardown step */
enum RTSPCallbackAction {
    RTSP_ACTION_SERVER_SETUP,
    RTSP_ACTION_SERVER_TEARDOWN,
    RTSP_ACTION_CLIENT_SETUP,
    RTSP_ACTION_CLIENT_TEARDOWN,
};

typedef struct RTSPActionServerSetup {
    uint32_t ipaddr;
    char transport_option[512];
} RTSPActionServerSetup;

typedef int FFRTSPCallback(enum RTSPCallbackAction action,
                           const char *session_id,
                           char *buf, int buf_size,
                           void *arg);

int rtsp_init(void);
void rtsp_parse_line(RTSPHeader *reply, const char *buf);

#if LIBAVFORMAT_VERSION_INT < (53 << 16)
extern int rtsp_default_protocols;
#endif
extern int rtsp_rtp_port_min;
extern int rtsp_rtp_port_max;

int rtsp_pause(AVFormatContext *s);
int rtsp_resume(AVFormatContext *s);

#endif /* FFMPEG_RTSP_H */