Mercurial > libavformat.hg
annotate rtsp.h @ 4168:0ebf63d12ea4 libavformat
Export RTSPState and RTSPStream from rtsp.c into rtsp.h. This allows future
access to these structures in functions that will be located in rtp_asf.c.
See "[PATCH] RTSP-MS 2/15: export RTSPState and RTSPStream" mailinglist
thread.
author | rbultje |
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date | Wed, 07 Jan 2009 14:53:04 +0000 |
parents | 13e9b0d3a314 |
children | 619845a9bab3 |
rev | line source |
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0 | 1 /* |
2 * RTSP definitions | |
3 * Copyright (c) 2002 Fabrice Bellard. | |
4 * | |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
0 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 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 | |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
3853 | 21 #ifndef FFMPEG_RTSP_H |
22 #define FFMPEG_RTSP_H | |
0 | 23 |
2171 | 24 #include <stdint.h> |
25 #include "avformat.h" | |
0 | 26 #include "rtspcodes.h" |
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27 #include "rtp.h" |
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28 #include "network.h" |
0 | 29 |
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30 enum RTSPLowerTransport { |
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31 RTSP_LOWER_TRANSPORT_UDP = 0, |
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32 RTSP_LOWER_TRANSPORT_TCP = 1, |
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33 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, |
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34 /** |
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35 * This is not part of public API and shouldn't be used outside of ffmpeg. |
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36 */ |
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37 RTSP_LOWER_TRANSPORT_LAST |
0 | 38 }; |
39 | |
40 #define RTSP_DEFAULT_PORT 554 | |
41 #define RTSP_MAX_TRANSPORTS 8 | |
179 | 42 #define RTSP_TCP_MAX_PACKET_SIZE 1472 |
774 | 43 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2 |
44 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 | |
45 #define RTSP_RTP_PORT_MIN 5000 | |
46 #define RTSP_RTP_PORT_MAX 10000 | |
0 | 47 |
48 typedef struct RTSPTransportField { | |
1871 | 49 int interleaved_min, interleaved_max; /**< interleave ids, if TCP transport */ |
50 int port_min, port_max; /**< RTP ports */ | |
51 int client_port_min, client_port_max; /**< RTP ports */ | |
52 int server_port_min, server_port_max; /**< RTP ports */ | |
53 int ttl; /**< ttl value */ | |
54 uint32_t destination; /**< destination IP address */ | |
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55 int transport; |
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56 enum RTSPLowerTransport lower_transport; |
0 | 57 } RTSPTransportField; |
58 | |
59 typedef struct RTSPHeader { | |
60 int content_length; | |
1871 | 61 enum RTSPStatusCode status_code; /**< response code from server */ |
0 | 62 int nb_transports; |
1871 | 63 /** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ |
885 | 64 int64_t range_start, range_end; |
0 | 65 RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; |
1871 | 66 int seq; /**< sequence number */ |
0 | 67 char session_id[512]; |
3855 | 68 char real_challenge[64]; /**< the RealChallenge1 field from the server */ |
0 | 69 } RTSPHeader; |
70 | |
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71 enum RTSPClientState { |
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72 RTSP_STATE_IDLE, |
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73 RTSP_STATE_PLAYING, |
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74 RTSP_STATE_PAUSED, |
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75 }; |
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76 |
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77 enum RTSPServerType { |
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78 RTSP_SERVER_RTP, /*< Standard-compliant RTP-server */ |
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79 RTSP_SERVER_REAL, /*< Realmedia-style server */ |
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80 RTSP_SERVER_LAST |
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81 }; |
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82 |
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83 enum RTSPTransport { |
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84 RTSP_TRANSPORT_RTP, |
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85 RTSP_TRANSPORT_RDT, |
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86 RTSP_TRANSPORT_LAST |
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87 }; |
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88 |
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89 typedef struct RTSPState { |
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90 URLContext *rtsp_hd; /* RTSP TCP connexion handle */ |
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91 int nb_rtsp_streams; |
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92 struct RTSPStream **rtsp_streams; |
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93 |
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94 enum RTSPClientState state; |
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95 int64_t seek_timestamp; |
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96 |
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97 /* XXX: currently we use unbuffered input */ |
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98 // ByteIOContext rtsp_gb; |
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99 int seq; /* RTSP command sequence number */ |
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100 char session_id[512]; |
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101 enum RTSPTransport transport; |
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102 enum RTSPLowerTransport lower_transport; |
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103 enum RTSPServerType server_type; |
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104 char last_reply[2048]; /* XXX: allocate ? */ |
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105 void *cur_tx; |
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106 int need_subscription; |
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107 enum AVDiscard real_setup_cache[MAX_STREAMS]; |
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108 char last_subscription[1024]; |
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109 } RTSPState; |
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110 |
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111 typedef struct RTSPStream { |
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112 URLContext *rtp_handle; /* RTP stream handle */ |
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113 void *tx_ctx; /* RTP/RDT parse context */ |
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114 |
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115 int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ |
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116 int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ |
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117 char control_url[1024]; /* url for this stream (from SDP) */ |
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118 |
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119 int sdp_port; /* port (from SDP content - not used in RTSP) */ |
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120 struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ |
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121 int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ |
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122 int sdp_payload_type; /* payload type - only used in SDP */ |
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123 RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */ |
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124 |
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125 RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) |
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126 PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol) |
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127 } RTSPStream; |
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128 |
1871 | 129 /** the callback can be used to extend the connection setup/teardown step */ |
0 | 130 enum RTSPCallbackAction { |
131 RTSP_ACTION_SERVER_SETUP, | |
132 RTSP_ACTION_SERVER_TEARDOWN, | |
133 RTSP_ACTION_CLIENT_SETUP, | |
134 RTSP_ACTION_CLIENT_TEARDOWN, | |
135 }; | |
136 | |
137 typedef struct RTSPActionServerSetup { | |
65 | 138 uint32_t ipaddr; |
0 | 139 char transport_option[512]; |
140 } RTSPActionServerSetup; | |
141 | |
885 | 142 typedef int FFRTSPCallback(enum RTSPCallbackAction action, |
0 | 143 const char *session_id, |
144 char *buf, int buf_size, | |
145 void *arg); | |
146 | |
147 int rtsp_init(void); | |
148 void rtsp_parse_line(RTSPHeader *reply, const char *buf); | |
149 | |
2884 | 150 #if LIBAVFORMAT_VERSION_INT < (53 << 16) |
0 | 151 extern int rtsp_default_protocols; |
2884 | 152 #endif |
0 | 153 extern int rtsp_rtp_port_min; |
154 extern int rtsp_rtp_port_max; | |
179 | 155 |
156 int rtsp_pause(AVFormatContext *s); | |
157 int rtsp_resume(AVFormatContext *s); | |
0 | 158 |
3853 | 159 #endif /* FFMPEG_RTSP_H */ |