Mercurial > libavformat.hg
annotate rtsp.h @ 4217:2f3780431c5a libavformat
consider other codecs packets contain only one access unit, not just audio
author | bcoudurier |
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date | Wed, 14 Jan 2009 23:53:31 +0000 |
parents | 6af3e7ab7cbb |
children | 77e0c7511d41 |
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 */ |
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69 char server[64]; |
0 | 70 } RTSPHeader; |
71 | |
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72 enum RTSPClientState { |
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73 RTSP_STATE_IDLE, |
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74 RTSP_STATE_PLAYING, |
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75 RTSP_STATE_PAUSED, |
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76 }; |
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77 |
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78 enum RTSPServerType { |
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79 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ |
4170 | 80 RTSP_SERVER_REAL, /**< Realmedia-style server */ |
81 RTSP_SERVER_WMS, /**< Windows Media server */ | |
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82 RTSP_SERVER_LAST |
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83 }; |
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84 |
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85 enum RTSPTransport { |
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86 RTSP_TRANSPORT_RTP, |
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87 RTSP_TRANSPORT_RDT, |
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88 RTSP_TRANSPORT_LAST |
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89 }; |
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90 |
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91 typedef struct RTSPState { |
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92 URLContext *rtsp_hd; /* RTSP TCP connexion handle */ |
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93 int nb_rtsp_streams; |
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94 struct RTSPStream **rtsp_streams; |
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95 |
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96 enum RTSPClientState state; |
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97 int64_t seek_timestamp; |
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98 |
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99 /* XXX: currently we use unbuffered input */ |
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100 // ByteIOContext rtsp_gb; |
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101 int seq; /* RTSP command sequence number */ |
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102 char session_id[512]; |
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103 enum RTSPTransport transport; |
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104 enum RTSPLowerTransport lower_transport; |
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105 enum RTSPServerType server_type; |
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106 char last_reply[2048]; /* XXX: allocate ? */ |
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107 void *cur_tx; |
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108 int need_subscription; |
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109 enum AVDiscard real_setup_cache[MAX_STREAMS]; |
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110 char last_subscription[1024]; |
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111 } RTSPState; |
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112 |
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113 typedef struct RTSPStream { |
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114 URLContext *rtp_handle; /* RTP stream handle */ |
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115 void *tx_ctx; /* RTP/RDT parse context */ |
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116 |
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117 int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ |
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118 int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ |
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119 char control_url[1024]; /* url for this stream (from SDP) */ |
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120 |
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121 int sdp_port; /* port (from SDP content - not used in RTSP) */ |
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122 struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ |
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123 int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ |
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124 int sdp_payload_type; /* payload type - only used in SDP */ |
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125 RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */ |
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126 |
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127 RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) |
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128 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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129 } RTSPStream; |
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130 |
1871 | 131 /** the callback can be used to extend the connection setup/teardown step */ |
0 | 132 enum RTSPCallbackAction { |
133 RTSP_ACTION_SERVER_SETUP, | |
134 RTSP_ACTION_SERVER_TEARDOWN, | |
135 RTSP_ACTION_CLIENT_SETUP, | |
136 RTSP_ACTION_CLIENT_TEARDOWN, | |
137 }; | |
138 | |
139 typedef struct RTSPActionServerSetup { | |
65 | 140 uint32_t ipaddr; |
0 | 141 char transport_option[512]; |
142 } RTSPActionServerSetup; | |
143 | |
885 | 144 typedef int FFRTSPCallback(enum RTSPCallbackAction action, |
0 | 145 const char *session_id, |
146 char *buf, int buf_size, | |
147 void *arg); | |
148 | |
149 int rtsp_init(void); | |
150 void rtsp_parse_line(RTSPHeader *reply, const char *buf); | |
151 | |
2884 | 152 #if LIBAVFORMAT_VERSION_INT < (53 << 16) |
0 | 153 extern int rtsp_default_protocols; |
2884 | 154 #endif |
0 | 155 extern int rtsp_rtp_port_min; |
156 extern int rtsp_rtp_port_max; | |
179 | 157 |
158 int rtsp_pause(AVFormatContext *s); | |
159 int rtsp_resume(AVFormatContext *s); | |
0 | 160 |
3853 | 161 #endif /* FFMPEG_RTSP_H */ |