Mercurial > libavformat.hg
annotate rtsp.h @ 5479:17744e7f9e4c libavformat
Remove a useless local variable.
author | lucabe |
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date | Mon, 28 Dec 2009 13:41:09 +0000 |
parents | 715913c5514f |
children | 016f3d4fe56d |
rev | line source |
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0 | 1 /* |
2 * RTSP definitions | |
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3 * Copyright (c) 2002 Fabrice Bellard |
0 | 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 */ |
5104 | 21 #ifndef AVFORMAT_RTSP_H |
22 #define AVFORMAT_RTSP_H | |
0 | 23 |
2171 | 24 #include <stdint.h> |
25 #include "avformat.h" | |
0 | 26 #include "rtspcodes.h" |
4388 | 27 #include "rtpdec.h" |
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28 #include "network.h" |
0 | 29 |
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30 /** |
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31 * Network layer over which RTP/etc packet data will be transported. |
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32 */ |
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33 enum RTSPLowerTransport { |
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34 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */ |
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35 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */ |
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36 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */ |
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37 RTSP_LOWER_TRANSPORT_NB |
0 | 38 }; |
39 | |
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40 /** |
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41 * Packet profile of the data that we will be receiving. Real servers |
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42 * commonly send RDT (although they can sometimes send RTP as well), |
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43 * whereas most others will send RTP. |
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44 */ |
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45 enum RTSPTransport { |
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46 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ |
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47 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ |
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48 RTSP_TRANSPORT_NB |
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49 }; |
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50 |
0 | 51 #define RTSP_DEFAULT_PORT 554 |
52 #define RTSP_MAX_TRANSPORTS 8 | |
179 | 53 #define RTSP_TCP_MAX_PACKET_SIZE 1472 |
774 | 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 | |
0 | 58 |
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59 /** |
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60 * This describes a single item in the "Transport:" line of one stream as |
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61 * negotiated by the SETUP RTSP command. Multiple transports are comma- |
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62 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp; |
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63 * client_port=1000-1001;server_port=1800-1801") and described in separate |
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64 * RTSPTransportFields. |
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65 */ |
0 | 66 typedef struct RTSPTransportField { |
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67 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts |
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68 * with a '$', stream length and stream ID. If the stream ID is within |
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69 * the range of this interleaved_min-max, then the packet belongs to |
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70 * this stream. */ |
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71 int interleaved_min, interleaved_max; |
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72 |
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73 /** UDP multicast port range; the ports to which we should connect to |
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74 * receive multicast UDP data. */ |
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75 int port_min, port_max; |
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76 |
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77 /** UDP client ports; these should be the local ports of the UDP RTP |
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78 * (and RTCP) sockets over which we receive RTP/RTCP data. */ |
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79 int client_port_min, client_port_max; |
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80 |
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81 /** UDP unicast server port range; the ports to which we should connect |
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82 * to receive unicast UDP RTP/RTCP data. */ |
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83 int server_port_min, server_port_max; |
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84 |
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85 /** time-to-live value (required for multicast); the amount of HOPs that |
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86 * packets will be allowed to make before being discarded. */ |
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87 int ttl; |
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88 |
1871 | 89 uint32_t destination; /**< destination IP address */ |
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90 |
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91 /** data/packet transport protocol; e.g. RTP or RDT */ |
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92 enum RTSPTransport transport; |
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93 |
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94 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */ |
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95 enum RTSPLowerTransport lower_transport; |
0 | 96 } RTSPTransportField; |
97 | |
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98 /** |
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99 * This describes the server response to each RTSP command. |
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100 */ |
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101 typedef struct RTSPMessageHeader { |
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102 /** length of the data following this header */ |
0 | 103 int content_length; |
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104 |
1871 | 105 enum RTSPStatusCode status_code; /**< response code from server */ |
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106 |
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107 /** number of items in the 'transports' variable below */ |
0 | 108 int nb_transports; |
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109 |
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110 /** Time range of the streams that the server will stream. In |
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111 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ |
885 | 112 int64_t range_start, range_end; |
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113 |
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114 /** describes the complete "Transport:" line of the server in response |
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115 * to a SETUP RTSP command by the client */ |
0 | 116 RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; |
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117 |
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118 int seq; /**< sequence number */ |
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119 |
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120 /** the "Session:" field. This value is initially set by the server and |
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121 * should be re-transmitted by the client in every RTSP command. */ |
0 | 122 char session_id[512]; |
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123 |
5333 | 124 /** the "Location:" field. This value is used to handle redirection. |
125 */ | |
126 char location[4096]; | |
127 | |
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128 /** the "RealChallenge1:" field from the server */ |
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129 char real_challenge[64]; |
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130 |
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131 /** the "Server: field, which can be used to identify some special-case |
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132 * servers that are not 100% standards-compliant. We use this to identify |
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133 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where |
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134 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers |
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135 * use something like "Helix [..] Server Version v.e.r.sion (platform) |
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136 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)", |
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137 * where platform is the output of $uname -msr | sed 's/ /-/g'. */ |
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138 char server[64]; |
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139 |
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140 /** The "timeout" comes as part of the server response to the "SETUP" |
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141 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the |
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142 * time, in seconds, that the server will go without traffic over the |
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143 * RTSP/TCP connection before it closes the connection. To prevent |
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144 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller |
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145 * than this value. */ |
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146 int timeout; |
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147 |
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148 /** The "Notice" or "X-Notice" field value. See |
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149 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00 |
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150 * for a complete list of supported values. */ |
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151 int notice; |
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152 } RTSPMessageHeader; |
0 | 153 |
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154 /** |
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155 * Client state, i.e. whether we are currently receiving data (PLAYING) or |
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156 * setup-but-not-receiving (PAUSED). State can be changed in applications |
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157 * by calling av_read_play/pause(). |
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158 */ |
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159 enum RTSPClientState { |
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160 RTSP_STATE_IDLE, /**< not initialized */ |
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161 RTSP_STATE_PLAYING, /**< initialized and receiving data */ |
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162 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ |
5013 | 163 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ |
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164 }; |
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165 |
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166 /** |
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167 * Identifies particular servers that require special handling, such as |
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168 * standards-incompliant "Transport:" lines in the SETUP request. |
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169 */ |
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170 enum RTSPServerType { |
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171 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ |
4170 | 172 RTSP_SERVER_REAL, /**< Realmedia-style server */ |
173 RTSP_SERVER_WMS, /**< Windows Media server */ | |
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174 RTSP_SERVER_NB |
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175 }; |
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176 |
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177 /** |
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178 * Private data for the RTSP demuxer. |
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179 * |
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180 * @todo Use ByteIOContext instead of URLContext |
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181 */ |
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182 typedef struct RTSPState { |
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183 URLContext *rtsp_hd; /* RTSP TCP connexion handle */ |
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184 |
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185 /** number of items in the 'rtsp_streams' variable */ |
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186 int nb_rtsp_streams; |
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187 |
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188 struct RTSPStream **rtsp_streams; /**< streams in this session */ |
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189 |
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190 /** indicator of whether we are currently receiving data from the |
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191 * server. Basically this isn't more than a simple cache of the |
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192 * last PLAY/PAUSE command sent to the server, to make sure we don't |
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193 * send 2x the same unexpectedly or commands in the wrong state. */ |
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194 enum RTSPClientState state; |
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195 |
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196 /** the seek value requested when calling av_seek_frame(). This value |
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197 * is subsequently used as part of the "Range" parameter when emitting |
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198 * the RTSP PLAY command. If we are currently playing, this command is |
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199 * called instantly. If we are currently paused, this command is called |
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200 * whenever we resume playback. Either way, the value is only used once, |
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201 * see rtsp_read_play() and rtsp_read_seek(). */ |
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202 int64_t seek_timestamp; |
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203 |
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204 /* XXX: currently we use unbuffered input */ |
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205 // ByteIOContext rtsp_gb; |
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206 |
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207 int seq; /**< RTSP command sequence number */ |
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208 |
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209 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session |
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210 * identifier that the client should re-transmit in each RTSP command */ |
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211 char session_id[512]; |
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212 |
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213 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that |
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214 * the server will go without traffic on the RTSP/TCP line before it |
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215 * closes the connection. */ |
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216 int timeout; |
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217 |
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218 /** timestamp of the last RTSP command that we sent to the RTSP server. |
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219 * This is used to calculate when to send dummy commands to keep the |
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220 * connection alive, in conjunction with timeout. */ |
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221 int64_t last_cmd_time; |
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222 |
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223 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */ |
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224 enum RTSPTransport transport; |
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225 |
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226 /** the negotiated network layer transport protocol; e.g. TCP or UDP |
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227 * uni-/multicast */ |
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228 enum RTSPLowerTransport lower_transport; |
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229 |
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230 /** brand of server that we're talking to; e.g. WMS, REAL or other. |
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231 * Detected based on the value of RTSPMessageHeader->server or the presence |
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232 * of RTSPMessageHeader->real_challenge */ |
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233 enum RTSPServerType server_type; |
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234 |
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235 /** base64-encoded authorization lines (username:password) */ |
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236 char *auth_b64; |
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237 |
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238 /** The last reply of the server to a RTSP command */ |
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239 char last_reply[2048]; /* XXX: allocate ? */ |
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240 |
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241 /** RTSPStream->transport_priv of the last stream that we read a |
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242 * packet from */ |
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243 void *cur_transport_priv; |
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244 |
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245 /** The following are used for Real stream selection */ |
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246 //@{ |
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247 /** whether we need to send a "SET_PARAMETER Subscribe:" command */ |
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248 int need_subscription; |
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249 |
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250 /** stream setup during the last frame read. This is used to detect if |
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251 * we need to subscribe or unsubscribe to any new streams. */ |
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252 enum AVDiscard real_setup_cache[MAX_STREAMS]; |
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253 |
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254 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command. |
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255 * this is used to send the same "Unsubscribe:" if stream setup changed, |
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256 * before sending a new "Subscribe:" command. */ |
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257 char last_subscription[1024]; |
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258 //@} |
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259 |
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260 /** The following are used for RTP/ASF streams */ |
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261 //@{ |
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262 /** ASF demuxer context for the embedded ASF stream from WMS servers */ |
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263 AVFormatContext *asf_ctx; |
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264 |
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265 /** cache for position of the asf demuxer, since we load a new |
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266 * data packet in the bytecontext for each incoming RTSP packet. */ |
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267 uint64_t asf_pb_pos; |
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268 //@} |
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269 } RTSPState; |
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270 |
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271 /** |
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272 * Describes a single stream, as identified by a single m= line block in the |
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273 * SDP content. In the case of RDT, one RTSPStream can represent multiple |
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274 * AVStreams. In this case, each AVStream in this set has similar content |
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275 * (but different codec/bitrate). |
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276 */ |
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277 typedef struct RTSPStream { |
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278 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ |
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279 void *transport_priv; /**< RTP/RDT parse context */ |
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280 |
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281 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */ |
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282 int stream_index; |
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283 |
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284 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max |
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285 * for the selected transport. Only used for TCP. */ |
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286 int interleaved_min, interleaved_max; |
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287 |
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288 char control_url[1024]; /**< url for this stream (from SDP) */ |
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289 |
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290 /** The following are used only in SDP, not RTSP */ |
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291 //@{ |
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292 int sdp_port; /**< port (from SDP content) */ |
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293 struct in_addr sdp_ip; /**< IP address (from SDP content) */ |
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294 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */ |
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295 int sdp_payload_type; /**< payload type */ |
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296 //@} |
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297 |
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298 /** rtp payload parsing infos from SDP (i.e. mapping between private |
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299 * payload IDs and media-types (string), so that we can derive what |
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300 * type of payload we're dealing with (and how to parse it). */ |
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301 RTPPayloadData rtp_payload_data; |
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302 |
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303 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */ |
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304 //@{ |
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305 /** handler structure */ |
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306 RTPDynamicProtocolHandler *dynamic_handler; |
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307 |
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308 /** private data associated with the dynamic protocol */ |
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309 PayloadContext *dynamic_protocol_context; |
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310 //@} |
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311 } RTSPStream; |
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312 |
0 | 313 int rtsp_init(void); |
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314 void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf); |
0 | 315 |
2884 | 316 #if LIBAVFORMAT_VERSION_INT < (53 << 16) |
0 | 317 extern int rtsp_default_protocols; |
2884 | 318 #endif |
0 | 319 extern int rtsp_rtp_port_min; |
320 extern int rtsp_rtp_port_max; | |
179 | 321 |
322 int rtsp_pause(AVFormatContext *s); | |
323 int rtsp_resume(AVFormatContext *s); | |
0 | 324 |
5104 | 325 #endif /* AVFORMAT_RTSP_H */ |