Mercurial > libavformat.hg
annotate rtsp.h @ 6435:67433b0c29d5 libavformat
move h264 demuxer to its own file
author | aurel |
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date | Sun, 29 Aug 2010 21:28:51 +0000 |
parents | 2685bf2b0b78 |
children | 2e3e76e8ada4 |
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" |
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29 #include "httpauth.h" |
0 | 30 |
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31 /** |
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32 * Network layer over which RTP/etc packet data will be transported. |
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33 */ |
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34 enum RTSPLowerTransport { |
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35 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */ |
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36 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */ |
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37 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */ |
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38 RTSP_LOWER_TRANSPORT_NB |
0 | 39 }; |
40 | |
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41 /** |
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42 * Packet profile of the data that we will be receiving. Real servers |
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43 * commonly send RDT (although they can sometimes send RTP as well), |
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44 * whereas most others will send RTP. |
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45 */ |
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46 enum RTSPTransport { |
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47 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ |
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48 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ |
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49 RTSP_TRANSPORT_NB |
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50 }; |
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51 |
6112 | 52 /** |
53 * Transport mode for the RTSP data. This may be plain, or | |
54 * tunneled, which is done over HTTP. | |
55 */ | |
56 enum RTSPControlTransport { | |
57 RTSP_MODE_PLAIN, /**< Normal RTSP */ | |
58 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */ | |
59 }; | |
60 | |
0 | 61 #define RTSP_DEFAULT_PORT 554 |
62 #define RTSP_MAX_TRANSPORTS 8 | |
179 | 63 #define RTSP_TCP_MAX_PACKET_SIZE 1472 |
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64 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1 |
774 | 65 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 |
66 #define RTSP_RTP_PORT_MIN 5000 | |
67 #define RTSP_RTP_PORT_MAX 10000 | |
0 | 68 |
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69 /** |
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70 * This describes a single item in the "Transport:" line of one stream as |
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71 * negotiated by the SETUP RTSP command. Multiple transports are comma- |
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72 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp; |
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73 * client_port=1000-1001;server_port=1800-1801") and described in separate |
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74 * RTSPTransportFields. |
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75 */ |
0 | 76 typedef struct RTSPTransportField { |
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77 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts |
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78 * with a '$', stream length and stream ID. If the stream ID is within |
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79 * the range of this interleaved_min-max, then the packet belongs to |
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80 * this stream. */ |
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81 int interleaved_min, interleaved_max; |
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82 |
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83 /** UDP multicast port range; the ports to which we should connect to |
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84 * receive multicast UDP data. */ |
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85 int port_min, port_max; |
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86 |
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87 /** UDP client ports; these should be the local ports of the UDP RTP |
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88 * (and RTCP) sockets over which we receive RTP/RTCP data. */ |
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89 int client_port_min, client_port_max; |
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90 |
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91 /** UDP unicast server port range; the ports to which we should connect |
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92 * to receive unicast UDP RTP/RTCP data. */ |
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93 int server_port_min, server_port_max; |
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94 |
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95 /** time-to-live value (required for multicast); the amount of HOPs that |
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96 * packets will be allowed to make before being discarded. */ |
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97 int ttl; |
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98 |
6412 | 99 struct sockaddr_storage destination; /**< destination IP address */ |
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100 |
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101 /** data/packet transport protocol; e.g. RTP or RDT */ |
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102 enum RTSPTransport transport; |
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103 |
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104 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */ |
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105 enum RTSPLowerTransport lower_transport; |
0 | 106 } RTSPTransportField; |
107 | |
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108 /** |
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109 * This describes the server response to each RTSP command. |
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110 */ |
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111 typedef struct RTSPMessageHeader { |
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112 /** length of the data following this header */ |
0 | 113 int content_length; |
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114 |
1871 | 115 enum RTSPStatusCode status_code; /**< response code from server */ |
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116 |
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117 /** number of items in the 'transports' variable below */ |
0 | 118 int nb_transports; |
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119 |
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120 /** Time range of the streams that the server will stream. In |
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121 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ |
885 | 122 int64_t range_start, range_end; |
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123 |
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124 /** describes the complete "Transport:" line of the server in response |
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125 * to a SETUP RTSP command by the client */ |
0 | 126 RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; |
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127 |
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128 int seq; /**< sequence number */ |
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129 |
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130 /** the "Session:" field. This value is initially set by the server and |
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131 * should be re-transmitted by the client in every RTSP command. */ |
0 | 132 char session_id[512]; |
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133 |
5333 | 134 /** the "Location:" field. This value is used to handle redirection. |
135 */ | |
136 char location[4096]; | |
137 | |
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138 /** the "RealChallenge1:" field from the server */ |
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139 char real_challenge[64]; |
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140 |
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141 /** the "Server: field, which can be used to identify some special-case |
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142 * servers that are not 100% standards-compliant. We use this to identify |
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143 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where |
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144 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers |
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145 * use something like "Helix [..] Server Version v.e.r.sion (platform) |
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146 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)", |
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147 * where platform is the output of $uname -msr | sed 's/ /-/g'. */ |
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148 char server[64]; |
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149 |
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150 /** The "timeout" comes as part of the server response to the "SETUP" |
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151 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the |
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152 * time, in seconds, that the server will go without traffic over the |
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153 * RTSP/TCP connection before it closes the connection. To prevent |
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154 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller |
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155 * than this value. */ |
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156 int timeout; |
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157 |
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158 /** The "Notice" or "X-Notice" field value. See |
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159 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00 |
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160 * for a complete list of supported values. */ |
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161 int notice; |
6348 | 162 |
163 /** The "reason" is meant to specify better the meaning of the error code | |
164 * returned | |
165 */ | |
166 char reason[256]; | |
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167 } RTSPMessageHeader; |
0 | 168 |
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169 /** |
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170 * Client state, i.e. whether we are currently receiving data (PLAYING) or |
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171 * setup-but-not-receiving (PAUSED). State can be changed in applications |
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172 * by calling av_read_play/pause(). |
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173 */ |
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174 enum RTSPClientState { |
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175 RTSP_STATE_IDLE, /**< not initialized */ |
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176 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ |
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177 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ |
5013 | 178 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ |
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179 }; |
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180 |
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181 /** |
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182 * Identifies particular servers that require special handling, such as |
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183 * standards-incompliant "Transport:" lines in the SETUP request. |
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184 */ |
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185 enum RTSPServerType { |
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186 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ |
4170 | 187 RTSP_SERVER_REAL, /**< Realmedia-style server */ |
188 RTSP_SERVER_WMS, /**< Windows Media server */ | |
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189 RTSP_SERVER_NB |
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190 }; |
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191 |
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192 /** |
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193 * Private data for the RTSP demuxer. |
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194 * |
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195 * @todo Use ByteIOContext instead of URLContext |
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196 */ |
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197 typedef struct RTSPState { |
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198 URLContext *rtsp_hd; /* RTSP TCP connection handle */ |
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199 |
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200 /** number of items in the 'rtsp_streams' variable */ |
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201 int nb_rtsp_streams; |
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202 |
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203 struct RTSPStream **rtsp_streams; /**< streams in this session */ |
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204 |
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205 /** indicator of whether we are currently receiving data from the |
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206 * server. Basically this isn't more than a simple cache of the |
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207 * last PLAY/PAUSE command sent to the server, to make sure we don't |
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208 * send 2x the same unexpectedly or commands in the wrong state. */ |
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209 enum RTSPClientState state; |
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210 |
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211 /** the seek value requested when calling av_seek_frame(). This value |
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212 * is subsequently used as part of the "Range" parameter when emitting |
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213 * the RTSP PLAY command. If we are currently playing, this command is |
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214 * called instantly. If we are currently paused, this command is called |
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215 * whenever we resume playback. Either way, the value is only used once, |
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216 * see rtsp_read_play() and rtsp_read_seek(). */ |
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217 int64_t seek_timestamp; |
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218 |
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219 /* XXX: currently we use unbuffered input */ |
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220 // ByteIOContext rtsp_gb; |
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221 |
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222 int seq; /**< RTSP command sequence number */ |
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223 |
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224 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session |
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225 * identifier that the client should re-transmit in each RTSP command */ |
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226 char session_id[512]; |
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227 |
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228 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that |
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229 * the server will go without traffic on the RTSP/TCP line before it |
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230 * closes the connection. */ |
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231 int timeout; |
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232 |
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233 /** timestamp of the last RTSP command that we sent to the RTSP server. |
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234 * This is used to calculate when to send dummy commands to keep the |
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235 * connection alive, in conjunction with timeout. */ |
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236 int64_t last_cmd_time; |
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237 |
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238 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */ |
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239 enum RTSPTransport transport; |
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240 |
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241 /** the negotiated network layer transport protocol; e.g. TCP or UDP |
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242 * uni-/multicast */ |
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243 enum RTSPLowerTransport lower_transport; |
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244 |
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245 /** brand of server that we're talking to; e.g. WMS, REAL or other. |
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246 * Detected based on the value of RTSPMessageHeader->server or the presence |
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247 * of RTSPMessageHeader->real_challenge */ |
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248 enum RTSPServerType server_type; |
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249 |
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250 /** plaintext authorization line (username:password) */ |
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251 char auth[128]; |
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252 |
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253 /** authentication state */ |
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254 HTTPAuthState auth_state; |
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255 |
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256 /** The last reply of the server to a RTSP command */ |
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257 char last_reply[2048]; /* XXX: allocate ? */ |
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258 |
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259 /** RTSPStream->transport_priv of the last stream that we read a |
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260 * packet from */ |
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261 void *cur_transport_priv; |
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262 |
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263 /** The following are used for Real stream selection */ |
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264 //@{ |
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265 /** whether we need to send a "SET_PARAMETER Subscribe:" command */ |
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266 int need_subscription; |
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267 |
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268 /** stream setup during the last frame read. This is used to detect if |
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269 * we need to subscribe or unsubscribe to any new streams. */ |
6357 | 270 enum AVDiscard *real_setup_cache; |
271 | |
272 /** current stream setup. This is a temporary buffer used to compare | |
273 * current setup to previous frame setup. */ | |
274 enum AVDiscard *real_setup; | |
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275 |
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276 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command. |
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277 * this is used to send the same "Unsubscribe:" if stream setup changed, |
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278 * before sending a new "Subscribe:" command. */ |
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279 char last_subscription[1024]; |
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280 //@} |
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281 |
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282 /** The following are used for RTP/ASF streams */ |
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283 //@{ |
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284 /** ASF demuxer context for the embedded ASF stream from WMS servers */ |
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285 AVFormatContext *asf_ctx; |
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286 |
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287 /** cache for position of the asf demuxer, since we load a new |
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288 * data packet in the bytecontext for each incoming RTSP packet. */ |
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289 uint64_t asf_pb_pos; |
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290 //@} |
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291 |
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292 /** some MS RTSP streams contain a URL in the SDP that we need to use |
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293 * for all subsequent RTSP requests, rather than the input URI; in |
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294 * other cases, this is a copy of AVFormatContext->filename. */ |
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295 char control_uri[1024]; |
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296 |
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297 /** The synchronized start time of the output streams. */ |
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298 int64_t start_time; |
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299 |
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300 /** Additional output handle, used when input and output are done |
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301 * separately, eg for HTTP tunneling. */ |
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302 URLContext *rtsp_hd_out; |
6112 | 303 |
304 /** RTSP transport mode, such as plain or tunneled. */ | |
305 enum RTSPControlTransport control_transport; | |
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306 |
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307 /* Number of RTCP BYE packets the RTSP session has received. |
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308 * An EOF is propagated back if nb_byes == nb_streams. |
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309 * This is reset after a seek. */ |
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310 int nb_byes; |
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311 } RTSPState; |
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312 |
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313 /** |
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314 * Describes a single stream, as identified by a single m= line block in the |
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315 * SDP content. In the case of RDT, one RTSPStream can represent multiple |
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316 * AVStreams. In this case, each AVStream in this set has similar content |
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317 * (but different codec/bitrate). |
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318 */ |
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319 typedef struct RTSPStream { |
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320 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ |
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321 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */ |
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322 |
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323 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */ |
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324 int stream_index; |
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325 |
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326 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max |
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327 * for the selected transport. Only used for TCP. */ |
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328 int interleaved_min, interleaved_max; |
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329 |
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330 char control_url[1024]; /**< url for this stream (from SDP) */ |
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331 |
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332 /** The following are used only in SDP, not RTSP */ |
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333 //@{ |
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334 int sdp_port; /**< port (from SDP content) */ |
6411 | 335 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */ |
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336 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */ |
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337 int sdp_payload_type; /**< payload type */ |
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338 //@} |
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339 |
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340 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */ |
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341 //@{ |
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342 /** handler structure */ |
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343 RTPDynamicProtocolHandler *dynamic_handler; |
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344 |
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345 /** private data associated with the dynamic protocol */ |
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346 PayloadContext *dynamic_protocol_context; |
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347 //@} |
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348 } RTSPStream; |
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349 |
5890 | 350 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, |
351 HTTPAuthState *auth_state); | |
0 | 352 |
2884 | 353 #if LIBAVFORMAT_VERSION_INT < (53 << 16) |
0 | 354 extern int rtsp_default_protocols; |
2884 | 355 #endif |
0 | 356 extern int rtsp_rtp_port_min; |
357 extern int rtsp_rtp_port_max; | |
179 | 358 |
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359 /** |
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360 * Send a command to the RTSP server without waiting for the reply. |
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361 * |
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362 * @param s RTSP (de)muxer context |
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363 * @param method the method for the request |
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364 * @param url the target url for the request |
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365 * @param headers extra header lines to include in the request |
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366 * @param send_content if non-null, the data to send as request body content |
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367 * @param send_content_length the length of the send_content data, or 0 if |
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368 * send_content is null |
6095 | 369 * |
370 * @return zero if success, nonzero otherwise | |
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371 */ |
6095 | 372 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s, |
6096 | 373 const char *method, const char *url, |
374 const char *headers, | |
375 const unsigned char *send_content, | |
376 int send_content_length); | |
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377 /** |
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378 * Send a command to the RTSP server without waiting for the reply. |
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379 * |
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380 * @see rtsp_send_cmd_with_content_async |
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381 */ |
6095 | 382 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, |
6096 | 383 const char *url, const char *headers); |
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384 |
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385 /** |
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386 * Send a command to the RTSP server and wait for the reply. |
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387 * |
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388 * @param s RTSP (de)muxer context |
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389 * @param method the method for the request |
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390 * @param url the target url for the request |
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391 * @param headers extra header lines to include in the request |
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392 * @param reply pointer where the RTSP message header will be stored |
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393 * @param content_ptr pointer where the RTSP message body, if any, will |
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394 * be stored (length is in reply) |
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395 * @param send_content if non-null, the data to send as request body content |
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396 * @param send_content_length the length of the send_content data, or 0 if |
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397 * send_content is null |
6095 | 398 * |
399 * @return zero if success, nonzero otherwise | |
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400 */ |
6095 | 401 int ff_rtsp_send_cmd_with_content(AVFormatContext *s, |
6096 | 402 const char *method, const char *url, |
403 const char *headers, | |
404 RTSPMessageHeader *reply, | |
405 unsigned char **content_ptr, | |
406 const unsigned char *send_content, | |
407 int send_content_length); | |
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408 |
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409 /** |
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410 * Send a command to the RTSP server and wait for the reply. |
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411 * |
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412 * @see rtsp_send_cmd_with_content |
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413 */ |
6095 | 414 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, |
6096 | 415 const char *url, const char *headers, |
416 RTSPMessageHeader *reply, unsigned char **content_ptr); | |
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417 |
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418 /** |
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419 * Read a RTSP message from the server, or prepare to read data |
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420 * packets if we're reading data interleaved over the TCP/RTSP |
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421 * connection as well. |
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422 * |
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423 * @param s RTSP (de)muxer context |
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424 * @param reply pointer where the RTSP message header will be stored |
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425 * @param content_ptr pointer where the RTSP message body, if any, will |
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426 * be stored (length is in reply) |
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427 * @param return_on_interleaved_data whether the function may return if we |
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428 * encounter a data marker ('$'), which precedes data |
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429 * packets over interleaved TCP/RTSP connections. If this |
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430 * is set, this function will return 1 after encountering |
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431 * a '$'. If it is not set, the function will skip any |
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432 * data packets (if they are encountered), until a reply |
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433 * has been fully parsed. If no more data is available |
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434 * without parsing a reply, it will return an error. |
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435 * |
5909 | 436 * @return 1 if a data packets is ready to be received, -1 on error, |
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437 * and 0 on success. |
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438 */ |
5697 | 439 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, |
5700 | 440 unsigned char **content_ptr, |
441 int return_on_interleaved_data); | |
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442 |
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443 /** |
5845 | 444 * Skip a RTP/TCP interleaved packet. |
445 */ | |
446 void ff_rtsp_skip_packet(AVFormatContext *s); | |
447 | |
448 /** | |
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449 * Connect to the RTSP server and set up the individual media streams. |
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450 * This can be used for both muxers and demuxers. |
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451 * |
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452 * @param s RTSP (de)muxer context |
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453 * |
5909 | 454 * @return 0 on success, < 0 on error. Cleans up all allocations done |
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455 * within the function on error. |
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456 */ |
5697 | 457 int ff_rtsp_connect(AVFormatContext *s); |
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458 |
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459 /** |
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460 * Close and free all streams within the RTSP (de)muxer |
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461 * |
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462 * @param s RTSP (de)muxer context |
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463 */ |
5697 | 464 void ff_rtsp_close_streams(AVFormatContext *s); |
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465 |
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466 /** |
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467 * Close all connection handles within the RTSP (de)muxer |
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468 * |
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469 * @param rt RTSP (de)muxer context |
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470 */ |
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471 void ff_rtsp_close_connections(AVFormatContext *rt); |
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472 |
5104 | 473 #endif /* AVFORMAT_RTSP_H */ |