annotate rtsp.h @ 6219:a036426dc8e6 libavformat

Remove angular brackets from Doxygen comments; Doxygen confuses them for HTML.
author diego
date Fri, 02 Jul 2010 12:48:32 +0000
parents 736165b749f8
children c709176da1ad
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1 /*
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2 * RTSP definitions
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3 * Copyright (c) 2002 Fabrice Bellard
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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
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8 * modify it under the terms of the GNU Lesser General Public
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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.
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11 *
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12 * FFmpeg is distributed in the hope that it will be useful,
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 * Lesser General Public License for more details.
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16 *
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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
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20 */
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21 #ifndef AVFORMAT_RTSP_H
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22 #define AVFORMAT_RTSP_H
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23
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24 #include <stdint.h>
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25 #include "avformat.h"
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26 #include "rtspcodes.h"
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27 #include "rtpdec.h"
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28 #include "network.h"
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29 #include "httpauth.h"
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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
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39 };
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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
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52 /**
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53 * Transport mode for the RTSP data. This may be plain, or
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54 * tunneled, which is done over HTTP.
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55 */
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56 enum RTSPControlTransport {
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57 RTSP_MODE_PLAIN, /**< Normal RTSP */
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58 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
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59 };
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60
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61 #define RTSP_DEFAULT_PORT 554
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62 #define RTSP_MAX_TRANSPORTS 8
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63 #define RTSP_TCP_MAX_PACKET_SIZE 1472
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64 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
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65 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
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66 #define RTSP_RTP_PORT_MIN 5000
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67 #define RTSP_RTP_PORT_MAX 10000
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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 */
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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
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99 uint32_t 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;
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106 } RTSPTransportField;
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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 */
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113 int content_length;
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114
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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 */
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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 */
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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 */
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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. */
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132 char session_id[512];
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133
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134 /** the "Location:" field. This value is used to handle redirection.
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135 */
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136 char location[4096];
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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;
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162 } RTSPMessageHeader;
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163
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164 /**
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165 * Client state, i.e. whether we are currently receiving data (PLAYING) or
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166 * setup-but-not-receiving (PAUSED). State can be changed in applications
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167 * by calling av_read_play/pause().
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168 */
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169 enum RTSPClientState {
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170 RTSP_STATE_IDLE, /**< not initialized */
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171 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
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172 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
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173 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
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174 };
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175
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176 /**
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177 * Identifies particular servers that require special handling, such as
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178 * standards-incompliant "Transport:" lines in the SETUP request.
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179 */
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180 enum RTSPServerType {
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181 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
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182 RTSP_SERVER_REAL, /**< Realmedia-style server */
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183 RTSP_SERVER_WMS, /**< Windows Media server */
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184 RTSP_SERVER_NB
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185 };
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186
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187 /**
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188 * Private data for the RTSP demuxer.
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189 *
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190 * @todo Use ByteIOContext instead of URLContext
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191 */
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192 typedef struct RTSPState {
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193 URLContext *rtsp_hd; /* RTSP TCP connection handle */
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194
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195 /** number of items in the 'rtsp_streams' variable */
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196 int nb_rtsp_streams;
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197
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198 struct RTSPStream **rtsp_streams; /**< streams in this session */
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199
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200 /** indicator of whether we are currently receiving data from the
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201 * server. Basically this isn't more than a simple cache of the
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202 * last PLAY/PAUSE command sent to the server, to make sure we don't
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203 * send 2x the same unexpectedly or commands in the wrong state. */
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204 enum RTSPClientState state;
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205
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206 /** the seek value requested when calling av_seek_frame(). This value
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207 * is subsequently used as part of the "Range" parameter when emitting
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208 * the RTSP PLAY command. If we are currently playing, this command is
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209 * called instantly. If we are currently paused, this command is called
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210 * whenever we resume playback. Either way, the value is only used once,
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211 * see rtsp_read_play() and rtsp_read_seek(). */
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212 int64_t seek_timestamp;
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213
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214 /* XXX: currently we use unbuffered input */
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215 // ByteIOContext rtsp_gb;
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216
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217 int seq; /**< RTSP command sequence number */
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218
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219 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
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220 * identifier that the client should re-transmit in each RTSP command */
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221 char session_id[512];
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222
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223 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
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224 * the server will go without traffic on the RTSP/TCP line before it
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225 * closes the connection. */
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226 int timeout;
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227
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228 /** timestamp of the last RTSP command that we sent to the RTSP server.
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229 * This is used to calculate when to send dummy commands to keep the
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230 * connection alive, in conjunction with timeout. */
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231 int64_t last_cmd_time;
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232
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233 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
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234 enum RTSPTransport transport;
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235
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236 /** the negotiated network layer transport protocol; e.g. TCP or UDP
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237 * uni-/multicast */
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238 enum RTSPLowerTransport lower_transport;
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239
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240 /** brand of server that we're talking to; e.g. WMS, REAL or other.
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241 * Detected based on the value of RTSPMessageHeader->server or the presence
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242 * of RTSPMessageHeader->real_challenge */
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243 enum RTSPServerType server_type;
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244
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245 /** plaintext authorization line (username:password) */
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246 char auth[128];
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247
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248 /** authentication state */
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249 HTTPAuthState auth_state;
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250
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251 /** The last reply of the server to a RTSP command */
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252 char last_reply[2048]; /* XXX: allocate ? */
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253
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254 /** RTSPStream->transport_priv of the last stream that we read a
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255 * packet from */
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256 void *cur_transport_priv;
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257
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258 /** The following are used for Real stream selection */
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259 //@{
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260 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
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261 int need_subscription;
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262
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263 /** stream setup during the last frame read. This is used to detect if
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264 * we need to subscribe or unsubscribe to any new streams. */
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265 enum AVDiscard real_setup_cache[MAX_STREAMS];
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266
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267 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
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268 * this is used to send the same "Unsubscribe:" if stream setup changed,
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269 * before sending a new "Subscribe:" command. */
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270 char last_subscription[1024];
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271 //@}
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272
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273 /** The following are used for RTP/ASF streams */
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274 //@{
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275 /** ASF demuxer context for the embedded ASF stream from WMS servers */
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276 AVFormatContext *asf_ctx;
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277
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278 /** cache for position of the asf demuxer, since we load a new
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279 * data packet in the bytecontext for each incoming RTSP packet. */
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280 uint64_t asf_pb_pos;
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281 //@}
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282
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283 /** some MS RTSP streams contain a URL in the SDP that we need to use
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284 * for all subsequent RTSP requests, rather than the input URI; in
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285 * other cases, this is a copy of AVFormatContext->filename. */
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286 char control_uri[1024];
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287
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288 /** The synchronized start time of the output streams. */
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289 int64_t start_time;
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290
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291 /** Additional output handle, used when input and output are done
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292 * separately, eg for HTTP tunneling. */
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293 URLContext *rtsp_hd_out;
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294
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295 /** RTSP transport mode, such as plain or tunneled. */
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296 enum RTSPControlTransport control_transport;
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297 } RTSPState;
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298
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299 /**
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300 * Describes a single stream, as identified by a single m= line block in the
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301 * SDP content. In the case of RDT, one RTSPStream can represent multiple
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302 * AVStreams. In this case, each AVStream in this set has similar content
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303 * (but different codec/bitrate).
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304 */
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305 typedef struct RTSPStream {
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306 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
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307 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
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308
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309 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
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310 int stream_index;
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311
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312 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
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313 * for the selected transport. Only used for TCP. */
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314 int interleaved_min, interleaved_max;
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315
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316 char control_url[1024]; /**< url for this stream (from SDP) */
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317
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318 /** The following are used only in SDP, not RTSP */
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319 //@{
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320 int sdp_port; /**< port (from SDP content) */
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321 struct in_addr sdp_ip; /**< IP address (from SDP content) */
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322 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
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323 int sdp_payload_type; /**< payload type */
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324 //@}
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325
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326 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
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327 //@{
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328 /** handler structure */
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329 RTPDynamicProtocolHandler *dynamic_handler;
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330
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331 /** private data associated with the dynamic protocol */
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332 PayloadContext *dynamic_protocol_context;
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333 //@}
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334 } RTSPStream;
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335
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336 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
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337 HTTPAuthState *auth_state);
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338
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339 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
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340 extern int rtsp_default_protocols;
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341 #endif
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342 extern int rtsp_rtp_port_min;
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343 extern int rtsp_rtp_port_max;
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344
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345 /**
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346 * Send a command to the RTSP server without waiting for the reply.
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347 *
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348 * @param s RTSP (de)muxer context
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349 * @param method the method for the request
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350 * @param url the target url for the request
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351 * @param headers extra header lines to include in the request
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352 * @param send_content if non-null, the data to send as request body content
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353 * @param send_content_length the length of the send_content data, or 0 if
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354 * send_content is null
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355 *
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356 * @return zero if success, nonzero otherwise
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357 */
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358 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
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359 const char *method, const char *url,
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360 const char *headers,
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361 const unsigned char *send_content,
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362 int send_content_length);
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363 /**
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364 * Send a command to the RTSP server without waiting for the reply.
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365 *
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366 * @see rtsp_send_cmd_with_content_async
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367 */
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368 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
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369 const char *url, const char *headers);
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370
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371 /**
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372 * Send a command to the RTSP server and wait for the reply.
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373 *
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374 * @param s RTSP (de)muxer context
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375 * @param method the method for the request
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376 * @param url the target url for the request
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377 * @param headers extra header lines to include in the request
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378 * @param reply pointer where the RTSP message header will be stored
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379 * @param content_ptr pointer where the RTSP message body, if any, will
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380 * be stored (length is in reply)
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381 * @param send_content if non-null, the data to send as request body content
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382 * @param send_content_length the length of the send_content data, or 0 if
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383 * send_content is null
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384 *
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385 * @return zero if success, nonzero otherwise
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386 */
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387 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
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388 const char *method, const char *url,
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389 const char *headers,
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390 RTSPMessageHeader *reply,
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391 unsigned char **content_ptr,
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392 const unsigned char *send_content,
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393 int send_content_length);
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394
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395 /**
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396 * Send a command to the RTSP server and wait for the reply.
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397 *
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398 * @see rtsp_send_cmd_with_content
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399 */
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400 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
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401 const char *url, const char *headers,
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402 RTSPMessageHeader *reply, unsigned char **content_ptr);
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403
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404 /**
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405 * Read a RTSP message from the server, or prepare to read data
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406 * packets if we're reading data interleaved over the TCP/RTSP
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407 * connection as well.
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408 *
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409 * @param s RTSP (de)muxer context
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410 * @param reply pointer where the RTSP message header will be stored
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411 * @param content_ptr pointer where the RTSP message body, if any, will
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412 * be stored (length is in reply)
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413 * @param return_on_interleaved_data whether the function may return if we
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414 * encounter a data marker ('$'), which precedes data
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415 * packets over interleaved TCP/RTSP connections. If this
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416 * is set, this function will return 1 after encountering
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417 * a '$'. If it is not set, the function will skip any
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418 * data packets (if they are encountered), until a reply
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419 * has been fully parsed. If no more data is available
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420 * without parsing a reply, it will return an error.
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421 *
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422 * @return 1 if a data packets is ready to be received, -1 on error,
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423 * and 0 on success.
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424 */
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425 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
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426 unsigned char **content_ptr,
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427 int return_on_interleaved_data);
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428
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429 /**
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430 * Skip a RTP/TCP interleaved packet.
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431 */
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432 void ff_rtsp_skip_packet(AVFormatContext *s);
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433
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434 /**
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435 * Connect to the RTSP server and set up the individual media streams.
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mstorsjo
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436 * This can be used for both muxers and demuxers.
87b26ad577c5 Add declarations and doxygen documentation of generic rtsp support functions
mstorsjo
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437 *
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mstorsjo
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438 * @param s RTSP (de)muxer context
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mstorsjo
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439 *
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b8041f85c327 Replace @returns by @return.
benoit
parents: 5890
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440 * @return 0 on success, < 0 on error. Cleans up all allocations done
5692
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mstorsjo
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441 * within the function on error.
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442 */
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rbultje
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443 int ff_rtsp_connect(AVFormatContext *s);
5692
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mstorsjo
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444
87b26ad577c5 Add declarations and doxygen documentation of generic rtsp support functions
mstorsjo
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445 /**
87b26ad577c5 Add declarations and doxygen documentation of generic rtsp support functions
mstorsjo
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446 * Close and free all streams within the RTSP (de)muxer
87b26ad577c5 Add declarations and doxygen documentation of generic rtsp support functions
mstorsjo
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447 *
87b26ad577c5 Add declarations and doxygen documentation of generic rtsp support functions
mstorsjo
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448 * @param s RTSP (de)muxer context
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449 */
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rbultje
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450 void ff_rtsp_close_streams(AVFormatContext *s);
5692
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mstorsjo
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451
6093
e688babd14df RTSP: Add a second URLContext for outgoing messages
mstorsjo
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452 /**
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mstorsjo
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453 * Close all connection handles within the RTSP (de)muxer
e688babd14df RTSP: Add a second URLContext for outgoing messages
mstorsjo
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454 *
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mstorsjo
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455 * @param rt RTSP (de)muxer context
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mstorsjo
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456 */
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mstorsjo
parents: 6070
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457 void ff_rtsp_close_connections(AVFormatContext *rt);
e688babd14df RTSP: Add a second URLContext for outgoing messages
mstorsjo
parents: 6070
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458
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4c39baa3dfbf Use globally consistent include guard names.
stefano
parents: 5013
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459 #endif /* AVFORMAT_RTSP_H */