annotate rtsp.h @ 6435:67433b0c29d5 libavformat

move h264 demuxer to its own file
author aurel
date Sun, 29 Aug 2010 21:28:51 +0000
parents 2685bf2b0b78
children 2e3e76e8ada4
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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 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;
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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
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163 /** The "reason" is meant to specify better the meaning of the error code
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164 * returned
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165 */
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166 char reason[256];
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167 } RTSPMessageHeader;
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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 */
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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 */
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187 RTSP_SERVER_REAL, /**< Realmedia-style server */
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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. */
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270 enum AVDiscard *real_setup_cache;
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271
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272 /** current stream setup. This is a temporary buffer used to compare
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273 * current setup to previous frame setup. */
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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
016f3d4fe56d Use the control URI from the SDP (if present) rather than the input filename,
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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;
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303
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304 /** RTSP transport mode, such as plain or tunneled. */
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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) */
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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
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350 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
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351 HTTPAuthState *auth_state);
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352
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353 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
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354 extern int rtsp_default_protocols;
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355 #endif
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356 extern int rtsp_rtp_port_min;
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357 extern int rtsp_rtp_port_max;
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43ac45200138 suppressed RTSP abort hack - added PLAY/PAUSE
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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
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369 *
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370 * @return zero if success, nonzero otherwise
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371 */
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372 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
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373 const char *method, const char *url,
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374 const char *headers,
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375 const unsigned char *send_content,
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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 */
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382 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
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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
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398 *
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399 * @return zero if success, nonzero otherwise
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400 */
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401 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
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402 const char *method, const char *url,
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403 const char *headers,
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404 RTSPMessageHeader *reply,
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405 unsigned char **content_ptr,
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406 const unsigned char *send_content,
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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 */
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414 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
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415 const char *url, const char *headers,
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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 *
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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 */
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rbultje
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439 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
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440 unsigned char **content_ptr,
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441 int return_on_interleaved_data);
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442
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443 /**
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mstorsjo
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444 * Skip a RTP/TCP interleaved packet.
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445 */
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446 void ff_rtsp_skip_packet(AVFormatContext *s);
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447
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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 *
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benoit
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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 */
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rbultje
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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 */
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rbultje
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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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mstorsjo
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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
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stefano
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473 #endif /* AVFORMAT_RTSP_H */