annotate libaf/af.h @ 32797:18d7da3834d0

ffxandpcm works fine with the native avi demuxer.
author cehoyos
date Fri, 11 Feb 2011 12:39:46 +0000
parents 71a04d45357c
children a93891202051
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1 /*
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2 * This file is part of MPlayer.
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3 *
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4 * MPlayer is free software; you can redistribute it and/or modify
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5 * it under the terms of the GNU General Public License as published by
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6 * the Free Software Foundation; either version 2 of the License, or
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7 * (at your option) any later version.
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8 *
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9 * MPlayer is distributed in the hope that it will be useful,
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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12 * GNU General Public License for more details.
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13 *
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14 * You should have received a copy of the GNU General Public License along
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15 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
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16 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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17 */
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18
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19 #ifndef MPLAYER_AF_H
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20 #define MPLAYER_AF_H
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21
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22 #include <stdio.h>
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23
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24 #include "config.h"
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25
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26 #include "af_format.h"
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27 #include "control.h"
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28 #include "cpudetect.h"
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29 #include "mp_msg.h"
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30
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31 struct af_instance_s;
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32
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33 // Number of channels
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34 #ifndef AF_NCH
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35 #define AF_NCH 8
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36 #endif
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37
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38 // Audio data chunk
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39 typedef struct af_data_s
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40 {
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41 void* audio; // data buffer
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42 int len; // buffer length
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43 int rate; // sample rate
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44 int nch; // number of channels
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45 int format; // format
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46 int bps; // bytes per sample
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47 } af_data_t;
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48
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49
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50 // Flags used for defining the behavior of an audio filter
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51 #define AF_FLAGS_REENTRANT 0x00000000
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52 #define AF_FLAGS_NOT_REENTRANT 0x00000001
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53
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54 /* Audio filter information not specific for current instance, but for
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55 a specific filter */
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56 typedef struct af_info_s
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57 {
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58 const char *info;
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59 const char *name;
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60 const char *author;
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61 const char *comment;
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62 const int flags;
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63 int (*open)(struct af_instance_s* vf);
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64 } af_info_t;
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65
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66 // Linked list of audio filters
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67 typedef struct af_instance_s
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68 {
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69 const af_info_t* info;
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70 int (*control)(struct af_instance_s* af, int cmd, void* arg);
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71 void (*uninit)(struct af_instance_s* af);
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72 af_data_t* (*play)(struct af_instance_s* af, af_data_t* data);
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73 void* setup; // setup data for this specific instance and filter
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74 af_data_t* data; // configuration for outgoing data stream
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75 struct af_instance_s* next;
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76 struct af_instance_s* prev;
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77 double delay; /* Delay caused by the filter, in units of bytes read without
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78 * corresponding output */
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79 double mul; /* length multiplier: how much does this instance change
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80 the length of the buffer. */
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81 }af_instance_t;
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82
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83 // Initialization flags
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84 extern int* af_cpu_speed;
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85
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86 #define AF_INIT_AUTO 0x00000000
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87 #define AF_INIT_SLOW 0x00000001
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88 #define AF_INIT_FAST 0x00000002
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89 #define AF_INIT_FORCE 0x00000003
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90 #define AF_INIT_TYPE_MASK 0x00000003
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91
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92 #define AF_INIT_INT 0x00000000
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93 #define AF_INIT_FLOAT 0x00000004
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94 #define AF_INIT_FORMAT_MASK 0x00000004
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95
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96 // Default init type
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97 #ifndef AF_INIT_TYPE
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98 #define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_SLOW)
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99 #endif
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100
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101 // Configuration switches
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102 typedef struct af_cfg_s{
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103 int force; // Initialization type
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104 char** list; /* list of names of filters that are added to filter
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105 list during first initialization of stream */
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106 }af_cfg_t;
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107
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108 // Current audio stream
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109 typedef struct af_stream
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110 {
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111 // The first and last filter in the list
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112 af_instance_t* first;
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113 af_instance_t* last;
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114 // Storage for input and output data formats
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115 af_data_t input;
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116 af_data_t output;
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117 // Configuration for this stream
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118 af_cfg_t cfg;
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119 }af_stream_t;
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120
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121 /*********************************************
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122 // Return values
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123 */
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124
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125 #define AF_DETACH 2
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126 #define AF_OK 1
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127 #define AF_TRUE 1
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128 #define AF_FALSE 0
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129 #define AF_UNKNOWN -1
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130 #define AF_ERROR -2
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131 #define AF_FATAL -3
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132
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133
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134
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135 /*********************************************
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136 // Export functions
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137 */
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138
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139 /**
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140 * \defgroup af_chain Audio filter chain functions
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141 * \{
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142 * \param s filter chain
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143 */
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144
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145 /**
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146 * \brief Initialize the stream "s".
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147 * \return 0 on success, -1 on failure
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148 *
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149 * This function creates a new filter list if necessary, according
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150 * to the values set in input and output. Input and output should contain
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151 * the format of the current movie and the format of the preferred output
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152 * respectively.
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153 * Filters to convert to the preferred output format are inserted
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154 * automatically, except when they are set to 0.
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155 * The function is reentrant i.e. if called with an already initialized
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156 * stream the stream will be reinitialized.
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157 */
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158 int af_init(af_stream_t* s);
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159
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160 /**
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161 * \brief Uninit and remove all filters from audio filter chain
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162 */
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163 void af_uninit(af_stream_t* s);
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164
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165 /**
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166 * \brief Reinit the filter list from the given filter on downwards
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167 * \param Filter instance to begin the reinit from
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168 * \return AF_OK on success or AF_ERROR on failure
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169 */
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170 int af_reinit(af_stream_t* s, af_instance_t* af);
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171
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172 /**
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173 * \brief This function adds the filter "name" to the stream s.
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174 * \param name name of filter to add
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175 * \return pointer to the new filter, NULL if insert failed
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176 *
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177 * The filter will be inserted somewhere nice in the
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178 * list of filters (i.e. at the beginning unless the
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179 * first filter is the format filter (why??).
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180 */
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181 af_instance_t* af_add(af_stream_t* s, char* name);
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182
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183 /**
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184 * \brief Uninit and remove the filter "af"
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185 * \param af filter to remove
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186 */
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187 void af_remove(af_stream_t* s, af_instance_t* af);
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188
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189 /**
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190 * \brief find filter in chain by name
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191 * \param name name of the filter to find
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192 * \return first filter with right name or NULL if not found
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193 *
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194 * This function is used for finding already initialized filters
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195 */
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196 af_instance_t* af_get(af_stream_t* s, char* name);
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197
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198 /**
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199 * \brief filter data chunk through the filters in the list
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200 * \param data data to play
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201 * \return resulting data
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202 * \ingroup af_chain
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203 */
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204 af_data_t* af_play(af_stream_t* s, af_data_t* data);
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205
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206 /**
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207 * \brief send control to all filters, starting with the last until
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208 * one accepts the command with AF_OK.
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209 * \param cmd filter control command
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210 * \param arg argument for filter command
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211 * \return the accepting filter or NULL if none was found
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212 */
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213 af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg);
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214
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215 /**
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216 * \brief calculate average ratio of filter output lenth to input length
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217 * \return the ratio
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218 */
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219 double af_calc_filter_multiplier(af_stream_t* s);
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220
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221 /**
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222 * \brief Calculate the total delay caused by the filters
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223 * \return delay in bytes of "missing" output
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224 */
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225 double af_calc_delay(af_stream_t* s);
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226
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227 /** \} */ // end of af_chain group
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228
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229 // Helper functions and macros used inside the audio filters
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230
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231 /**
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232 * \defgroup af_filter Audio filter helper functions
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233 * \{
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234 */
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235
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236 /* Helper function called by the macro with the same name only to be
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237 called from inside filters */
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238 int af_resize_local_buffer(af_instance_t* af, af_data_t* data);
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239
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240 /* Helper function used to calculate the exact buffer length needed
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241 when buffers are resized. The returned length is >= than what is
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242 needed */
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243 int af_lencalc(double mul, af_data_t* data);
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244
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245 /**
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246 * \brief convert dB to gain value
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247 * \param n number of values to convert
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248 * \param in [in] values in dB, <= -200 will become 0 gain
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249 * \param out [out] gain values
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250 * \param k input values are divided by this
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251 * \param mi minimum dB value, input will be clamped to this
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252 * \param ma maximum dB value, input will be clamped to this
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253 * \return AF_ERROR on error, AF_OK otherwise
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254 */
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255 int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
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256
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257 /**
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258 * \brief convert gain value to dB
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259 * \param n number of values to convert
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260 * \param in [in] gain values, 0 wil become -200 dB
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261 * \param out [out] values in dB
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262 * \param k output values will be multiplied by this
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263 * \return AF_ERROR on error, AF_OK otherwise
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264 */
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265 int af_to_dB(int n, float* in, float* out, float k);
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266
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267 /**
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268 * \brief convert milliseconds to sample time
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269 * \param n number of values to convert
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270 * \param in [in] values in milliseconds
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271 * \param out [out] sample time values
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272 * \param rate sample rate
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273 * \param mi minimum ms value, input will be clamped to this
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274 * \param ma maximum ms value, input will be clamped to this
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275 * \return AF_ERROR on error, AF_OK otherwise
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276 */
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277 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
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278
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279 /**
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280 * \brief convert sample time to milliseconds
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281 * \param n number of values to convert
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282 * \param in [in] sample time values
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283 * \param out [out] values in milliseconds
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284 * \param rate sample rate
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285 * \return AF_ERROR on error, AF_OK otherwise
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286 */
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287 int af_to_ms(int n, int* in, float* out, int rate);
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288
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289 /**
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290 * \brief test if output format matches
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291 * \param af audio filter
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292 * \param out needed format, will be overwritten by available
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293 * format if they do not match
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294 * \return AF_FALSE if formats do not match, AF_OK if they match
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295 *
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296 * compares the format, bps, rate and nch values of af->data with out
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297 */
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298 int af_test_output(struct af_instance_s* af, af_data_t* out);
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299
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300 /**
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301 * \brief soft clipping function using sin()
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302 * \param a input value
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303 * \return clipped value
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304 */
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305 float af_softclip(float a);
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306
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307 /** \} */ // end of af_filter group, but more functions of this group below
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308
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309 /** Print a list of all available audio filters */
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310 void af_help(void);
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311
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312 /**
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313 * \brief fill the missing parameters in the af_data_t structure
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314 * \param data structure to fill
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315 * \ingroup af_filter
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316 *
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317 * Currently only sets bps based on format
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318 */
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319 void af_fix_parameters(af_data_t *data);
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320
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321 /** Memory reallocation macro: if a local buffer is used (i.e. if the
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322 filter doesn't operate on the incoming buffer this macro must be
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323 called to ensure the buffer is big enough.
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324 * \ingroup af_filter
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325 */
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326 #define RESIZE_LOCAL_BUFFER(a,d)\
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327 ((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK)
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328
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329 /* Some other useful macro definitions*/
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330 #ifndef min
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331 #define min(a,b)(((a)>(b))?(b):(a))
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332 #endif
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333
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334 #ifndef max
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335 #define max(a,b)(((a)>(b))?(a):(b))
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336 #endif
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337
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338 #ifndef clamp
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339 #define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
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340 #endif
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341
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342 #ifndef sign
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343 #define sign(a) (((a)>0)?(1):(-1))
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344 #endif
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345
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346 #ifndef lrnd
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347 #define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
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348 #endif
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349
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350 #endif /* MPLAYER_AF_H */