Mercurial > mplayer.hg
annotate libaf/af.h @ 30935:59f25d383d12
rename eatgv -> ffeatgv, lpcm -> fflpcm
for consitently naming all ffmpeg codecs
requested by compn
author | attila |
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date | Wed, 31 Mar 2010 06:33:08 +0000 |
parents | e75026202313 |
children | 0d18ce469b67 |
rev | line source |
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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 |
26029 | 19 #ifndef MPLAYER_AF_H |
20 #define MPLAYER_AF_H | |
12676 | 21 |
8167 | 22 #include <stdio.h> |
23 | |
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24 #include "config.h" |
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25 |
29051 | 26 #include "af_format.h" |
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27 #include "control.h" |
29051 | 28 #include "cpudetect.h" |
29049 | 29 #include "mp_msg.h" |
29050 | 30 |
7568 | 31 struct af_instance_s; |
32 | |
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33 // Number of channels |
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34 #ifndef AF_NCH |
29826 | 35 #define AF_NCH 8 |
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36 #endif |
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37 |
7568 | 38 // Audio data chunk |
39 typedef struct af_data_s | |
40 { | |
41 void* audio; // data buffer | |
42 int len; // buffer length | |
43 int rate; // sample rate | |
44 int nch; // number of channels | |
45 int format; // format | |
46 int bps; // bytes per sample | |
47 } af_data_t; | |
48 | |
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49 |
8167 | 50 // Flags used for defining the behavior of an audio filter |
7615 | 51 #define AF_FLAGS_REENTRANT 0x00000000 |
52 #define AF_FLAGS_NOT_REENTRANT 0x00000001 | |
53 | |
7568 | 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 |
7568 | 57 { |
58 const char *info; | |
59 const char *name; | |
60 const char *author; | |
61 const char *comment; | |
7615 | 62 const int flags; |
7568 | 63 int (*open)(struct af_instance_s* vf); |
64 } af_info_t; | |
65 | |
66 // Linked list of audio filters | |
67 typedef struct af_instance_s | |
68 { | |
69 af_info_t* info; | |
70 int (*control)(struct af_instance_s* af, int cmd, void* arg); | |
71 void (*uninit)(struct af_instance_s* af); | |
72 af_data_t* (*play)(struct af_instance_s* af, af_data_t* data); | |
73 void* setup; // setup data for this specific instance and filter | |
74 af_data_t* data; // configuration for outgoing data stream | |
75 struct af_instance_s* next; | |
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76 struct af_instance_s* prev; |
24900 | 77 double delay; /* Delay caused by the filter, in units of bytes read without |
78 * corresponding output */ | |
24888 | 79 double mul; /* length multiplier: how much does this instance change |
7568 | 80 the length of the buffer. */ |
81 }af_instance_t; | |
82 | |
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83 // Initialization flags |
8167 | 84 extern int* af_cpu_speed; |
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 |
8167 | 97 #ifndef AF_INIT_TYPE |
98 #define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_SLOW) | |
99 #endif | |
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 |
8167 | 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_s |
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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; |
8167 | 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 |
7568 | 121 /********************************************* |
122 // Return values | |
123 */ | |
124 | |
125 #define AF_DETACH 2 | |
126 #define AF_OK 1 | |
127 #define AF_TRUE 1 | |
128 #define AF_FALSE 0 | |
129 #define AF_UNKNOWN -1 | |
130 #define AF_ERROR -2 | |
8167 | 131 #define AF_FATAL -3 |
7568 | 132 |
133 | |
134 | |
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135 /********************************************* |
7568 | 136 // Export functions |
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137 */ |
7568 | 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 This function adds the filter "name" to the stream s. |
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167 * \param name name of filter to add |
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168 * \return pointer to the new filter, NULL if insert failed |
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169 * |
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170 * The filter will be inserted somewhere nice in the |
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171 * list of filters (i.e. at the beginning unless the |
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172 * first filter is the format filter (why??). |
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173 */ |
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174 af_instance_t* af_add(af_stream_t* s, char* name); |
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175 |
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176 /** |
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177 * \brief Uninit and remove the filter "af" |
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178 * \param af filter to remove |
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179 */ |
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180 void af_remove(af_stream_t* s, af_instance_t* af); |
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181 |
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182 /** |
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183 * \brief find filter in chain by name |
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184 * \param name name of the filter to find |
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185 * \return first filter with right name or NULL if not found |
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186 * |
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187 * This function is used for finding already initialized filters |
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188 */ |
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189 af_instance_t* af_get(af_stream_t* s, char* name); |
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190 |
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191 /** |
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192 * \brief filter data chunk through the filters in the list |
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193 * \param data data to play |
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194 * \return resulting data |
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195 * \ingroup af_chain |
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196 */ |
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197 af_data_t* af_play(af_stream_t* s, af_data_t* data); |
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198 |
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199 /** |
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200 * \brief send control to all filters, starting with the last until |
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201 * one accepts the command with AF_OK. |
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202 * \param cmd filter control command |
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203 * \param arg argument for filter command |
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204 * \return the accepting filter or NULL if none was found |
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205 */ |
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206 af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg); |
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207 |
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208 /** |
24892 | 209 * \brief calculate average ratio of filter output lenth to input length |
210 * \return the ratio | |
211 */ | |
212 double af_calc_filter_multiplier(af_stream_t* s); | |
7568 | 213 |
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214 /** |
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215 * \brief Calculate the total delay caused by the filters |
24900 | 216 * \return delay in bytes of "missing" output |
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217 */ |
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218 double af_calc_delay(af_stream_t* s); |
7568 | 219 |
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220 /** \} */ // end of af_chain group |
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221 |
7568 | 222 // Helper functions and macros used inside the audio filters |
223 | |
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224 /** |
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225 * \defgroup af_filter Audio filter helper functions |
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226 * \{ |
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227 */ |
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228 |
7568 | 229 /* Helper function called by the macro with the same name only to be |
230 called from inside filters */ | |
231 int af_resize_local_buffer(af_instance_t* af, af_data_t* data); | |
232 | |
233 /* Helper function used to calculate the exact buffer length needed | |
7589 | 234 when buffers are resized. The returned length is >= than what is |
235 needed */ | |
24888 | 236 int af_lencalc(double mul, af_data_t* data); |
7568 | 237 |
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238 /** |
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239 * \brief convert dB to gain value |
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240 * \param n number of values to convert |
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241 * \param in [in] values in dB, <= -200 will become 0 gain |
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242 * \param out [out] gain values |
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243 * \param k input values are divided by this |
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244 * \param mi minimum dB value, input will be clamped to this |
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245 * \param ma maximum dB value, input will be clamped to this |
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246 * \return AF_ERROR on error, AF_OK otherwise |
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247 */ |
8607 | 248 int af_from_dB(int n, float* in, float* out, float k, float mi, float ma); |
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249 |
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250 /** |
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251 * \brief convert gain value to dB |
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252 * \param n number of values to convert |
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253 * \param in [in] gain values, 0 wil become -200 dB |
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254 * \param out [out] values in dB |
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255 * \param k output values will be multiplied by this |
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256 * \return AF_ERROR on error, AF_OK otherwise |
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257 */ |
8607 | 258 int af_to_dB(int n, float* in, float* out, float k); |
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259 |
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260 /** |
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261 * \brief convert milliseconds to sample time |
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262 * \param n number of values to convert |
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263 * \param in [in] values in milliseconds |
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264 * \param out [out] sample time values |
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265 * \param rate sample rate |
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266 * \param mi minimum ms value, input will be clamped to this |
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267 * \param ma maximum ms value, input will be clamped to this |
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268 * \return AF_ERROR on error, AF_OK otherwise |
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269 */ |
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270 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma); |
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271 |
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272 /** |
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273 * \brief convert sample time to milliseconds |
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274 * \param n number of values to convert |
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275 * \param in [in] sample time values |
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276 * \param out [out] values in milliseconds |
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277 * \param rate sample rate |
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278 * \return AF_ERROR on error, AF_OK otherwise |
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279 */ |
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280 int af_to_ms(int n, int* in, float* out, int rate); |
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281 |
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282 /** |
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283 * \brief test if output format matches |
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284 * \param af audio filter |
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285 * \param out needed format, will be overwritten by available |
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286 * format if they do not match |
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287 * \return AF_FALSE if formats do not match, AF_OK if they match |
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288 * |
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289 * compares the format, bps, rate and nch values of af->data with out |
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290 */ |
8607 | 291 int af_test_output(struct af_instance_s* af, af_data_t* out); |
292 | |
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293 /** |
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294 * \brief soft clipping function using sin() |
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295 * \param a input value |
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296 * \return clipped value |
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297 */ |
14622 | 298 float af_softclip(float a); |
299 | |
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300 /** \} */ // end of af_filter group, but more functions of this group below |
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301 |
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302 /** Print a list of all available audio filters */ |
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303 void af_help(void); |
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304 |
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305 /** |
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306 * \brief fill the missing parameters in the af_data_t structure |
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307 * \param data structure to fill |
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308 * \ingroup af_filter |
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309 * |
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310 * Currently only sets bps based on format |
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311 */ |
14883 | 312 void af_fix_parameters(af_data_t *data); |
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313 |
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314 /** Memory reallocation macro: if a local buffer is used (i.e. if the |
7568 | 315 filter doesn't operate on the incoming buffer this macro must be |
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316 called to ensure the buffer is big enough. |
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317 * \ingroup af_filter |
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318 */ |
7568 | 319 #define RESIZE_LOCAL_BUFFER(a,d)\ |
7591 | 320 ((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK) |
7568 | 321 |
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322 /* Some other useful macro definitions*/ |
7568 | 323 #ifndef min |
324 #define min(a,b)(((a)>(b))?(b):(a)) | |
325 #endif | |
326 | |
327 #ifndef max | |
328 #define max(a,b)(((a)>(b))?(a):(b)) | |
329 #endif | |
330 | |
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331 #ifndef clamp |
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332 #define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a))) |
7568 | 333 #endif |
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334 |
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335 #ifndef sign |
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336 #define sign(a) (((a)>0)?(1):(-1)) |
8073 | 337 #endif |
338 | |
8167 | 339 #ifndef lrnd |
340 #define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5)) | |
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341 #endif |
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342 |
26029 | 343 #endif /* MPLAYER_AF_H */ |