annotate libaf/af_volnorm.c @ 14104:0e334ea33ebf

MPlayer-translations mailing list created.
author diego
date Sun, 05 Dec 2004 13:07:57 +0000
parents 14090f7300a8
children 3c56b18bbb0c
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1 /*=============================================================================
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2 //
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14090f7300a8 The full name of the GPL is GNU General Public License.
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3 // This software has been released under the terms of the GNU General Public
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4 // license. See http://www.gnu.org/copyleft/gpl.html for details.
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5 //
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6 // Copyright 2004 Alex Beregszaszi & Pierre Lombard
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7 //
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8 //=============================================================================
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9 */
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10
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11 #include <stdio.h>
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12 #include <stdlib.h>
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13 #include <string.h>
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14
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15 #include <unistd.h>
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16 #include <inttypes.h>
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17 #include <math.h>
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18 #include <limits.h>
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19
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20 #include "af.h"
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21
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22 // Methods:
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23 // 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1)
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24 // 2: uses several samples to smooth the variations (standard weighted mean
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25 // on past samples)
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26
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27 // Size of the memory array
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28 // FIXME: should depend on the frequency of the data (should be a few seconds)
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29 #define NSAMPLES 128
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30
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31 // If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we
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32 // choose to ignore the computed value as it's not significant enough
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33 // FIXME: should depend on the frequency of the data (0.5s maybe)
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34 #define MIN_SAMPLE_SIZE 32000
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35
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36 // mul is the value by which the samples are scaled
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37 // and has to be in [MUL_MIN, MUL_MAX]
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38 #define MUL_INIT 1.0
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39 #define MUL_MIN 0.1
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40 #define MUL_MAX 5.0
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41 // "Ideal" level
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42 #define MID_S16 (SHRT_MAX * 0.25)
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43 #define MID_FLOAT (INT_MAX * 0.25)
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44
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45 // Silence level
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46 // FIXME: should be relative to the level of the samples
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47 #define SIL_S16 (SHRT_MAX * 0.01)
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48 #define SIL_FLOAT (INT_MAX * 0.01) // FIXME
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49
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50 // smooth must be in ]0.0, 1.0[
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51 #define SMOOTH_MUL 0.06
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52 #define SMOOTH_LASTAVG 0.06
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53
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54 // Data for specific instances of this filter
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55 typedef struct af_volume_s
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56 {
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57 int method; // method used
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58 float mul;
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59 // method 1
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60 float lastavg; // history value of the filter
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61 // method 2
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62 int idx;
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63 struct {
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64 float avg; // average level of the sample
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65 int len; // sample size (weight)
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66 } mem[NSAMPLES];
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67 }af_volnorm_t;
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68
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69 // Initialization and runtime control
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70 static int control(struct af_instance_s* af, int cmd, void* arg)
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71 {
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72 af_volnorm_t* s = (af_volnorm_t*)af->setup;
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73
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74 switch(cmd){
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75 case AF_CONTROL_REINIT:
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76 // Sanity check
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77 if(!arg) return AF_ERROR;
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78
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79 af->data->rate = ((af_data_t*)arg)->rate;
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80 af->data->nch = ((af_data_t*)arg)->nch;
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81
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82 if(((af_data_t*)arg)->format != (AF_FORMAT_F | AF_FORMAT_NE) &&
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83 ((af_data_t*)arg)->format != (AF_FORMAT_SI | AF_FORMAT_NE))
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84 return AF_ERROR;
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85
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86 if(((af_data_t*)arg)->format == (AF_FORMAT_SI | AF_FORMAT_NE)){
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87 af->data->format = AF_FORMAT_SI | AF_FORMAT_NE;
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88 af->data->bps = 2;
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89 }else{
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90 af->data->format = AF_FORMAT_F | AF_FORMAT_NE;
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91 af->data->bps = 4;
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92 }
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93 return af_test_output(af,(af_data_t*)arg);
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94 case AF_CONTROL_COMMAND_LINE:{
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95 int i;
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96 sscanf((char*)arg,"%d", &i);
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97 if (i != 1 && i != 2)
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98 return AF_ERROR;
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99 s->method = i-1;
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100 return AF_OK;
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101 }
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102 }
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103 return AF_UNKNOWN;
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104 }
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105
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106 // Deallocate memory
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107 static void uninit(struct af_instance_s* af)
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108 {
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109 if(af->data)
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110 free(af->data);
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111 if(af->setup)
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112 free(af->setup);
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113 }
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114
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115 static void method1_int16(af_volnorm_t *s, af_data_t *c)
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116 {
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117 register int i = 0;
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118 int16_t *data = (int16_t*)c->audio; // Audio data
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119 int len = c->len/2; // Number of samples
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120 float curavg = 0.0, newavg, neededmul;
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121 int tmp;
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122
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123 for (i = 0; i < len; i++)
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124 {
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125 tmp = data[i];
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126 curavg += tmp * tmp;
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127 }
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128 curavg = sqrt(curavg / (float) len);
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129
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130 // Evaluate an adequate 'mul' coefficient based on previous state, current
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131 // samples level, etc
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132
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133 if (curavg > SIL_S16)
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134 {
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135 neededmul = MID_S16 / (curavg * s->mul);
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136 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
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137
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138 // clamp the mul coefficient
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139 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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140 }
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141
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142 // Scale & clamp the samples
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143 for (i = 0; i < len; i++)
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144 {
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145 tmp = s->mul * data[i];
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146 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
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147 data[i] = tmp;
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148 }
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149
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150 // Evaulation of newavg (not 100% accurate because of values clamping)
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151 newavg = s->mul * curavg;
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152
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153 // Stores computed values for future smoothing
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154 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
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155 }
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156
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157 static void method1_float(af_volnorm_t *s, af_data_t *c)
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158 {
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159 register int i = 0;
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160 float *data = (float*)c->audio; // Audio data
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161 int len = c->len/4; // Number of samples
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162 float curavg = 0.0, newavg, neededmul, tmp;
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163
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164 for (i = 0; i < len; i++)
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165 {
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166 tmp = data[i];
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167 curavg += tmp * tmp;
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168 }
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169 curavg = sqrt(curavg / (float) len);
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170
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171 // Evaluate an adequate 'mul' coefficient based on previous state, current
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172 // samples level, etc
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173
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174 if (curavg > SIL_FLOAT) // FIXME
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175 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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176 neededmul = MID_FLOAT / (curavg * s->mul);
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177 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
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178
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179 // clamp the mul coefficient
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180 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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181 }
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182
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183 // Scale & clamp the samples
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184 for (i = 0; i < len; i++)
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185 data[i] *= s->mul;
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186
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187 // Evaulation of newavg (not 100% accurate because of values clamping)
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188 newavg = s->mul * curavg;
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189
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190 // Stores computed values for future smoothing
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191 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
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192 }
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193
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194 static void method2_int16(af_volnorm_t *s, af_data_t *c)
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195 {
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196 register int i = 0;
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197 int16_t *data = (int16_t*)c->audio; // Audio data
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198 int len = c->len/2; // Number of samples
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199 float curavg = 0.0, newavg, avg = 0.0;
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200 int tmp, totallen = 0;
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201
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202 for (i = 0; i < len; i++)
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203 {
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204 tmp = data[i];
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205 curavg += tmp * tmp;
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206 }
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207 curavg = sqrt(curavg / (float) len);
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208
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209 // Evaluate an adequate 'mul' coefficient based on previous state, current
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210 // samples level, etc
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211 for (i = 0; i < NSAMPLES; i++)
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212 {
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213 avg += s->mem[i].avg * (float)s->mem[i].len;
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214 totallen += s->mem[i].len;
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215 }
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216
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217 if (totallen > MIN_SAMPLE_SIZE)
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218 {
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219 avg /= (float)totallen;
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220 if (avg >= SIL_S16)
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221 {
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222 s->mul = MID_S16 / avg;
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223 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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224 }
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225 }
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226
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227 // Scale & clamp the samples
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228 for (i = 0; i < len; i++)
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229 {
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230 tmp = s->mul * data[i];
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231 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
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232 data[i] = tmp;
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233 }
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234
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235 // Evaulation of newavg (not 100% accurate because of values clamping)
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236 newavg = s->mul * curavg;
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237
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238 // Stores computed values for future smoothing
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239 s->mem[s->idx].len = len;
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240 s->mem[s->idx].avg = newavg;
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241 s->idx = (s->idx + 1) % NSAMPLES;
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242 }
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243
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244 static void method2_float(af_volnorm_t *s, af_data_t *c)
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245 {
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246 register int i = 0;
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247 float *data = (float*)c->audio; // Audio data
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248 int len = c->len/4; // Number of samples
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249 float curavg = 0.0, newavg, avg = 0.0, tmp;
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250 int totallen = 0;
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251
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252 for (i = 0; i < len; i++)
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253 {
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254 tmp = data[i];
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255 curavg += tmp * tmp;
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256 }
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257 curavg = sqrt(curavg / (float) len);
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258
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259 // Evaluate an adequate 'mul' coefficient based on previous state, current
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260 // samples level, etc
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261 for (i = 0; i < NSAMPLES; i++)
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262 {
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263 avg += s->mem[i].avg * (float)s->mem[i].len;
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264 totallen += s->mem[i].len;
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265 }
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266
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267 if (totallen > MIN_SAMPLE_SIZE)
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268 {
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269 avg /= (float)totallen;
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270 if (avg >= SIL_FLOAT)
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271 {
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272 s->mul = MID_FLOAT / avg;
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273 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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274 }
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275 }
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276
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277 // Scale & clamp the samples
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278 for (i = 0; i < len; i++)
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279 data[i] *= s->mul;
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280
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281 // Evaulation of newavg (not 100% accurate because of values clamping)
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282 newavg = s->mul * curavg;
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283
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284 // Stores computed values for future smoothing
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285 s->mem[s->idx].len = len;
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286 s->mem[s->idx].avg = newavg;
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287 s->idx = (s->idx + 1) % NSAMPLES;
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288 }
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289
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290 // Filter data through filter
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291 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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292 {
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293 af_volnorm_t *s = af->setup;
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294
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295 if(af->data->format == (AF_FORMAT_SI | AF_FORMAT_NE))
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296 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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297 if (s->method)
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298 method2_int16(s, data);
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299 else
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300 method1_int16(s, data);
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301 }
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302 else if(af->data->format == (AF_FORMAT_F | AF_FORMAT_NE))
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303 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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304 if (s->method)
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305 method2_float(s, data);
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306 else
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307 method1_float(s, data);
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308 }
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309 return data;
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310 }
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311
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312 // Allocate memory and set function pointers
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313 static int open(af_instance_t* af){
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314 int i = 0;
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315 af->control=control;
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316 af->uninit=uninit;
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317 af->play=play;
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318 af->mul.n=1;
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319 af->mul.d=1;
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320 af->data=calloc(1,sizeof(af_data_t));
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321 af->setup=calloc(1,sizeof(af_volnorm_t));
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322 if(af->data == NULL || af->setup == NULL)
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323 return AF_ERROR;
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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324
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325 ((af_volnorm_t*)af->setup)->mul = MUL_INIT;
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326 ((af_volnorm_t*)af->setup)->lastavg = MID_S16;
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327 ((af_volnorm_t*)af->setup)->idx = 0;
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328 for (i = 0; i < NSAMPLES; i++)
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329 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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330 ((af_volnorm_t*)af->setup)->mem[i].len = 0;
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331 ((af_volnorm_t*)af->setup)->mem[i].avg = 0;
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332 }
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333 return AF_OK;
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334 }
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335
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336 // Description of this filter
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337 af_info_t af_info_volnorm = {
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338 "Volume normalizer filter",
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339 "volnorm",
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340 "Alex Beregszaszi & Pierre Lombard",
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341 "",
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342 AF_FLAGS_NOT_REENTRANT,
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343 open
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344 };