annotate libaf/af_volnorm.c @ 17921:baa1df31e661

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author kraymer
date Wed, 22 Mar 2006 18:19:15 +0000
parents 9352446d4fa8
children fd6f824ef894
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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
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42 // Silence level
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43 // FIXME: should be relative to the level of the samples
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44 #define SIL_S16 (SHRT_MAX * 0.01)
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45 #define SIL_FLOAT (INT_MAX * 0.01) // FIXME
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46
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47 // smooth must be in ]0.0, 1.0[
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48 #define SMOOTH_MUL 0.06
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49 #define SMOOTH_LASTAVG 0.06
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50
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51 #define DEFAULT_TARGET 0.25
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52
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53 // Data for specific instances of this filter
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54 typedef struct af_volume_s
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55 {
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56 int method; // method used
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57 float mul;
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58 // method 1
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59 float lastavg; // history value of the filter
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60 // method 2
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61 int idx;
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62 struct {
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63 float avg; // average level of the sample
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64 int len; // sample size (weight)
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65 } mem[NSAMPLES];
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66 // "Ideal" level
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67 float mid_s16;
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68 float mid_float;
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69 }af_volnorm_t;
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70
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71 // Initialization and runtime control
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72 static int control(struct af_instance_s* af, int cmd, void* arg)
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73 {
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74 af_volnorm_t* s = (af_volnorm_t*)af->setup;
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75
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76 switch(cmd){
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77 case AF_CONTROL_REINIT:
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78 // Sanity check
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79 if(!arg) return AF_ERROR;
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80
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81 af->data->rate = ((af_data_t*)arg)->rate;
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82 af->data->nch = ((af_data_t*)arg)->nch;
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83
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84 if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){
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85 af->data->format = AF_FORMAT_S16_NE;
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86 af->data->bps = 2;
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87 }else{
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88 af->data->format = AF_FORMAT_FLOAT_NE;
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89 af->data->bps = 4;
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90 }
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91 return af_test_output(af,(af_data_t*)arg);
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92 case AF_CONTROL_COMMAND_LINE:{
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93 int i = 0;
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94 float target = DEFAULT_TARGET;
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95 sscanf((char*)arg,"%d:%f", &i, &target);
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96 if (i != 1 && i != 2)
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97 return AF_ERROR;
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98 s->method = i-1;
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99 s->mid_s16 = ((float)SHRT_MAX) * target;
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100 s->mid_float = ((float)INT_MAX) * target;
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101 return AF_OK;
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102 }
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103 }
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104 return AF_UNKNOWN;
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105 }
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106
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107 // Deallocate memory
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108 static void uninit(struct af_instance_s* af)
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109 {
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110 if(af->data)
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111 free(af->data);
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112 if(af->setup)
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113 free(af->setup);
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114 }
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115
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116 static void method1_int16(af_volnorm_t *s, af_data_t *c)
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117 {
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118 register int i = 0;
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119 int16_t *data = (int16_t*)c->audio; // Audio data
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120 int len = c->len/2; // Number of samples
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121 float curavg = 0.0, newavg, neededmul;
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122 int tmp;
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123
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124 for (i = 0; i < len; i++)
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125 {
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126 tmp = data[i];
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127 curavg += tmp * tmp;
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128 }
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129 curavg = sqrt(curavg / (float) len);
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130
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131 // Evaluate an adequate 'mul' coefficient based on previous state, current
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132 // samples level, etc
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133
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134 if (curavg > SIL_S16)
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135 {
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136 neededmul = s->mid_s16 / (curavg * s->mul);
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137 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
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138
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139 // clamp the mul coefficient
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140 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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141 }
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142
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143 // Scale & clamp the samples
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144 for (i = 0; i < len; i++)
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145 {
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146 tmp = s->mul * data[i];
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147 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
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148 data[i] = tmp;
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149 }
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150
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151 // Evaulation of newavg (not 100% accurate because of values clamping)
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152 newavg = s->mul * curavg;
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153
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154 // Stores computed values for future smoothing
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155 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
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156 }
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157
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158 static void method1_float(af_volnorm_t *s, af_data_t *c)
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159 {
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160 register int i = 0;
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161 float *data = (float*)c->audio; // Audio data
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162 int len = c->len/4; // Number of samples
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163 float curavg = 0.0, newavg, neededmul, tmp;
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164
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165 for (i = 0; i < len; i++)
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166 {
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167 tmp = data[i];
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168 curavg += tmp * tmp;
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169 }
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170 curavg = sqrt(curavg / (float) len);
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171
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172 // Evaluate an adequate 'mul' coefficient based on previous state, current
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173 // samples level, etc
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174
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175 if (curavg > SIL_FLOAT) // FIXME
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176 {
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177 neededmul = s->mid_float / (curavg * s->mul);
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178 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
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179
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180 // clamp the mul coefficient
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181 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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182 }
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183
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184 // Scale & clamp the samples
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185 for (i = 0; i < len; i++)
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186 data[i] *= s->mul;
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187
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188 // Evaulation of newavg (not 100% accurate because of values clamping)
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189 newavg = s->mul * curavg;
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190
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191 // Stores computed values for future smoothing
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192 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
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193 }
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194
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195 static void method2_int16(af_volnorm_t *s, af_data_t *c)
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196 {
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197 register int i = 0;
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198 int16_t *data = (int16_t*)c->audio; // Audio data
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199 int len = c->len/2; // Number of samples
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200 float curavg = 0.0, newavg, avg = 0.0;
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201 int tmp, totallen = 0;
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202
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203 for (i = 0; i < len; i++)
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204 {
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205 tmp = data[i];
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206 curavg += tmp * tmp;
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207 }
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208 curavg = sqrt(curavg / (float) len);
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209
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210 // Evaluate an adequate 'mul' coefficient based on previous state, current
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211 // samples level, etc
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212 for (i = 0; i < NSAMPLES; i++)
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213 {
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214 avg += s->mem[i].avg * (float)s->mem[i].len;
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215 totallen += s->mem[i].len;
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216 }
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217
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218 if (totallen > MIN_SAMPLE_SIZE)
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219 {
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220 avg /= (float)totallen;
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221 if (avg >= SIL_S16)
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222 {
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223 s->mul = s->mid_s16 / avg;
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224 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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225 }
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226 }
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227
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228 // Scale & clamp the samples
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229 for (i = 0; i < len; i++)
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230 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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231 tmp = s->mul * data[i];
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232 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
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233 data[i] = tmp;
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234 }
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235
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236 // Evaulation of newavg (not 100% accurate because of values clamping)
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237 newavg = s->mul * curavg;
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238
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239 // Stores computed values for future smoothing
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240 s->mem[s->idx].len = len;
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241 s->mem[s->idx].avg = newavg;
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242 s->idx = (s->idx + 1) % NSAMPLES;
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243 }
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244
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245 static void method2_float(af_volnorm_t *s, af_data_t *c)
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246 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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247 register int i = 0;
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248 float *data = (float*)c->audio; // Audio data
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249 int len = c->len/4; // Number of samples
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250 float curavg = 0.0, newavg, avg = 0.0, tmp;
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251 int totallen = 0;
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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252
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253 for (i = 0; i < len; i++)
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254 {
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255 tmp = data[i];
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256 curavg += tmp * tmp;
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257 }
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258 curavg = sqrt(curavg / (float) len);
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259
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260 // Evaluate an adequate 'mul' coefficient based on previous state, current
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261 // samples level, etc
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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262 for (i = 0; i < NSAMPLES; i++)
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263 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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264 avg += s->mem[i].avg * (float)s->mem[i].len;
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265 totallen += s->mem[i].len;
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266 }
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267
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268 if (totallen > MIN_SAMPLE_SIZE)
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269 {
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270 avg /= (float)totallen;
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271 if (avg >= SIL_FLOAT)
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272 {
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273 s->mul = s->mid_float / avg;
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274 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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275 }
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276 }
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277
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278 // Scale & clamp the samples
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279 for (i = 0; i < len; i++)
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280 data[i] *= s->mul;
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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281
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282 // Evaulation of newavg (not 100% accurate because of values clamping)
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283 newavg = s->mul * curavg;
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284
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285 // Stores computed values for future smoothing
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286 s->mem[s->idx].len = len;
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287 s->mem[s->idx].avg = newavg;
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288 s->idx = (s->idx + 1) % NSAMPLES;
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289 }
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290
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291 // Filter data through filter
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292 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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293 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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294 af_volnorm_t *s = af->setup;
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295
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296 if(af->data->format == (AF_FORMAT_S16_NE))
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297 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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298 if (s->method)
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299 method2_int16(s, data);
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300 else
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301 method1_int16(s, data);
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302 }
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303 else if(af->data->format == (AF_FORMAT_FLOAT_NE))
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304 {
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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305 if (s->method)
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306 method2_float(s, data);
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307 else
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308 method1_float(s, data);
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309 }
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310 return data;
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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311 }
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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312
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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313 // Allocate memory and set function pointers
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314 static int open(af_instance_t* af){
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315 int i = 0;
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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316 af->control=control;
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317 af->uninit=uninit;
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318 af->play=play;
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319 af->mul.n=1;
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320 af->mul.d=1;
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321 af->data=calloc(1,sizeof(af_data_t));
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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322 af->setup=calloc(1,sizeof(af_volnorm_t));
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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323 if(af->data == NULL || af->setup == NULL)
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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324 return AF_ERROR;
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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325
81e62cbe57d9 reimplementation of the pl_extrastereo and pl_volnorm plugins
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326 ((af_volnorm_t*)af->setup)->mul = MUL_INIT;
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327 ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET;
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328 ((af_volnorm_t*)af->setup)->idx = 0;
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329 ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET;
9352446d4fa8 Adds a target parameter to the volnorm filter.
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330 ((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET;
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331 for (i = 0; i < NSAMPLES; i++)
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332 {
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333 ((af_volnorm_t*)af->setup)->mem[i].len = 0;
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334 ((af_volnorm_t*)af->setup)->mem[i].avg = 0;
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335 }
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336 return AF_OK;
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337 }
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338
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339 // Description of this filter
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340 af_info_t af_info_volnorm = {
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341 "Volume normalizer filter",
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342 "volnorm",
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343 "Alex Beregszaszi & Pierre Lombard",
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344 "",
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345 AF_FLAGS_NOT_REENTRANT,
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346 open
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347 };