Mercurial > mplayer.hg
annotate libaf/af_volnorm.c @ 23762:896a1faed1d2
Group all vo driver checks together.
author | diego |
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date | Sat, 14 Jul 2007 15:11:08 +0000 |
parents | 904e3f3f8bee |
children | b2402b4f0afa |
rev | line source |
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1 /*============================================================================= |
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2 // |
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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 <inttypes.h> |
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16 #include <math.h> |
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17 #include <limits.h> |
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18 |
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19 #include "af.h" |
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20 |
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21 // Methods: |
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22 // 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1) |
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23 // 2: uses several samples to smooth the variations (standard weighted mean |
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24 // on past samples) |
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25 |
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26 // Size of the memory array |
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27 // FIXME: should depend on the frequency of the data (should be a few seconds) |
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28 #define NSAMPLES 128 |
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29 |
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30 // If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we |
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31 // choose to ignore the computed value as it's not significant enough |
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32 // FIXME: should depend on the frequency of the data (0.5s maybe) |
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33 #define MIN_SAMPLE_SIZE 32000 |
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34 |
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35 // mul is the value by which the samples are scaled |
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36 // and has to be in [MUL_MIN, MUL_MAX] |
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37 #define MUL_INIT 1.0 |
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38 #define MUL_MIN 0.1 |
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39 #define MUL_MAX 5.0 |
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40 |
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41 // Silence level |
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42 // FIXME: should be relative to the level of the samples |
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43 #define SIL_S16 (SHRT_MAX * 0.01) |
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44 #define SIL_FLOAT (INT_MAX * 0.01) // FIXME |
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45 |
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46 // smooth must be in ]0.0, 1.0[ |
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47 #define SMOOTH_MUL 0.06 |
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48 #define SMOOTH_LASTAVG 0.06 |
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49 |
16971 | 50 #define DEFAULT_TARGET 0.25 |
51 | |
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52 // Data for specific instances of this filter |
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53 typedef struct af_volume_s |
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54 { |
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55 int method; // method used |
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56 float mul; |
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57 // method 1 |
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58 float lastavg; // history value of the filter |
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59 // method 2 |
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60 int idx; |
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61 struct { |
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62 float avg; // average level of the sample |
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63 int len; // sample size (weight) |
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64 } mem[NSAMPLES]; |
16971 | 65 // "Ideal" level |
66 float mid_s16; | |
67 float mid_float; | |
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68 }af_volnorm_t; |
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69 |
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70 // Initialization and runtime control |
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71 static int control(struct af_instance_s* af, int cmd, void* arg) |
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72 { |
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73 af_volnorm_t* s = (af_volnorm_t*)af->setup; |
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74 |
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75 switch(cmd){ |
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76 case AF_CONTROL_REINIT: |
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77 // Sanity check |
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78 if(!arg) return AF_ERROR; |
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79 |
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80 af->data->rate = ((af_data_t*)arg)->rate; |
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81 af->data->nch = ((af_data_t*)arg)->nch; |
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82 |
14245 | 83 if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){ |
84 af->data->format = AF_FORMAT_S16_NE; | |
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85 af->data->bps = 2; |
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86 }else{ |
14245 | 87 af->data->format = AF_FORMAT_FLOAT_NE; |
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88 af->data->bps = 4; |
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89 } |
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90 return af_test_output(af,(af_data_t*)arg); |
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91 case AF_CONTROL_COMMAND_LINE:{ |
16971 | 92 int i = 0; |
93 float target = DEFAULT_TARGET; | |
94 sscanf((char*)arg,"%d:%f", &i, &target); | |
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95 if (i != 1 && i != 2) |
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96 return AF_ERROR; |
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97 s->method = i-1; |
16971 | 98 s->mid_s16 = ((float)SHRT_MAX) * target; |
99 s->mid_float = ((float)INT_MAX) * target; | |
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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 { |
16971 | 135 neededmul = s->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 { |
16971 | 176 neededmul = s->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 { |
16971 | 222 s->mul = s->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 { |
16971 | 272 s->mul = s->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 |
14245 | 295 if(af->data->format == (AF_FORMAT_S16_NE)) |
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296 { |
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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 } |
14245 | 302 else if(af->data->format == (AF_FORMAT_FLOAT_NE)) |
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303 { |
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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 af_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; |
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324 |
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325 ((af_volnorm_t*)af->setup)->mul = MUL_INIT; |
16971 | 326 ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET; |
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327 ((af_volnorm_t*)af->setup)->idx = 0; |
16971 | 328 ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET; |
329 ((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET; | |
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330 for (i = 0; i < NSAMPLES; i++) |
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331 { |
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332 ((af_volnorm_t*)af->setup)->mem[i].len = 0; |
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333 ((af_volnorm_t*)af->setup)->mem[i].avg = 0; |
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334 } |
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335 return AF_OK; |
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336 } |
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337 |
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338 // Description of this filter |
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339 af_info_t af_info_volnorm = { |
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340 "Volume normalizer filter", |
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341 "volnorm", |
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342 "Alex Beregszaszi & Pierre Lombard", |
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343 "", |
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344 AF_FLAGS_NOT_REENTRANT, |
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345 af_open |
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346 }; |