Mercurial > mplayer.hg
annotate libaf/af_volnorm.c @ 16752:425863aab3df
fix mencoder multi-file with some files having audio but others dont
author | ods15 |
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date | Thu, 13 Oct 2005 19:30:07 +0000 |
parents | 815f03b7cee5 |
children | 9352446d4fa8 |
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 <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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83 af->data->format = AF_FORMAT_S16_NE; | |
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84 af->data->bps = 2; |
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85 }else{ |
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87 af->data->bps = 4; |
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88 } |
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89 return af_test_output(af,(af_data_t*)arg); |
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90 case AF_CONTROL_COMMAND_LINE:{ |
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91 int i; |
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92 sscanf((char*)arg,"%d", &i); |
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93 if (i != 1 && i != 2) |
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94 return AF_ERROR; |
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95 s->method = i-1; |
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96 return AF_OK; |
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97 } |
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98 } |
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99 return AF_UNKNOWN; |
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100 } |
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101 |
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102 // Deallocate memory |
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103 static void uninit(struct af_instance_s* af) |
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104 { |
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105 if(af->data) |
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106 free(af->data); |
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107 if(af->setup) |
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108 free(af->setup); |
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109 } |
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110 |
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111 static void method1_int16(af_volnorm_t *s, af_data_t *c) |
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112 { |
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113 register int i = 0; |
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114 int16_t *data = (int16_t*)c->audio; // Audio data |
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115 int len = c->len/2; // Number of samples |
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116 float curavg = 0.0, newavg, neededmul; |
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117 int tmp; |
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118 |
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119 for (i = 0; i < len; i++) |
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120 { |
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121 tmp = data[i]; |
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122 curavg += tmp * tmp; |
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123 } |
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124 curavg = sqrt(curavg / (float) len); |
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125 |
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126 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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127 // samples level, etc |
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128 |
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129 if (curavg > SIL_S16) |
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130 { |
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131 neededmul = MID_S16 / (curavg * s->mul); |
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132 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; |
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133 |
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134 // clamp the mul coefficient |
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135 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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136 } |
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137 |
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138 // Scale & clamp the samples |
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139 for (i = 0; i < len; i++) |
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140 { |
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141 tmp = s->mul * data[i]; |
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142 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); |
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143 data[i] = tmp; |
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144 } |
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145 |
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146 // Evaulation of newavg (not 100% accurate because of values clamping) |
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147 newavg = s->mul * curavg; |
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148 |
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149 // Stores computed values for future smoothing |
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150 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; |
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151 } |
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152 |
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153 static void method1_float(af_volnorm_t *s, af_data_t *c) |
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154 { |
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155 register int i = 0; |
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156 float *data = (float*)c->audio; // Audio data |
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157 int len = c->len/4; // Number of samples |
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158 float curavg = 0.0, newavg, neededmul, tmp; |
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159 |
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160 for (i = 0; i < len; i++) |
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161 { |
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162 tmp = data[i]; |
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163 curavg += tmp * tmp; |
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164 } |
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165 curavg = sqrt(curavg / (float) len); |
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166 |
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167 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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168 // samples level, etc |
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169 |
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170 if (curavg > SIL_FLOAT) // FIXME |
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171 { |
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172 neededmul = MID_FLOAT / (curavg * s->mul); |
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173 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; |
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174 |
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175 // clamp the mul coefficient |
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176 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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177 } |
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178 |
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179 // Scale & clamp the samples |
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180 for (i = 0; i < len; i++) |
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181 data[i] *= s->mul; |
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182 |
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183 // Evaulation of newavg (not 100% accurate because of values clamping) |
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184 newavg = s->mul * curavg; |
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185 |
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186 // Stores computed values for future smoothing |
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187 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; |
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188 } |
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189 |
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190 static void method2_int16(af_volnorm_t *s, af_data_t *c) |
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191 { |
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192 register int i = 0; |
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193 int16_t *data = (int16_t*)c->audio; // Audio data |
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194 int len = c->len/2; // Number of samples |
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195 float curavg = 0.0, newavg, avg = 0.0; |
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196 int tmp, totallen = 0; |
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197 |
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198 for (i = 0; i < len; i++) |
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199 { |
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200 tmp = data[i]; |
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201 curavg += tmp * tmp; |
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202 } |
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203 curavg = sqrt(curavg / (float) len); |
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204 |
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205 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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206 // samples level, etc |
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207 for (i = 0; i < NSAMPLES; i++) |
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208 { |
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209 avg += s->mem[i].avg * (float)s->mem[i].len; |
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210 totallen += s->mem[i].len; |
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211 } |
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212 |
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213 if (totallen > MIN_SAMPLE_SIZE) |
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214 { |
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215 avg /= (float)totallen; |
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216 if (avg >= SIL_S16) |
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217 { |
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218 s->mul = MID_S16 / avg; |
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219 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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220 } |
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221 } |
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222 |
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223 // Scale & clamp the samples |
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224 for (i = 0; i < len; i++) |
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225 { |
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226 tmp = s->mul * data[i]; |
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227 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); |
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228 data[i] = tmp; |
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229 } |
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230 |
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231 // Evaulation of newavg (not 100% accurate because of values clamping) |
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232 newavg = s->mul * curavg; |
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233 |
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234 // Stores computed values for future smoothing |
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235 s->mem[s->idx].len = len; |
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236 s->mem[s->idx].avg = newavg; |
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237 s->idx = (s->idx + 1) % NSAMPLES; |
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238 } |
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239 |
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240 static void method2_float(af_volnorm_t *s, af_data_t *c) |
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241 { |
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242 register int i = 0; |
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243 float *data = (float*)c->audio; // Audio data |
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244 int len = c->len/4; // Number of samples |
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245 float curavg = 0.0, newavg, avg = 0.0, tmp; |
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246 int totallen = 0; |
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247 |
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248 for (i = 0; i < len; i++) |
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249 { |
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250 tmp = data[i]; |
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251 curavg += tmp * tmp; |
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252 } |
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253 curavg = sqrt(curavg / (float) len); |
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254 |
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255 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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256 // samples level, etc |
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257 for (i = 0; i < NSAMPLES; i++) |
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258 { |
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259 avg += s->mem[i].avg * (float)s->mem[i].len; |
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260 totallen += s->mem[i].len; |
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261 } |
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262 |
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263 if (totallen > MIN_SAMPLE_SIZE) |
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264 { |
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265 avg /= (float)totallen; |
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266 if (avg >= SIL_FLOAT) |
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267 { |
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268 s->mul = MID_FLOAT / avg; |
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269 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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270 } |
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271 } |
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272 |
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273 // Scale & clamp the samples |
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274 for (i = 0; i < len; i++) |
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275 data[i] *= s->mul; |
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276 |
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277 // Evaulation of newavg (not 100% accurate because of values clamping) |
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278 newavg = s->mul * curavg; |
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279 |
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280 // Stores computed values for future smoothing |
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281 s->mem[s->idx].len = len; |
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282 s->mem[s->idx].avg = newavg; |
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283 s->idx = (s->idx + 1) % NSAMPLES; |
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284 } |
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285 |
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286 // Filter data through filter |
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287 static af_data_t* play(struct af_instance_s* af, af_data_t* data) |
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288 { |
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289 af_volnorm_t *s = af->setup; |
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290 |
14245 | 291 if(af->data->format == (AF_FORMAT_S16_NE)) |
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292 { |
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293 if (s->method) |
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294 method2_int16(s, data); |
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295 else |
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296 method1_int16(s, data); |
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297 } |
14245 | 298 else if(af->data->format == (AF_FORMAT_FLOAT_NE)) |
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299 { |
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300 if (s->method) |
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301 method2_float(s, data); |
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302 else |
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303 method1_float(s, data); |
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304 } |
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305 return data; |
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306 } |
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307 |
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308 // Allocate memory and set function pointers |
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309 static int open(af_instance_t* af){ |
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310 int i = 0; |
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311 af->control=control; |
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312 af->uninit=uninit; |
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313 af->play=play; |
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314 af->mul.n=1; |
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315 af->mul.d=1; |
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316 af->data=calloc(1,sizeof(af_data_t)); |
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317 af->setup=calloc(1,sizeof(af_volnorm_t)); |
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318 if(af->data == NULL || af->setup == NULL) |
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319 return AF_ERROR; |
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320 |
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321 ((af_volnorm_t*)af->setup)->mul = MUL_INIT; |
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322 ((af_volnorm_t*)af->setup)->lastavg = MID_S16; |
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323 ((af_volnorm_t*)af->setup)->idx = 0; |
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324 for (i = 0; i < NSAMPLES; i++) |
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325 { |
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326 ((af_volnorm_t*)af->setup)->mem[i].len = 0; |
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327 ((af_volnorm_t*)af->setup)->mem[i].avg = 0; |
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328 } |
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329 return AF_OK; |
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330 } |
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331 |
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332 // Description of this filter |
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333 af_info_t af_info_volnorm = { |
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334 "Volume normalizer filter", |
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335 "volnorm", |
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336 "Alex Beregszaszi & Pierre Lombard", |
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337 "", |
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338 AF_FLAGS_NOT_REENTRANT, |
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339 open |
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340 }; |