Mercurial > mplayer.hg
annotate libaf/af_volnorm.c @ 32204:0a81c931ecb5
Mark functs not used outside of the file as static, remove from header; fixes:
libvo/vesa_lvo.c:187: warning: no previous prototype for 'vlvo_draw_slice'
libvo/vesa_lvo.c:205: warning: no previous prototype for 'vlvo_draw_frame'
libvo/vesa_lvo.c:214: warning: no previous prototype for 'vlvo_flip_page'
libvo/vesa_lvo.c:275: warning: no previous prototype for 'vlvo_draw_osd'
libvo/vesa_lvo.c:312: warning: no previous prototype for 'vlvo_query_info'
author | diego |
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date | Fri, 17 Sep 2010 10:31:24 +0000 |
parents | 0f1b5b68af32 |
children | 8fa2f43cb760 |
rev | line source |
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1 /* |
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2 * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard |
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3 * |
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4 * This file is part of MPlayer. |
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5 * |
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6 * MPlayer is free software; you can redistribute it and/or modify |
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7 * it under the terms of the GNU General Public License as published by |
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8 * the Free Software Foundation; either version 2 of the License, or |
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9 * (at your option) any later version. |
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10 * |
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11 * MPlayer is distributed in the hope that it will be useful, |
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12 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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14 * GNU General Public License for more details. |
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15 * |
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16 * You should have received a copy of the GNU General Public License along |
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17 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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18 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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19 */ |
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20 |
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21 #include <stdio.h> |
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22 #include <stdlib.h> |
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23 #include <string.h> |
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24 |
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25 #include <inttypes.h> |
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26 #include <math.h> |
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27 #include <limits.h> |
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28 |
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29 #include "af.h" |
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30 |
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31 // Methods: |
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32 // 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1) |
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33 // 2: uses several samples to smooth the variations (standard weighted mean |
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34 // on past samples) |
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35 |
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36 // Size of the memory array |
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37 // FIXME: should depend on the frequency of the data (should be a few seconds) |
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38 #define NSAMPLES 128 |
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39 |
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40 // If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we |
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41 // choose to ignore the computed value as it's not significant enough |
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42 // FIXME: should depend on the frequency of the data (0.5s maybe) |
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43 #define MIN_SAMPLE_SIZE 32000 |
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44 |
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45 // mul is the value by which the samples are scaled |
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46 // and has to be in [MUL_MIN, MUL_MAX] |
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47 #define MUL_INIT 1.0 |
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48 #define MUL_MIN 0.1 |
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49 #define MUL_MAX 5.0 |
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50 |
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51 // Silence level |
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52 // FIXME: should be relative to the level of the samples |
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53 #define SIL_S16 (SHRT_MAX * 0.01) |
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54 #define SIL_FLOAT (INT_MAX * 0.01) // FIXME |
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55 |
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56 // smooth must be in ]0.0, 1.0[ |
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57 #define SMOOTH_MUL 0.06 |
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58 #define SMOOTH_LASTAVG 0.06 |
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59 |
16971 | 60 #define DEFAULT_TARGET 0.25 |
61 | |
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62 // Data for specific instances of this filter |
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63 typedef struct af_volume_s |
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64 { |
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65 int method; // method used |
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66 float mul; |
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67 // method 1 |
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68 float lastavg; // history value of the filter |
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69 // method 2 |
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70 int idx; |
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71 struct { |
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72 float avg; // average level of the sample |
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73 int len; // sample size (weight) |
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74 } mem[NSAMPLES]; |
16971 | 75 // "Ideal" level |
76 float mid_s16; | |
77 float mid_float; | |
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78 }af_volnorm_t; |
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79 |
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80 // Initialization and runtime control |
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81 static int control(struct af_instance_s* af, int cmd, void* arg) |
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82 { |
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83 af_volnorm_t* s = (af_volnorm_t*)af->setup; |
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84 |
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85 switch(cmd){ |
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86 case AF_CONTROL_REINIT: |
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87 // Sanity check |
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88 if(!arg) return AF_ERROR; |
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89 |
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90 af->data->rate = ((af_data_t*)arg)->rate; |
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91 af->data->nch = ((af_data_t*)arg)->nch; |
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92 |
14245 | 93 if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){ |
94 af->data->format = AF_FORMAT_S16_NE; | |
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95 af->data->bps = 2; |
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96 }else{ |
14245 | 97 af->data->format = AF_FORMAT_FLOAT_NE; |
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98 af->data->bps = 4; |
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99 } |
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100 return af_test_output(af,(af_data_t*)arg); |
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101 case AF_CONTROL_COMMAND_LINE:{ |
16971 | 102 int i = 0; |
103 float target = DEFAULT_TARGET; | |
104 sscanf((char*)arg,"%d:%f", &i, &target); | |
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105 if (i != 1 && i != 2) |
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106 return AF_ERROR; |
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107 s->method = i-1; |
16971 | 108 s->mid_s16 = ((float)SHRT_MAX) * target; |
109 s->mid_float = ((float)INT_MAX) * target; | |
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110 return AF_OK; |
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111 } |
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112 } |
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113 return AF_UNKNOWN; |
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114 } |
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115 |
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116 // Deallocate memory |
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117 static void uninit(struct af_instance_s* af) |
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118 { |
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119 if(af->data) |
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120 free(af->data); |
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121 if(af->setup) |
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122 free(af->setup); |
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123 } |
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124 |
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125 static void method1_int16(af_volnorm_t *s, af_data_t *c) |
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126 { |
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127 register int i = 0; |
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128 int16_t *data = (int16_t*)c->audio; // Audio data |
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129 int len = c->len/2; // Number of samples |
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130 float curavg = 0.0, newavg, neededmul; |
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131 int tmp; |
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132 |
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133 for (i = 0; i < len; i++) |
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134 { |
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135 tmp = data[i]; |
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136 curavg += tmp * tmp; |
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137 } |
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138 curavg = sqrt(curavg / (float) len); |
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139 |
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140 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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141 // samples level, etc |
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142 |
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143 if (curavg > SIL_S16) |
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144 { |
16971 | 145 neededmul = s->mid_s16 / (curavg * s->mul); |
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146 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; |
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147 |
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148 // clamp the mul coefficient |
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149 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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150 } |
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151 |
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152 // Scale & clamp the samples |
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153 for (i = 0; i < len; i++) |
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154 { |
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155 tmp = s->mul * data[i]; |
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156 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); |
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157 data[i] = tmp; |
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158 } |
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159 |
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160 // Evaulation of newavg (not 100% accurate because of values clamping) |
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161 newavg = s->mul * curavg; |
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162 |
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163 // Stores computed values for future smoothing |
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164 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; |
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165 } |
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166 |
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167 static void method1_float(af_volnorm_t *s, af_data_t *c) |
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168 { |
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169 register int i = 0; |
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170 float *data = (float*)c->audio; // Audio data |
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171 int len = c->len/4; // Number of samples |
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172 float curavg = 0.0, newavg, neededmul, tmp; |
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173 |
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174 for (i = 0; i < len; i++) |
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175 { |
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176 tmp = data[i]; |
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177 curavg += tmp * tmp; |
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178 } |
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179 curavg = sqrt(curavg / (float) len); |
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180 |
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181 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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182 // samples level, etc |
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183 |
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184 if (curavg > SIL_FLOAT) // FIXME |
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185 { |
16971 | 186 neededmul = s->mid_float / (curavg * s->mul); |
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187 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; |
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188 |
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189 // clamp the mul coefficient |
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190 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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191 } |
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192 |
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193 // Scale & clamp the samples |
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194 for (i = 0; i < len; i++) |
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195 data[i] *= s->mul; |
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196 |
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197 // Evaulation of newavg (not 100% accurate because of values clamping) |
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198 newavg = s->mul * curavg; |
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199 |
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200 // Stores computed values for future smoothing |
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201 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; |
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202 } |
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203 |
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204 static void method2_int16(af_volnorm_t *s, af_data_t *c) |
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205 { |
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206 register int i = 0; |
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207 int16_t *data = (int16_t*)c->audio; // Audio data |
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208 int len = c->len/2; // Number of samples |
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209 float curavg = 0.0, newavg, avg = 0.0; |
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210 int tmp, totallen = 0; |
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211 |
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212 for (i = 0; i < len; i++) |
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213 { |
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214 tmp = data[i]; |
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215 curavg += tmp * tmp; |
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216 } |
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217 curavg = sqrt(curavg / (float) len); |
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218 |
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219 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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220 // samples level, etc |
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221 for (i = 0; i < NSAMPLES; i++) |
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222 { |
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223 avg += s->mem[i].avg * (float)s->mem[i].len; |
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224 totallen += s->mem[i].len; |
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225 } |
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226 |
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227 if (totallen > MIN_SAMPLE_SIZE) |
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228 { |
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229 avg /= (float)totallen; |
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230 if (avg >= SIL_S16) |
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231 { |
16971 | 232 s->mul = s->mid_s16 / avg; |
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233 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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234 } |
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235 } |
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236 |
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237 // Scale & clamp the samples |
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238 for (i = 0; i < len; i++) |
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239 { |
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240 tmp = s->mul * data[i]; |
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241 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); |
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242 data[i] = tmp; |
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243 } |
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244 |
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245 // Evaulation of newavg (not 100% accurate because of values clamping) |
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246 newavg = s->mul * curavg; |
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247 |
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248 // Stores computed values for future smoothing |
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249 s->mem[s->idx].len = len; |
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250 s->mem[s->idx].avg = newavg; |
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251 s->idx = (s->idx + 1) % NSAMPLES; |
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252 } |
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253 |
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254 static void method2_float(af_volnorm_t *s, af_data_t *c) |
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255 { |
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256 register int i = 0; |
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257 float *data = (float*)c->audio; // Audio data |
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258 int len = c->len/4; // Number of samples |
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259 float curavg = 0.0, newavg, avg = 0.0, tmp; |
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260 int totallen = 0; |
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261 |
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262 for (i = 0; i < len; i++) |
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263 { |
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264 tmp = data[i]; |
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265 curavg += tmp * tmp; |
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266 } |
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267 curavg = sqrt(curavg / (float) len); |
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268 |
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269 // Evaluate an adequate 'mul' coefficient based on previous state, current |
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270 // samples level, etc |
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271 for (i = 0; i < NSAMPLES; i++) |
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272 { |
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273 avg += s->mem[i].avg * (float)s->mem[i].len; |
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274 totallen += s->mem[i].len; |
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275 } |
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276 |
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277 if (totallen > MIN_SAMPLE_SIZE) |
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278 { |
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279 avg /= (float)totallen; |
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280 if (avg >= SIL_FLOAT) |
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281 { |
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283 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); |
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284 } |
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285 } |
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286 |
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287 // Scale & clamp the samples |
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288 for (i = 0; i < len; i++) |
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289 data[i] *= s->mul; |
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290 |
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291 // Evaulation of newavg (not 100% accurate because of values clamping) |
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292 newavg = s->mul * curavg; |
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293 |
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294 // Stores computed values for future smoothing |
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295 s->mem[s->idx].len = len; |
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296 s->mem[s->idx].avg = newavg; |
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297 s->idx = (s->idx + 1) % NSAMPLES; |
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298 } |
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299 |
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300 // Filter data through filter |
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301 static af_data_t* play(struct af_instance_s* af, af_data_t* data) |
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302 { |
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303 af_volnorm_t *s = af->setup; |
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304 |
14245 | 305 if(af->data->format == (AF_FORMAT_S16_NE)) |
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306 { |
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307 if (s->method) |
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308 method2_int16(s, data); |
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309 else |
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310 method1_int16(s, data); |
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311 } |
14245 | 312 else if(af->data->format == (AF_FORMAT_FLOAT_NE)) |
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313 { |
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314 if (s->method) |
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315 method2_float(s, data); |
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316 else |
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317 method1_float(s, data); |
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318 } |
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319 return data; |
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320 } |
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321 |
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322 // Allocate memory and set function pointers |
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323 static int af_open(af_instance_t* af){ |
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324 int i = 0; |
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325 af->control=control; |
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326 af->uninit=uninit; |
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327 af->play=play; |
24888 | 328 af->mul=1; |
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329 af->data=calloc(1,sizeof(af_data_t)); |
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330 af->setup=calloc(1,sizeof(af_volnorm_t)); |
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331 if(af->data == NULL || af->setup == NULL) |
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332 return AF_ERROR; |
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333 |
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334 ((af_volnorm_t*)af->setup)->mul = MUL_INIT; |
16971 | 335 ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET; |
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336 ((af_volnorm_t*)af->setup)->idx = 0; |
16971 | 337 ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET; |
338 ((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET; | |
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339 for (i = 0; i < NSAMPLES; i++) |
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340 { |
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341 ((af_volnorm_t*)af->setup)->mem[i].len = 0; |
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342 ((af_volnorm_t*)af->setup)->mem[i].avg = 0; |
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343 } |
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344 return AF_OK; |
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345 } |
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346 |
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347 // Description of this filter |
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348 af_info_t af_info_volnorm = { |
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349 "Volume normalizer filter", |
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350 "volnorm", |
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351 "Alex Beregszaszi & Pierre Lombard", |
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352 "", |
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353 AF_FLAGS_NOT_REENTRANT, |
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354 af_open |
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355 }; |