annotate libaf/af_volnorm.c @ 35405:d6305a78a11e

Fix code that detects streams temporarily lacking data to work properly with e.g. DVDs.
author reimar
date Wed, 28 Nov 2012 19:15:34 +0000
parents 8fa2f43cb760
children 1dca5ff3d3c6
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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
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60 #define DEFAULT_TARGET 0.25
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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];
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75 // "Ideal" level
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76 float mid_s16;
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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
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93 if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){
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94 af->data->format = AF_FORMAT_S16_NE;
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95 af->data->bps = 2;
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96 }else{
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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:{
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102 int i = 0;
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103 float target = DEFAULT_TARGET;
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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;
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108 s->mid_s16 = ((float)SHRT_MAX) * target;
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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 free(af->data);
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120 free(af->setup);
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121 }
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122
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123 static void method1_int16(af_volnorm_t *s, af_data_t *c)
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124 {
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125 register int i = 0;
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126 int16_t *data = (int16_t*)c->audio; // Audio data
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127 int len = c->len/2; // Number of samples
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128 float curavg = 0.0, newavg, neededmul;
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129 int tmp;
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130
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131 for (i = 0; i < len; i++)
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132 {
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133 tmp = data[i];
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134 curavg += tmp * tmp;
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135 }
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136 curavg = sqrt(curavg / (float) len);
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137
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138 // Evaluate an adequate 'mul' coefficient based on previous state, current
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139 // samples level, etc
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140
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141 if (curavg > SIL_S16)
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142 {
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143 neededmul = s->mid_s16 / (curavg * s->mul);
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144 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
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145
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146 // clamp the mul coefficient
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147 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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148 }
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149
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150 // Scale & clamp the samples
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151 for (i = 0; i < len; i++)
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152 {
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153 tmp = s->mul * data[i];
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154 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
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155 data[i] = tmp;
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156 }
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157
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158 // Evaulation of newavg (not 100% accurate because of values clamping)
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159 newavg = s->mul * curavg;
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160
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161 // Stores computed values for future smoothing
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162 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
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163 }
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164
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165 static void method1_float(af_volnorm_t *s, af_data_t *c)
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166 {
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167 register int i = 0;
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168 float *data = (float*)c->audio; // Audio data
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169 int len = c->len/4; // Number of samples
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170 float curavg = 0.0, newavg, neededmul, tmp;
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171
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172 for (i = 0; i < len; i++)
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173 {
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174 tmp = data[i];
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175 curavg += tmp * tmp;
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176 }
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177 curavg = sqrt(curavg / (float) len);
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178
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179 // Evaluate an adequate 'mul' coefficient based on previous state, current
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180 // samples level, etc
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181
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182 if (curavg > SIL_FLOAT) // FIXME
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183 {
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184 neededmul = s->mid_float / (curavg * s->mul);
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185 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
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186
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187 // clamp the mul coefficient
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188 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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189 }
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190
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191 // Scale & clamp the samples
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192 for (i = 0; i < len; i++)
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193 data[i] *= s->mul;
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194
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195 // Evaulation of newavg (not 100% accurate because of values clamping)
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196 newavg = s->mul * curavg;
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197
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198 // Stores computed values for future smoothing
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199 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
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200 }
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201
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202 static void method2_int16(af_volnorm_t *s, af_data_t *c)
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203 {
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204 register int i = 0;
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205 int16_t *data = (int16_t*)c->audio; // Audio data
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206 int len = c->len/2; // Number of samples
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207 float curavg = 0.0, newavg, avg = 0.0;
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208 int tmp, totallen = 0;
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209
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210 for (i = 0; i < len; i++)
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211 {
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212 tmp = data[i];
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213 curavg += tmp * tmp;
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214 }
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215 curavg = sqrt(curavg / (float) len);
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216
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217 // Evaluate an adequate 'mul' coefficient based on previous state, current
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218 // samples level, etc
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219 for (i = 0; i < NSAMPLES; i++)
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220 {
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221 avg += s->mem[i].avg * (float)s->mem[i].len;
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222 totallen += s->mem[i].len;
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223 }
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224
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225 if (totallen > MIN_SAMPLE_SIZE)
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226 {
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227 avg /= (float)totallen;
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228 if (avg >= SIL_S16)
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229 {
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230 s->mul = s->mid_s16 / avg;
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231 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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232 }
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233 }
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234
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235 // Scale & clamp the samples
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236 for (i = 0; i < len; i++)
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237 {
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238 tmp = s->mul * data[i];
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239 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
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240 data[i] = tmp;
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241 }
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242
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243 // Evaulation of newavg (not 100% accurate because of values clamping)
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244 newavg = s->mul * curavg;
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245
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246 // Stores computed values for future smoothing
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247 s->mem[s->idx].len = len;
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248 s->mem[s->idx].avg = newavg;
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249 s->idx = (s->idx + 1) % NSAMPLES;
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250 }
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251
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252 static void method2_float(af_volnorm_t *s, af_data_t *c)
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253 {
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254 register int i = 0;
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255 float *data = (float*)c->audio; // Audio data
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256 int len = c->len/4; // Number of samples
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257 float curavg = 0.0, newavg, avg = 0.0, tmp;
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258 int totallen = 0;
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259
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260 for (i = 0; i < len; i++)
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261 {
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262 tmp = data[i];
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263 curavg += tmp * tmp;
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264 }
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265 curavg = sqrt(curavg / (float) len);
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266
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267 // Evaluate an adequate 'mul' coefficient based on previous state, current
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268 // samples level, etc
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269 for (i = 0; i < NSAMPLES; i++)
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270 {
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271 avg += s->mem[i].avg * (float)s->mem[i].len;
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272 totallen += s->mem[i].len;
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273 }
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274
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275 if (totallen > MIN_SAMPLE_SIZE)
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276 {
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277 avg /= (float)totallen;
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278 if (avg >= SIL_FLOAT)
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279 {
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280 s->mul = s->mid_float / avg;
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281 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
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282 }
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283 }
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284
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285 // Scale & clamp the samples
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286 for (i = 0; i < len; i++)
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287 data[i] *= s->mul;
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288
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289 // Evaulation of newavg (not 100% accurate because of values clamping)
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290 newavg = s->mul * curavg;
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291
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292 // Stores computed values for future smoothing
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293 s->mem[s->idx].len = len;
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294 s->mem[s->idx].avg = newavg;
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295 s->idx = (s->idx + 1) % NSAMPLES;
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296 }
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297
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298 // Filter data through filter
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299 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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300 {
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301 af_volnorm_t *s = af->setup;
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302
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303 if(af->data->format == (AF_FORMAT_S16_NE))
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304 {
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305 if (s->method)
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306 method2_int16(s, data);
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307 else
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308 method1_int16(s, data);
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309 }
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310 else if(af->data->format == (AF_FORMAT_FLOAT_NE))
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311 {
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312 if (s->method)
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313 method2_float(s, data);
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314 else
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315 method1_float(s, data);
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316 }
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317 return data;
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318 }
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319
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320 // Allocate memory and set function pointers
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321 static int af_open(af_instance_t* af){
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322 int i = 0;
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323 af->control=control;
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324 af->uninit=uninit;
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325 af->play=play;
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326 af->mul=1;
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327 af->data=calloc(1,sizeof(af_data_t));
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328 af->setup=calloc(1,sizeof(af_volnorm_t));
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329 if(af->data == NULL || af->setup == NULL)
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330 return AF_ERROR;
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331
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332 ((af_volnorm_t*)af->setup)->mul = MUL_INIT;
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333 ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET;
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334 ((af_volnorm_t*)af->setup)->idx = 0;
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335 ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET;
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336 ((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET;
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337 for (i = 0; i < NSAMPLES; i++)
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338 {
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339 ((af_volnorm_t*)af->setup)->mem[i].len = 0;
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340 ((af_volnorm_t*)af->setup)->mem[i].avg = 0;
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341 }
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342 return AF_OK;
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343 }
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344
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345 // Description of this filter
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346 af_info_t af_info_volnorm = {
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347 "Volume normalizer filter",
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348 "volnorm",
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349 "Alex Beregszaszi & Pierre Lombard",
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350 "",
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351 AF_FLAGS_NOT_REENTRANT,
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352 af_open
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353 };