annotate libaf/af_resample.c @ 9675:62c5a17038ba

XML version of MPlayer's doc
author nicolas
date Sun, 23 Mar 2003 23:35:12 +0000
parents 295c20099f12
children 14090f7300a8
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1 /*=============================================================================
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2 //
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3 // This software has been released under the terms of the GNU 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 2002 Anders Johansson ajh@atri.curtin.edu.au
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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 /* This audio filter changes the sample rate. */
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12 #include <stdio.h>
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13 #include <stdlib.h>
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14 #include <unistd.h>
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15 #include <inttypes.h>
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16
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17 #include "af.h"
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18 #include "dsp.h"
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19
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20 /* Below definition selects the length of each poly phase component.
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21 Valid definitions are L8 and L16, where the number denotes the
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22 length of the filter. This definition affects the computational
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23 complexity (see play()), the performance (see filter.h) and the
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24 memory usage. The filterlenght is choosen to 8 if the machine is
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25 slow and to 16 if the machine is fast and has MMX.
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26 */
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27
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28 #if !defined(HAVE_MMX) // This machine is slow
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29 #define L8
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30 #else
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31 #define L16
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32 #endif
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33
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34 #include "af_resample.h"
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35
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36 // Filtering types
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37 #define RSMP_LIN (0<<0) // Linear interpolation
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38 #define RSMP_INT (1<<0) // 16 bit integer
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39 #define RSMP_FLOAT (2<<0) // 32 bit floating point
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40 #define RSMP_MASK (3<<0)
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41
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42 // Defines for sloppy or exact resampling
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43 #define FREQ_SLOPPY (0<<2)
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44 #define FREQ_EXACT (1<<2)
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45 #define FREQ_MASK (1<<2)
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46
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47 // Accuracy for linear interpolation
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48 #define STEPACCURACY 32
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49
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50 // local data
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51 typedef struct af_resample_s
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52 {
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53 void* w; // Current filter weights
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54 void** xq; // Circular buffers
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55 uint32_t xi; // Index for circular buffers
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56 uint32_t wi; // Index for w
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57 uint32_t i; // Number of new samples to put in x queue
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58 uint32_t dn; // Down sampling factor
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59 uint32_t up; // Up sampling factor
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60 uint64_t step; // Step size for linear interpolation
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61 uint64_t pt; // Pointer remainder for linear interpolation
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62 int setup; // Setup parameters cmdline or through postcreate
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63 } af_resample_t;
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64
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65 // Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
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66 static inline int gcd(register int a, register int b)
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67 {
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68 register int r = min(a,b);
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69 a=max(a,b);
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70 b=r;
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71
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72 r=a%b;
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73 while(r!=0){
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74 a=b;
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75 b=r;
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76 r=a%b;
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77 }
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78 return b;
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79 }
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80
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81 // Fast linear interpolation resample with modest audio quality
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82 static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
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83 {
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84 uint32_t len = 0; // Number of input samples
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85 uint32_t nch = l->nch; // Words pre transfer
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86 uint64_t step = s->step;
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87 int16_t* in16 = ((int16_t*)c->audio);
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88 int16_t* out16 = ((int16_t*)l->audio);
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89 int32_t* in32 = ((int32_t*)c->audio);
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90 int32_t* out32 = ((int32_t*)l->audio);
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91 uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
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92 uint64_t pt = s->pt;
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93 uint16_t tmp;
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94
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95 switch (nch){
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96 case 1:
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97 while(pt < end){
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98 out16[len++]=in16[pt>>STEPACCURACY];
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99 pt+=step;
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100 }
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101 s->pt=pt & ((1LL<<STEPACCURACY)-1);
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102 break;
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103 case 2:
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104 end/=2;
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105 while(pt < end){
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106 out32[len++]=in32[pt>>STEPACCURACY];
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107 pt+=step;
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108 }
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109 len=(len<<1);
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110 s->pt=pt & ((1LL<<STEPACCURACY)-1);
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111 break;
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112 default:
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113 end /=nch;
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114 while(pt < end){
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115 tmp=nch;
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116 do {
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117 tmp--;
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118 out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
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119 } while (tmp);
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120 len+=nch;
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121 pt+=step;
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122 }
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123 s->pt=pt & ((1LL<<STEPACCURACY)-1);
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124 }
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125 return len;
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126 }
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127
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128 /* Determine resampling type and format */
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129 static int set_types(struct af_instance_s* af, af_data_t* data)
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130 {
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131 af_resample_t* s = af->setup;
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132 int rv = AF_OK;
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133 float rd = 0;
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134
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135 // Make sure this filter isn't redundant
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136 if((af->data->rate == data->rate) || (af->data->rate == 0))
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137 return AF_DETACH;
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138
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139 /* If sloppy and small resampling difference (2%) */
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140 rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate;
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141 if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) &&
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142 (data->format != (AF_FORMAT_NE | AF_FORMAT_F))) ||
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143 ((s->setup & RSMP_MASK) == RSMP_LIN)){
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144 s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
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145 af->data->format = AF_FORMAT_NE | AF_FORMAT_SI;
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146 af->data->bps = 2;
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147 af_msg(AF_MSG_VERBOSE,"[resample] Using linear interpolation. \n");
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148 }
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149 else{
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150 /* If the input format is float or if float is explicitly selected
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151 use float, otherwise use int */
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152 if((data->format == (AF_FORMAT_NE | AF_FORMAT_F)) ||
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153 ((s->setup & RSMP_MASK) == RSMP_FLOAT)){
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154 s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
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155 af->data->format = AF_FORMAT_NE | AF_FORMAT_F;
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156 af->data->bps = 4;
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157 }
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158 else{
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159 s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
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160 af->data->format = AF_FORMAT_NE | AF_FORMAT_SI;
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161 af->data->bps = 2;
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162 }
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163 af_msg(AF_MSG_VERBOSE,"[resample] Using %s processing and %s frequecy"
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164 " conversion.\n",
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165 ((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer",
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166 ((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact");
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167 }
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168
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169 if(af->data->format != data->format || af->data->bps != data->bps)
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170 rv = AF_FALSE;
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171 data->format = af->data->format;
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172 data->bps = af->data->bps;
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173 af->data->nch = data->nch;
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174 return rv;
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175 }
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176
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177 // Initialization and runtime control
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178 static int control(struct af_instance_s* af, int cmd, void* arg)
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179 {
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180 switch(cmd){
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181 case AF_CONTROL_REINIT:{
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182 af_resample_t* s = (af_resample_t*)af->setup;
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183 af_data_t* n = (af_data_t*)arg; // New configureation
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184 int i,d = 0;
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185 int rv = AF_OK;
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186
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187 // Free space for circular bufers
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188 if(s->xq){
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189 for(i=1;i<af->data->nch;i++)
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190 if(s->xq[i])
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191 free(s->xq[i]);
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192 free(s->xq);
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193 }
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194
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195 if(AF_DETACH == (rv = set_types(af,n)))
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196 return AF_DETACH;
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197
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198 // If linear interpolation
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199 if((s->setup & RSMP_MASK) == RSMP_LIN){
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200 s->pt=0LL;
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201 s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
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202 af_msg(AF_MSG_DEBUG0,"[resample] Linear interpolation step: 0x%016X.\n",
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203 s->step);
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204 af->mul.n = af->data->rate;
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205 af->mul.d = n->rate;
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206 return rv;
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207 }
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208
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209 // Calculate up and down sampling factors
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210 d=gcd(af->data->rate,n->rate);
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211
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212 // If sloppy resampling is enabled limit the upsampling factor
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213 if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){
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214 int up=af->data->rate/2;
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215 int dn=n->rate/2;
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216 int m=2;
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217 while(af->data->rate/(d*m) > 5000){
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218 d=gcd(up,dn);
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219 up/=2; dn/=2; m*=2;
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220 }
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221 d*=m;
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222 }
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223
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224 // Create space for circular bufers
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225 s->xq = malloc(n->nch*sizeof(void*));
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226 for(i=0;i<n->nch;i++)
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227 s->xq[i] = malloc(2*L*af->data->bps);
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228 s->xi = 0;
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229
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230 // Check if the the design needs to be redone
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231 if(s->up != af->data->rate/d || s->dn != n->rate/d){
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232 float* w;
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233 float* wt;
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234 float fc;
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235 int j;
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236 s->up = af->data->rate/d;
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237 s->dn = n->rate/d;
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238
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239 // Calculate cuttof frequency for filter
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240 fc = 1/(float)(max(s->up,s->dn));
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241 // Allocate space for polyphase filter bank and protptype filter
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242 w = malloc(sizeof(float) * s->up *L);
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243 if(NULL != s->w)
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244 free(s->w);
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245 s->w = malloc(L*s->up*af->data->bps);
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246
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247 // Design prototype filter type using Kaiser window with beta = 10
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248 if(NULL == w || NULL == s->w ||
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249 -1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
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250 af_msg(AF_MSG_ERROR,"[resample] Unable to design prototype filter.\n");
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251 return AF_ERROR;
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252 }
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253 // Copy data from prototype to polyphase filter
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254 wt=w;
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255 for(j=0;j<L;j++){//Columns
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256 for(i=0;i<s->up;i++){//Rows
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257 if((s->setup & RSMP_MASK) == RSMP_INT){
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258 float t=(float)s->up*32767.0*(*wt);
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259 ((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
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260 }
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261 else
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262 ((float*)s->w)[i*L+j] = (float)s->up*(*wt);
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263 wt++;
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264 }
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265 }
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266 free(w);
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267 af_msg(AF_MSG_VERBOSE,"[resample] New filter designed up: %i "
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268 "down: %i\n", s->up, s->dn);
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269 }
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270
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fbd5445cc853 Adding function for calculating the delay caused by the filters
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271 // Set multiplier and delay
fbd5445cc853 Adding function for calculating the delay caused by the filters
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272 af->delay = (double)(1000*L/2)/((double)n->rate);
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273 af->mul.n = s->up;
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274 af->mul.d = s->dn;
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275 return rv;
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276 }
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277 case AF_CONTROL_COMMAND_LINE:{
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278 af_resample_t* s = (af_resample_t*)af->setup;
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279 int rate=0;
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280 int type=RSMP_INT;
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281 int sloppy=1;
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282 sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type);
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283 s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) |
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284 (clamp(type,RSMP_LIN,RSMP_FLOAT));
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285 return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
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286 }
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287 case AF_CONTROL_POST_CREATE:
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288 if((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
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289 ((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
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290 return AF_OK;
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291 case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
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292 // Reinit must be called after this function has been called
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293
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294 // Sanity check
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d09c125a88fa Changing frequency limit for resampling
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295 if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
8607
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296 af_msg(AF_MSG_ERROR,"[resample] The output sample frequency "
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297 "must be between 8kHz and 192kHz. Current value is %i \n",
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298 ((int*)arg)[0]);
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299 return AF_ERROR;
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300 }
d08513b9fed6 Adding new audio output filter layer libaf
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301
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302 af->data->rate=((int*)arg)[0];
8607
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303 af_msg(AF_MSG_VERBOSE,"[resample] Changing sample rate "
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304 "to %iHz\n",af->data->rate);
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305 return AF_OK;
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diff changeset
306 }
d08513b9fed6 Adding new audio output filter layer libaf
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307 return AF_UNKNOWN;
d08513b9fed6 Adding new audio output filter layer libaf
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parents:
diff changeset
308 }
d08513b9fed6 Adding new audio output filter layer libaf
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309
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310 // Deallocate memory
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311 static void uninit(struct af_instance_s* af)
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312 {
d08513b9fed6 Adding new audio output filter layer libaf
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313 if(af->data)
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314 free(af->data);
d08513b9fed6 Adding new audio output filter layer libaf
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315 }
d08513b9fed6 Adding new audio output filter layer libaf
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316
d08513b9fed6 Adding new audio output filter layer libaf
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diff changeset
317 // Filter data through filter
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diff changeset
318 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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319 {
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320 int len = 0; // Length of output data
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321 af_data_t* c = data; // Current working data
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322 af_data_t* l = af->data; // Local data
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323 af_resample_t* s = (af_resample_t*)af->setup;
d08513b9fed6 Adding new audio output filter layer libaf
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diff changeset
324
d08513b9fed6 Adding new audio output filter layer libaf
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325 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
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326 return NULL;
d08513b9fed6 Adding new audio output filter layer libaf
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327
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328 // Run resampling
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329 switch(s->setup & RSMP_MASK){
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330 case(RSMP_INT):
8607
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331 # define FORMAT_I 1
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332 if(s->up>s->dn){
d6f40a06867b Changes includes:
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333 # define UP
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334 # include "af_resample.h"
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335 # undef UP
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336 }
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337 else{
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diff changeset
338 # define DN
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diff changeset
339 # include "af_resample.h"
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340 # undef DN
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341 }
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diff changeset
342 break;
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diff changeset
343 case(RSMP_FLOAT):
8607
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344 # undef FORMAT_I
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345 # define FORMAT_F 1
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diff changeset
346 if(s->up>s->dn){
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diff changeset
347 # define UP
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diff changeset
348 # include "af_resample.h"
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349 # undef UP
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diff changeset
350 }
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diff changeset
351 else{
d6f40a06867b Changes includes:
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diff changeset
352 # define DN
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diff changeset
353 # include "af_resample.h"
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354 # undef DN
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355 }
d6f40a06867b Changes includes:
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diff changeset
356 break;
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diff changeset
357 case(RSMP_LIN):
8607
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diff changeset
358 len = linint(c, l, s);
d6f40a06867b Changes includes:
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359 break;
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360 }
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361
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362 // Set output data
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363 c->audio = l->audio;
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364 c->len = len*l->bps;
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365 c->rate = l->rate;
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366
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367 return c;
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368 }
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369
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370 // Allocate memory and set function pointers
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371 static int open(af_instance_t* af){
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372 af->control=control;
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373 af->uninit=uninit;
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374 af->play=play;
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375 af->mul.n=1;
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376 af->mul.d=1;
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377 af->data=calloc(1,sizeof(af_data_t));
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378 af->setup=calloc(1,sizeof(af_resample_t));
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379 if(af->data == NULL || af->setup == NULL)
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380 return AF_ERROR;
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381 ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
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382 return AF_OK;
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383 }
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384
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385 // Description of this plugin
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386 af_info_t af_info_resample = {
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387 "Sample frequency conversion",
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388 "resample",
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389 "Anders",
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390 "",
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391 AF_FLAGS_REENTRANT,
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392 open
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393 };
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394