7568
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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
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13 #define PLUGIN
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14
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15 #include <stdio.h>
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16 #include <stdlib.h>
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17 #include <unistd.h>
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18 #include <inttypes.h>
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19
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20 #include "../config.h"
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21 #include "../mp_msg.h"
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22 #include "../libao2/afmt.h"
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23
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24 #include "af.h"
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25 #include "dsp.h"
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26
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27 /* Below definition selects the length of each poly phase component.
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28 Valid definitions are L8 and L16, where the number denotes the
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29 length of the filter. This definition affects the computational
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30 complexity (see play()), the performance (see filter.h) and the
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31 memory usage. The filterlenght is choosen to 8 if the machine is
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32 slow and to 16 if the machine is fast and has MMX.
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33 */
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34
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35 #if defined(HAVE_SSE) && !defined(HAVE_3DNOW) // This machine is slow
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36
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37 #define L 8 // Filter length
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38 // Unrolled loop to speed up execution
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39 #define FIR(x,w,y){ \
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40 int16_t a = (w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3]) >> 16; \
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41 int16_t b = (w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7]) >> 16; \
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42 (y[0]) = a+b; \
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43 }
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44
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45 #else /* Fast machine */
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46
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47 #define L 16
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48 // Unrolled loop to speed up execution
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49 #define FIR(x,w,y){ \
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50 int16_t a = (w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] ) >> 16; \
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51 int16_t b = (w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] ) >> 16; \
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52 int16_t c = (w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11]) >> 16; \
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53 int16_t d = (w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15]) >> 16; \
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54 y[0] = (a+b+c+d) >> 1; \
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55 }
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56
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57 #endif /* Fast machine */
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58
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59 // Macro to add data to circular que
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60 #define ADDQUE(xi,xq,in)\
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61 xq[xi]=xq[xi+L]=(*in);\
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62 xi=(--xi)&(L-1);
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63
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64
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65
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66 // local data
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67 typedef struct af_resample_s
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68 {
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69 int16_t* w; // Current filter weights
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70 int16_t** xq; // Circular buffers
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71 int16_t xi; // Index for circular buffers
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72 int16_t wi; // Index for w
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73 uint16_t i; // Number of new samples to put in x queue
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74 uint16_t dn; // Down sampling factor
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75 uint16_t up; // Up sampling factor
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76 } af_resample_t;
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77
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78 // Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
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79 inline int gcd(register int a, register int b)
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80 {
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81 register int r = min(a,b);
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82 a=max(a,b);
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83 b=r;
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84
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85 r=a%b;
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86 while(r!=0){
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87 a=b;
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88 b=r;
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89 r=a%b;
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90 }
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91 return b;
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92 }
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93
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94 static int upsample(af_data_t* c,af_data_t* l, af_resample_t* s)
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95 {
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96 uint16_t ci = l->nch; // Index for channels
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97 uint16_t len = 0; // Number of input samples
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98 uint16_t nch = l->nch; // Number of channels
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99 uint16_t inc = s->up/s->dn;
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100 uint16_t level = s->up%s->dn;
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101 uint16_t up = s->up;
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102 uint16_t dn = s->dn;
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103
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104 register int16_t* w = s->w;
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105 register uint16_t wi = 0;
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106 register uint16_t xi = 0;
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107
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108 // Index current channel
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109 while(ci--){
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110 // Temporary pointers
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111 register int16_t* x = s->xq[ci];
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112 register int16_t* in = ((int16_t*)c->audio)+ci;
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113 register int16_t* out = ((int16_t*)l->audio)+ci;
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114 int16_t* end = in+c->len/2; // Block loop end
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115 wi = s->wi; xi = s->xi;
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116
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117 while(in < end){
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118 register uint16_t i = inc;
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119 if(wi<level) i++;
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120
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121 ADDQUE(xi,x,in);
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122 in+=nch;
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123 while(i--){
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124 // Run the FIR filter
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125 FIR((&x[xi]),(&w[wi*L]),out);
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126 len++; out+=nch;
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127 // Update wi to point at the correct polyphase component
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128 wi=(wi+dn)%up;
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129 }
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130 }
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131 }
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132 // Save values that needs to be kept for next time
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133 s->wi = wi;
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134 s->xi = xi;
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135 return len;
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136 }
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137
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138 static int downsample(af_data_t* c,af_data_t* l, af_resample_t* s)
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139 {
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140 uint16_t ci = l->nch; // Index for channels
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141 uint16_t len = 0; // Number of output samples
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142 uint16_t nch = l->nch; // Number of channels
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143 uint16_t inc = s->dn/s->up;
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144 uint16_t level = s->dn%s->up;
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145 uint16_t up = s->up;
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146 uint16_t dn = s->dn;
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147
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148 register uint16_t i = 0;
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149 register uint16_t wi = 0;
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150 register uint16_t xi = 0;
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151
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152 // Index current channel
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153 while(ci--){
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154 // Temporary pointers
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155 register int16_t* x = s->xq[ci];
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156 register int16_t* in = ((int16_t*)c->audio)+ci;
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157 register int16_t* out = ((int16_t*)l->audio)+ci;
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158 register int16_t* end = in+c->len/2; // Block loop end
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159 i = s->i; wi = s->wi; xi = s->xi;
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160
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161 while(in < end){
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162
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163 ADDQUE(xi,x,in);
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164 in+=nch;
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165 if(!--i){
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166 // Run the FIR filter
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167 FIR((&x[xi]),(&s->w[wi*L]),out);
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168 len++; out+=nch;
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169
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170 // Update wi to point at the correct polyphase component
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171 wi=(wi+dn)%up;
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172
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173 // Insert i number of new samples in queue
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174 i = inc;
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175 if(wi<level) i++;
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176 }
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177 }
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178 }
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179 // Save values that needs to be kept for next time
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180 s->wi = wi;
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181 s->xi = xi;
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182 s->i = i;
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183
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184 return len;
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185 }
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186
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187 // Initialization and runtime control
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188 static int control(struct af_instance_s* af, int cmd, void* arg)
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189 {
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190 switch(cmd){
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191 case AF_CONTROL_REINIT:{
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192 af_resample_t* s = (af_resample_t*)af->setup;
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193 af_data_t* n = (af_data_t*)arg; // New configureation
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194 int i,d = 0;
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195 int rv = AF_OK;
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196
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197 // Make sure this filter isn't redundant
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198 if(af->data->rate == n->rate)
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199 return AF_DETACH;
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200
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201 // Create space for circular bufers (if nesessary)
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202 if(af->data->nch != n->nch){
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203 // First free the old ones
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204 if(s->xq){
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205 for(i=1;i<af->data->nch;i++)
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206 if(s->xq[i])
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207 free(s->xq[i]);
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208 free(s->xq);
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209 }
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210 // ... then create new
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211 s->xq = malloc(n->nch*sizeof(int16_t*));
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212 for(i=0;i<n->nch;i++)
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213 s->xq[i] = malloc(2*L*sizeof(int16_t));
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214 s->xi = 0;
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215 }
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216
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217 // Set parameters
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218 af->data->nch = n->nch;
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219 af->data->format = AFMT_S16_LE;
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220 af->data->bps = 2;
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221 if(af->data->format != n->format || af->data->bps != n->bps)
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222 rv = AF_FALSE;
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223 n->format = AFMT_S16_LE;
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224 n->bps = 2;
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225
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226 // Calculate up and down sampling factors
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227 d=gcd(af->data->rate,n->rate);
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228
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229 // Check if the the design needs to be redone
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230 if(s->up != af->data->rate/d || s->dn != n->rate/d){
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231 float* w;
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232 float* wt;
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233 float fc;
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234 int j;
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235 s->up = af->data->rate/d;
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236 s->dn = n->rate/d;
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237
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238 // Calculate cuttof frequency for filter
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239 fc = 1/(float)(max(s->up,s->dn));
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240 // Allocate space for polyphase filter bank and protptype filter
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241 w = malloc(sizeof(float) * s->up *L);
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242 if(NULL != s->w)
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243 free(s->w);
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244 s->w = malloc(L*s->up*sizeof(int16_t));
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245
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246 // Design prototype filter type using Kaiser window with beta = 10
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247 if(NULL == w || NULL == s->w ||
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248 -1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
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249 mp_msg(MSGT_AFILTER,MSGL_ERR,"[resample] Unable to design prototype filter.\n");
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250 return AF_ERROR;
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251 }
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252 // Copy data from prototype to polyphase filter
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253 wt=w;
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254 for(j=0;j<L;j++){//Columns
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255 for(i=0;i<s->up;i++){//Rows
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256 float t=(float)s->up*32767.0*(*wt);
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257 s->w[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
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258 wt++;
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259 }
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260 }
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261 free(w);
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262 mp_msg(MSGT_AFILTER,MSGL_V,"[resample] New filter designed up: %i down: %i\n", s->up, s->dn);
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263 }
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264
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265 // Set multiplier
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266 af->mul.n = s->up;
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267 af->mul.d = s->dn;
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268 return rv;
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269 }
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270 case AF_CONTROL_RESAMPLE:
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271 // Reinit must be called after this function has been called
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272
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273 // Sanity check
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274 if(((int*)arg)[0] <= 8000 || ((int*)arg)[0] > 192000){
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275 mp_msg(MSGT_AFILTER,MSGL_ERR,"[resample] The output sample frequency must be between 8kHz and 192kHz. Current value is %i \n",((int*)arg)[0]);
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276 return AF_ERROR;
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277 }
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278
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279 af->data->rate=((int*)arg)[0];
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280 mp_msg(MSGT_AFILTER,MSGL_STATUS,"[resample] Changing sample rate to %iHz\n",af->data->rate);
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281 return AF_OK;
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282 }
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283 return AF_UNKNOWN;
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284 }
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285
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286 // Deallocate memory
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287 static void uninit(struct af_instance_s* af)
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288 {
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289 if(af->data)
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290 free(af->data);
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291 }
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292
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293 // Filter data through filter
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294 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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295 {
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296 int len = 0; // Length of output data
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297 af_data_t* c = data; // Current working data
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298 af_data_t* l = af->data; // Local data
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299 af_resample_t* s = (af_resample_t*)af->setup;
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300
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301 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
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302 return NULL;
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303
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304 // Run resampling
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305 if(s->up>s->dn)
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306 len = upsample(c,l,s);
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307 else
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308 len = downsample(c,l,s);
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309
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310 // Set output data
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311 c->audio = l->audio;
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312 c->len = len*2;
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313 c->rate = l->rate;
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314
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315 return c;
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316 }
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317
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318 // Allocate memory and set function pointers
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319 static int open(af_instance_t* af){
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320 af->control=control;
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321 af->uninit=uninit;
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322 af->play=play;
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323 af->mul.n=1;
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324 af->mul.d=1;
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325 af->data=calloc(1,sizeof(af_data_t));
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326 af->setup=calloc(1,sizeof(af_resample_t));
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327 if(af->data == NULL || af->setup == NULL)
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328 return AF_ERROR;
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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 plugin
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333 af_info_t af_info_resample = {
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334 "Sample frequency conversion",
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335 "resample",
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336 "Anders",
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337 "",
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338 open
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339 };
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340
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