annotate libaf/af_resample.h @ 26773:7763b34ba58e

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author ben
date Sun, 18 May 2008 14:55:26 +0000
parents 32e21d1beb48
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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 General Public
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4 // license. See http://www.gnu.org/copyleft/gpl.html for details.
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5 //
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6 // Copyright 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 file contains the resampling engine, the sample format is
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12 controlled by the FORMAT parameter, the filter length by the L
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13 parameter and the resampling type by UP and DN. This file should
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14 only be included by af_resample.c
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15 */
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16
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17 /* This header intentionally has no multiple inclusion guards. It is meant to
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18 * be included multiple times and generates different code depending on the
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19 * value of certain #defines. */
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20
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21 #undef L
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22 #undef SHIFT
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23 #undef FORMAT
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24 #undef FIR
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25 #undef ADDQUE
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26
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27 /* The length Lxx definition selects the length of each poly phase
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28 component. Valid definitions are L8 and L16 where the number
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29 defines the nuber of taps. This definition affects the
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30 computational complexity, the performance and the memory usage.
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31 */
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32
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33 /* The FORMAT_x parameter selects the sample format type currently
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34 float and int16 are supported. Thes two formats are selected by
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35 defining eiter FORMAT_F or FORMAT_I. The advantage of using float
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36 is that the amplitude and therefore the SNR isn't affected by the
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37 filtering, the disadvantage is that it is a lot slower.
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38 */
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39
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40 #if defined(FORMAT_I)
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41 #define SHIFT >>16
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42 #define FORMAT int16_t
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43 #else
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44 #define SHIFT
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45 #define FORMAT float
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46 #endif
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47
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48 // Short filter
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49 #if defined(L8)
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50
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51 #define L 8 // Filter length
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52 // Unrolled loop to speed up execution
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53 #define FIR(x,w,y) \
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54 (y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
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55 + w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT
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56
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57
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58
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59 #else /* L8/L16 */
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60
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61 #define L 16
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62 // Unrolled loop to speed up execution
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63 #define FIR(x,w,y) \
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64 y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
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65 + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
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66 + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
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67 + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) SHIFT
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68
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69 #endif /* L8/L16 */
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70
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71 // Macro to add data to circular que
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72 #define ADDQUE(xi,xq,in)\
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73 xq[xi]=xq[(xi)+L]=*(in);\
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74 xi=((xi)-1)&(L-1);
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75
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76 #if defined(UP)
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77
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78 uint32_t ci = l->nch; // Index for channels
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79 uint32_t nch = l->nch; // Number of channels
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80 uint32_t inc = s->up/s->dn;
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81 uint32_t level = s->up%s->dn;
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82 uint32_t up = s->up;
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83 uint32_t dn = s->dn;
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84 uint32_t ns = c->len/l->bps;
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85 register FORMAT* w = s->w;
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86
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87 register uint32_t wi = 0;
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88 register uint32_t xi = 0;
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89
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90 // Index current channel
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91 while(ci--){
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92 // Temporary pointers
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93 register FORMAT* x = s->xq[ci];
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94 register FORMAT* in = ((FORMAT*)c->audio)+ci;
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95 register FORMAT* out = ((FORMAT*)l->audio)+ci;
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96 FORMAT* end = in+ns; // Block loop end
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97 wi = s->wi; xi = s->xi;
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98
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99 while(in < end){
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100 register uint32_t i = inc;
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101 if(wi<level) i++;
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102
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103 ADDQUE(xi,x,in);
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104 in+=nch;
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105 while(i--){
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106 // Run the FIR filter
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107 FIR((&x[xi]),(&w[wi*L]),out);
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108 len++; out+=nch;
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109 // Update wi to point at the correct polyphase component
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110 wi=(wi+dn)%up;
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111 }
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112 }
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113
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114 }
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115 // Save values that needs to be kept for next time
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116 s->wi = wi;
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117 s->xi = xi;
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118 #endif /* UP */
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119
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120 #if defined(DN) /* DN */
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121 uint32_t ci = l->nch; // Index for channels
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122 uint32_t nch = l->nch; // Number of channels
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123 uint32_t inc = s->dn/s->up;
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124 uint32_t level = s->dn%s->up;
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125 uint32_t up = s->up;
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126 uint32_t dn = s->dn;
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127 uint32_t ns = c->len/l->bps;
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128 FORMAT* w = s->w;
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129
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130 register int32_t i = 0;
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131 register uint32_t wi = 0;
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132 register uint32_t xi = 0;
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133
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134 // Index current channel
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135 while(ci--){
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136 // Temporary pointers
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137 register FORMAT* x = s->xq[ci];
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138 register FORMAT* in = ((FORMAT*)c->audio)+ci;
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139 register FORMAT* out = ((FORMAT*)l->audio)+ci;
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140 register FORMAT* end = in+ns; // Block loop end
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141 i = s->i; wi = s->wi; xi = s->xi;
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142
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143 while(in < end){
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144
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145 ADDQUE(xi,x,in);
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146 in+=nch;
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147 if((--i)<=0){
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148 // Run the FIR filter
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149 FIR((&x[xi]),(&w[wi*L]),out);
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150 len++; out+=nch;
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151
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152 // Update wi to point at the correct polyphase component
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153 wi=(wi+dn)%up;
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154
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155 // Insert i number of new samples in queue
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156 i = inc;
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157 if(wi<level) i++;
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158 }
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159 }
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160 }
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161 // Save values that needs to be kept for next time
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162 s->wi = wi;
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163 s->xi = xi;
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164 s->i = i;
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165 #endif /* DN */