1 /* 2 * Copyright (C) 2007 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #include <math.h> 18 #include <stdio.h> 19 #include <unistd.h> 20 #include <stdlib.h> 21 #include <string.h> 22 23 static inline double sinc(double x) { 24 if (fabs(x) == 0.0f) return 1.0f; 25 return sin(x) / x; 26 } 27 28 static inline double sqr(double x) { 29 return x*x; 30 } 31 32 static inline int64_t toint(double x, int64_t maxval) { 33 int64_t v; 34 35 v = static_cast<int64_t>(floor(x * maxval + 0.5)); 36 if (v >= maxval) { 37 return maxval - 1; // error! 38 } 39 return v; 40 } 41 42 static double I0(double x) { 43 // from the Numerical Recipes in C p. 237 44 double ax,ans,y; 45 ax=fabs(x); 46 if (ax < 3.75) { 47 y=x/3.75; 48 y*=y; 49 ans=1.0+y*(3.5156229+y*(3.0899424+y*(1.2067492 50 +y*(0.2659732+y*(0.360768e-1+y*0.45813e-2))))); 51 } else { 52 y=3.75/ax; 53 ans=(exp(ax)/sqrt(ax))*(0.39894228+y*(0.1328592e-1 54 +y*(0.225319e-2+y*(-0.157565e-2+y*(0.916281e-2 55 +y*(-0.2057706e-1+y*(0.2635537e-1+y*(-0.1647633e-1 56 +y*0.392377e-2)))))))); 57 } 58 return ans; 59 } 60 61 static double kaiser(int k, int N, double beta) { 62 if (k < 0 || k > N) 63 return 0; 64 return I0(beta * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(beta); 65 } 66 67 static void usage(char* name) { 68 fprintf(stderr, 69 "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]" 70 " [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] [-l lerp]\n" 71 " %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]" 72 " [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] -p M/N\n" 73 " -h this help message\n" 74 " -d debug, print comma-separated coefficient table\n" 75 " -p generate poly-phase filter coefficients, with sample increment M/N\n" 76 " -s sample rate (48000)\n" 77 " -c cut-off frequency (20478)\n" 78 " -n number of zero-crossings on one side (8)\n" 79 " -l number of lerping bits (4)\n" 80 " -m number of polyphases (related to -l, default 16)\n" 81 " -f output format, can be fixed-point or floating-point (fixed)\n" 82 " -b kaiser window parameter beta (7.865 [-80dB])\n" 83 " -v attenuation in dBFS (0)\n", 84 name, name 85 ); 86 exit(0); 87 } 88 89 int main(int argc, char** argv) 90 { 91 // nc is the number of bits to store the coefficients 92 int nc = 32; 93 bool polyphase = false; 94 unsigned int polyM = 160; 95 unsigned int polyN = 147; 96 bool debug = false; 97 double Fs = 48000; 98 double Fc = 20478; 99 double atten = 1; 100 int format = 0; 101 102 // in order to keep the errors associated with the linear 103 // interpolation of the coefficients below the quantization error 104 // we must satisfy: 105 // 2^nz >= 2^(nc/2) 106 // 107 // for 16 bit coefficients that would be 256 108 // 109 // note that increasing nz only increases memory requirements, 110 // but doesn't increase the amount of computation to do. 111 // 112 // 113 // see: 114 // Smith, J.O. Digital Audio Resampling Home Page 115 // https://ccrma.stanford.edu/~jos/resample/, 2011-03-29 116 // 117 118 // | 0.1102*(A - 8.7) A > 50 119 // beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21) 21 <= A <= 50 120 // | 0 A < 21 121 // with A is the desired stop-band attenuation in dBFS 122 // 123 // for eg: 124 // 125 // 30 dB 2.210 126 // 40 dB 3.384 127 // 50 dB 4.538 128 // 60 dB 5.658 129 // 70 dB 6.764 130 // 80 dB 7.865 131 // 90 dB 8.960 132 // 100 dB 10.056 133 double beta = 7.865; 134 135 // 2*nzc = (A - 8) / (2.285 * dw) 136 // with dw the transition width = 2*pi*dF/Fs 137 // 138 int nzc = 8; 139 140 /* 141 * Example: 142 * 44.1 KHz to 48 KHz resampling 143 * 100 dB rejection above 28 KHz 144 * (the spectrum will fold around 24 KHz and we want 100 dB rejection 145 * at the point where the folding reaches 20 KHz) 146 * ...___|_____ 147 * | \| 148 * | ____/|\____ 149 * |/alias| \ 150 * ------/------+------\---------> KHz 151 * 20 24 28 152 * 153 * Transition band 8 KHz, or dw = 1.0472 154 * 155 * beta = 10.056 156 * nzc = 20 157 */ 158 159 int M = 1 << 4; // number of phases for interpolation 160 int ch; 161 while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:")) != -1) { 162 switch (ch) { 163 case 'd': 164 debug = true; 165 break; 166 case 'p': 167 if (sscanf(optarg, "%u/%u", &polyM, &polyN) != 2) { 168 usage(argv[0]); 169 } 170 polyphase = true; 171 break; 172 case 's': 173 Fs = atof(optarg); 174 break; 175 case 'c': 176 Fc = atof(optarg); 177 break; 178 case 'n': 179 nzc = atoi(optarg); 180 break; 181 case 'm': 182 M = atoi(optarg); 183 break; 184 case 'l': 185 M = 1 << atoi(optarg); 186 break; 187 case 'f': 188 if (!strcmp(optarg, "fixed")) { 189 format = 0; 190 } 191 else if (!strcmp(optarg, "fixed16")) { 192 format = 0; 193 nc = 16; 194 } 195 else if (!strcmp(optarg, "float")) { 196 format = 1; 197 } 198 else { 199 usage(argv[0]); 200 } 201 break; 202 case 'b': 203 beta = atof(optarg); 204 break; 205 case 'v': 206 atten = pow(10, -fabs(atof(optarg))*0.05 ); 207 break; 208 case 'h': 209 default: 210 usage(argv[0]); 211 break; 212 } 213 } 214 215 // cut off frequency ratio Fc/Fs 216 double Fcr = Fc / Fs; 217 218 // total number of coefficients (one side) 219 220 const int N = M * nzc; 221 222 // lerp (which is most useful if M is a power of 2) 223 224 int nz = 0; // recalculate nz as the bits needed to represent M 225 for (int i = M-1 ; i; i>>=1, nz++); 226 // generate the right half of the filter 227 if (!debug) { 228 printf("// cmd-line: "); 229 for (int i=1 ; i<argc ; i++) { 230 printf("%s ", argv[i]); 231 } 232 printf("\n"); 233 if (!polyphase) { 234 printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N); 235 printf("const int32_t RESAMPLE_FIR_INT_PHASES = %d;\n", M); 236 printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc); 237 } else { 238 printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", 2*nzc*polyN); 239 printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", 2*nzc); 240 } 241 if (!format) { 242 printf("const int32_t RESAMPLE_FIR_COEF_BITS = %d;\n", nc); 243 } 244 printf("\n"); 245 printf("static %s resampleFIR[] = {", !format ? "int32_t" : "float"); 246 } 247 248 if (!polyphase) { 249 for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation 250 for (int j=0 ; j<nzc ; j++) { 251 int ix = j*M + i; 252 double x = (2.0 * M_PI * ix * Fcr) / M; 253 double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr; 254 y *= atten; 255 256 if (!debug) { 257 if (j == 0) 258 printf("\n "); 259 } 260 if (!format) { 261 int64_t yi = toint(y, 1ULL<<(nc-1)); 262 if (nc > 16) { 263 printf("0x%08x, ", int32_t(yi)); 264 } else { 265 printf("0x%04x, ", int32_t(yi)&0xffff); 266 } 267 } else { 268 printf("%.9g%s ", y, debug ? "," : "f,"); 269 } 270 } 271 } 272 } else { 273 for (unsigned int j=0 ; j<polyN ; j++) { 274 // calculate the phase 275 double p = ((polyM*j) % polyN) / double(polyN); 276 if (!debug) printf("\n "); 277 else printf("\n"); 278 // generate a FIR per phase 279 for (int i=-nzc ; i<nzc ; i++) { 280 double x = 2.0 * M_PI * Fcr * (i + p); 281 double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;; 282 y *= atten; 283 if (!format) { 284 int64_t yi = toint(y, 1ULL<<(nc-1)); 285 if (nc > 16) { 286 printf("0x%08x, ", int32_t(yi)); 287 } else { 288 printf("0x%04x, ", int32_t(yi)&0xffff); 289 } 290 } else { 291 printf("%.9g%s", y, debug ? "" : "f"); 292 } 293 294 if (debug && (i==nzc-1)) { 295 } else { 296 printf(", "); 297 } 298 } 299 } 300 } 301 302 if (!debug) { 303 printf("\n};"); 304 } 305 printf("\n"); 306 return 0; 307 } 308 309 // http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html 310
