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Diffstat (limited to 'tools/resampler_tools/fir.cpp')
| -rw-r--r-- | tools/resampler_tools/fir.cpp | 281 |
1 files changed, 281 insertions, 0 deletions
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp new file mode 100644 index 00000000..cc3d509a --- /dev/null +++ b/tools/resampler_tools/fir.cpp @@ -0,0 +1,281 @@ +/* + * Copyright (C) 2007 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include <math.h> +#include <stdio.h> +#include <unistd.h> +#include <stdlib.h> +#include <string.h> + +static double sinc(double x) { + if (fabs(x) == 0.0f) return 1.0f; + return sin(x) / x; +} + +static double sqr(double x) { + return x*x; +} + +static double I0(double x) { + // from the Numerical Recipes in C p. 237 + double ax,ans,y; + ax=fabs(x); + if (ax < 3.75) { + y=x/3.75; + y*=y; + ans=1.0+y*(3.5156229+y*(3.0899424+y*(1.2067492 + +y*(0.2659732+y*(0.360768e-1+y*0.45813e-2))))); + } else { + y=3.75/ax; + ans=(exp(ax)/sqrt(ax))*(0.39894228+y*(0.1328592e-1 + +y*(0.225319e-2+y*(-0.157565e-2+y*(0.916281e-2 + +y*(-0.2057706e-1+y*(0.2635537e-1+y*(-0.1647633e-1 + +y*0.392377e-2)))))))); + } + return ans; +} + +static double kaiser(int k, int N, double beta) { + if (k < 0 || k > N) + return 0; + return I0(beta * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(beta); +} + + +static void usage(char* name) { + fprintf(stderr, + "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] [-l lerp]\n" + " %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] -p M/N\n" + " -h this help message\n" + " -d debug, print comma-separated coefficient table\n" + " -p generate poly-phase filter coefficients, with sample increment M/N\n" + " -s sample rate (48000)\n" + " -c cut-off frequency (20478)\n" + " -n number of zero-crossings on one side (8)\n" + " -l number of lerping bits (4)\n" + " -f output format, can be fixed-point or floating-point (fixed)\n" + " -b kaiser window parameter beta (7.865 [-80dB])\n" + " -v attenuation in dBFS (0)\n", + name, name + ); + exit(0); +} + +int main(int argc, char** argv) +{ + // nc is the number of bits to store the coefficients + const int nc = 32; + + bool polyphase = false; + unsigned int polyM = 160; + unsigned int polyN = 147; + bool debug = false; + double Fs = 48000; + double Fc = 20478; + double atten = 1; + int format = 0; + + + // in order to keep the errors associated with the linear + // interpolation of the coefficients below the quantization error + // we must satisfy: + // 2^nz >= 2^(nc/2) + // + // for 16 bit coefficients that would be 256 + // + // note that increasing nz only increases memory requirements, + // but doesn't increase the amount of computation to do. + // + // + // see: + // Smith, J.O. Digital Audio Resampling Home Page + // https://ccrma.stanford.edu/~jos/resample/, 2011-03-29 + // + int nz = 4; + + // | 0.1102*(A - 8.7) A > 50 + // beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21) 21 <= A <= 50 + // | 0 A < 21 + // with A is the desired stop-band attenuation in dBFS + // + // for eg: + // + // 30 dB 2.210 + // 40 dB 3.384 + // 50 dB 4.538 + // 60 dB 5.658 + // 70 dB 6.764 + // 80 dB 7.865 + // 90 dB 8.960 + // 100 dB 10.056 + double beta = 7.865; + + + // 2*nzc = (A - 8) / (2.285 * dw) + // with dw the transition width = 2*pi*dF/Fs + // + int nzc = 8; + + // + // Example: + // 44.1 KHz to 48 KHz resampling + // 100 dB rejection above 28 KHz + // (the spectrum will fold around 24 KHz and we want 100 dB rejection + // at the point where the folding reaches 20 KHz) + // ...___|_____ + // | \| + // | ____/|\____ + // |/alias| \ + // ------/------+------\---------> KHz + // 20 24 28 + + // Transition band 8 KHz, or dw = 1.0472 + // + // beta = 10.056 + // nzc = 20 + // + + int ch; + while ((ch = getopt(argc, argv, ":hds:c:n:f:l:b:p:v:")) != -1) { + switch (ch) { + case 'd': + debug = true; + break; + case 'p': + if (sscanf(optarg, "%u/%u", &polyM, &polyN) != 2) { + usage(argv[0]); + } + polyphase = true; + break; + case 's': + Fs = atof(optarg); + break; + case 'c': + Fc = atof(optarg); + break; + case 'n': + nzc = atoi(optarg); + break; + case 'l': + nz = atoi(optarg); + break; + case 'f': + if (!strcmp(optarg,"fixed")) format = 0; + else if (!strcmp(optarg,"float")) format = 1; + else usage(argv[0]); + break; + case 'b': + beta = atof(optarg); + break; + case 'v': + atten = pow(10, -fabs(atof(optarg))*0.05 ); + break; + case 'h': + default: + usage(argv[0]); + break; + } + } + + // cut off frequency ratio Fc/Fs + double Fcr = Fc / Fs; + + + // total number of coefficients (one side) + const int M = (1 << nz); + const int N = M * nzc; + + // generate the right half of the filter + if (!debug) { + printf("// cmd-line: "); + for (int i=1 ; i<argc ; i++) { + printf("%s ", argv[i]); + } + printf("\n"); + if (!polyphase) { + printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N); + printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS = %d;\n", nz); + printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc); + } else { + printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", 2*nzc*polyN); + printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", 2*nzc); + } + if (!format) { + printf("const int32_t RESAMPLE_FIR_COEF_BITS = %d;\n", nc); + } + printf("\n"); + printf("static %s resampleFIR[] = {", !format ? "int32_t" : "float"); + } + + if (!polyphase) { + for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation + for (int j=0 ; j<nzc ; j++) { + int ix = j*M + i; + double x = (2.0 * M_PI * ix * Fcr) / (1 << nz); + double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr; + y *= atten; + + if (!debug) { + if (j == 0) + printf("\n "); + } + + if (!format) { + int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5); + if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1; + printf("0x%08x, ", int32_t(yi)); + } else { + printf("%.9g%s ", y, debug ? "," : "f,"); + } + } + } + } else { + for (int j=0 ; j<polyN ; j++) { + // calculate the phase + double p = ((polyM*j) % polyN) / double(polyN); + if (!debug) printf("\n "); + else printf("\n"); + // generate a FIR per phase + for (int i=-nzc ; i<nzc ; i++) { + double x = 2.0 * M_PI * Fcr * (i + p); + double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;; + y *= atten; + if (!format) { + int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5); + if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1; + printf("0x%08x", int32_t(yi)); + } else { + printf("%.9g%s", y, debug ? "" : "f"); + } + + if (debug && (i==nzc-1)) { + } else { + printf(", "); + } + } + } + } + + if (!debug) { + printf("\n};"); + } + printf("\n"); + return 0; +} + +// http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html + + |