/*
* 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 inline double sinc(double x) {
if (fabs(x) == 0.0f) return 1.0f;
return sin(x) / x;
}
static inline double sqr(double x) {
return x*x;
}
static inline int64_t toint(double x, int64_t maxval) {
int64_t v;
v = static_cast<int64_t>(floor(x * maxval + 0.5));
if (v >= maxval) {
return maxval - 1; // error!
}
return v;
}
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] [-D] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
" [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] [-l lerp]\n"
" %s [-h] [-d] [-D] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
" [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] -p M/N\n"
" -h this help message\n"
" -d debug, print comma-separated coefficient table\n"
" -D generate extra declarations\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"
" -m number of polyphases (related to -l, default 16)\n"
" -f output format, can be fixed, fixed16, or float (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
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; // 0=fixed, 1=float
bool declarations = false;
// 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
//
// | 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 M = 1 << 4; // number of phases for interpolation
int ch;
while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:D")) != -1) {
switch (ch) {
case 'd':
debug = true;
break;
case 'D':
declarations = 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 'm':
M = atoi(optarg);
break;
case 'l':
M = 1 << atoi(optarg);
break;
case 'f':
if (!strcmp(optarg, "fixed")) {
format = 0;
}
else if (!strcmp(optarg, "fixed16")) {
format = 0;
nc = 16;
}
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 N = M * nzc;
// lerp (which is most useful if M is a power of 2)
int nz = 0; // recalculate nz as the bits needed to represent M
for (int i = M-1 ; i; i>>=1, nz++);
// generate the right half of the filter
if (!debug) {
printf("// cmd-line:");
for (int i=0 ; i<argc ; i++) {
printf(" %s", argv[i]);
}
printf("\n");
if (declarations) {
if (!polyphase) {
printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N);
printf("const int32_t RESAMPLE_FIR_INT_PHASES = %d;\n", M);
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) / M;
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 = toint(y, 1ULL<<(nc-1));
if (nc > 16) {
printf("0x%08x,", int32_t(yi));
} else {
printf("0x%04x,", int32_t(yi)&0xffff);
}
} else {
printf("%.9g%s", y, debug ? "," : "f,");
}
if (j != nzc-1) {
printf(" ");
}
}
}
} else {
for (unsigned 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 = toint(y, 1ULL<<(nc-1));
if (nc > 16) {
printf("0x%08x,", int32_t(yi));
} else {
printf("0x%04x,", int32_t(yi)&0xffff);
}
} else {
printf("%.9g%s", y, debug ? "," : "f,");
}
if (i != nzc-1) {
printf(" ");
}
}
}
}
if (!debug && declarations) {
printf("\n};");
}
printf("\n");
return 0;
}
// http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html