This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler.

# Copyright 2005 Google Inc. All Rights Reserved.

#

# Makefile for resampler_tools 

#

LOCAL_PATH:= $(call my-dir)

include $(CLEAR_VARS)

LOCAL_SRC_FILES := \

	fir.cpp

LOCAL_TARGET := fir

include $(BUILD_HOST_EXECUTABLE)

/*

 * Copyright (C) 2007 Google Inc.

 *

 * 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>

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 alpha) {

    if (k < 0 || k > N)

        return 0;

    return I0(M_PI*alpha * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(M_PI*alpha);

}

int main(int argc, char** argv)

{

    // nc is the number of bits to store the coefficients

    int nc = 16;

    // ni is the minimum number of bits needed for interpolation

    // (not used for generating the coefficients)

    const int ni = nc / 2;

    // nzc is the number of zero-crossing on one half of the filter

    int nzc = 12;

    // alpha parameter of the kaiser window

    // Larger numbers reduce ripples in the rejection band but increase

    // the width of the transition band. In reality there doesn't seem to be

    // a good reason to choose a big number because of the limited range

    // of our coefficients (16 bits).

    double alpha = 3.0;

    // cut off frequency ratio Fc/Fs

    double Fcr = 20000.0 / 44100.0;

    // 2^nz is the number coefficients per zero-crossing

    // (int theory this should be 1<<(nc/2))

    const int nz = 4;

    // total number of coefficients

    const int N = (1 << 4) * nzc;

    // generate the right half of the filter

    printf("const int32_t RESAMPLE_FIR_SIZE           = %d;\n", N);

    printf("const int32_t RESAMPLE_FIR_NUM_COEF       = %d;\n", nzc);

    printf("const int32_t RESAMPLE_FIR_COEF_BITS      = %d;\n", nc);

    printf("const int32_t RESAMPLE_FIR_LERP_FRAC_BITS = %d;\n", ni);

    printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS  = %d;\n", nz);

    printf("\n");

    printf("static int16_t resampleFIR[%d] = {", N);

    for (int i=0 ; i<N ; i++)

    {

        double x = (2.0 * M_PI * i * Fcr) / (1 << 4);

        double y = kaiser(i+N, 2*N, alpha) * sinc(x);

        int yi = floor(y * (1<<(nc-1)) + 0.5);

        if (yi >= (1<<(nc-1))) yi = (1<<(nc-1))-1;        

        //printf("%6d\n", yi);

        if ((i % (1 << 4)) == 0) printf("\n    ");

        printf("0x%04x, ", yi & 0xFFFF);

    }

    printf("\n};\n");

    return 0;

 }

 No newline at end of file