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

Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067
Signed-off-by: Glenn Kasten <gkasten@google.com>

1 files changed, 109 insertions, 0 deletions

diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
new file mode 100644
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+++ b/tools/resampler_tools/fir.cpp
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+/*
+ * 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;
+ }
+ 
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