diff --git a/tools/resampler_tools/Android.mk b/tools/resampler_tools/Android.mk
new file mode 100644
index 0000000..e8cbe39
--- /dev/null
+++ b/tools/resampler_tools/Android.mk
@@ -0,0 +1,17 @@
+# Copyright 2005 The Android Open Source Project
+#
+# Android.mk for resampler_tools 
+#
+
+
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+	fir.cpp
+
+LOCAL_MODULE := fir
+
+include $(BUILD_HOST_EXECUTABLE)
+
+
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
new file mode 100644
index 0000000..cc3d509
--- /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
+
+