Remove obsolete directory tools/resampler_tools/
Current resampler is located in media/libaudioprocessing/
Test: builds OK
Change-Id: I65ff763662f38db4d70852d23d1066d68c823b5b
diff --git a/tools/OWNERS b/tools/OWNERS
deleted file mode 100644
index f9cb567..0000000
--- a/tools/OWNERS
+++ /dev/null
@@ -1 +0,0 @@
-gkasten@google.com
diff --git a/tools/resampler_tools/Android.bp b/tools/resampler_tools/Android.bp
deleted file mode 100644
index 7549359..0000000
--- a/tools/resampler_tools/Android.bp
+++ /dev/null
@@ -1,15 +0,0 @@
-// Copyright 2005 The Android Open Source Project
-//
-// Android.mk for resampler_tools
-//
-
-cc_binary_host {
- name: "fir",
-
- srcs: ["fir.cpp"],
-
- cflags: [
- "-Werror",
- "-Wall",
- ],
-}
diff --git a/tools/resampler_tools/OWNERS b/tools/resampler_tools/OWNERS
deleted file mode 100644
index b4a6798..0000000
--- a/tools/resampler_tools/OWNERS
+++ /dev/null
@@ -1 +0,0 @@
-hunga@google.com
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
deleted file mode 100644
index fe4d212..0000000
--- a/tools/resampler_tools/fir.cpp
+++ /dev/null
@@ -1,318 +0,0 @@
-/*
- * 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