Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
index cc3d509..27a9b05 100644
--- a/tools/resampler_tools/fir.cpp
+++ b/tools/resampler_tools/fir.cpp
@@ -20,15 +20,25 @@
#include <stdlib.h>
#include <string.h>
-static double sinc(double x) {
+static inline double sinc(double x) {
if (fabs(x) == 0.0f) return 1.0f;
return sin(x) / x;
}
-static double sqr(double 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(y * 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;
@@ -54,11 +64,12 @@
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"
+ "usage: %s [-h] [-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] [-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"
" -p generate poly-phase filter coefficients, with sample increment M/N\n"
@@ -66,6 +77,7 @@
" -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-point or floating-point (fixed)\n"
" -b kaiser window parameter beta (7.865 [-80dB])\n"
" -v attenuation in dBFS (0)\n",
@@ -77,8 +89,7 @@
int main(int argc, char** argv)
{
// nc is the number of bits to store the coefficients
- const int nc = 32;
-
+ int nc = 32;
bool polyphase = false;
unsigned int polyM = 160;
unsigned int polyN = 147;
@@ -88,7 +99,6 @@
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:
@@ -104,7 +114,6 @@
// 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
@@ -123,7 +132,6 @@
// 100 dB 10.056
double beta = 7.865;
-
// 2*nzc = (A - 8) / (2.285 * dw)
// with dw the transition width = 2*pi*dF/Fs
//
@@ -148,8 +156,9 @@
// nzc = 20
//
+ int M = 1 << 4; // number of phases for interpolation
int ch;
- while ((ch = getopt(argc, argv, ":hds:c:n:f:l:b:p:v:")) != -1) {
+ while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:")) != -1) {
switch (ch) {
case 'd':
debug = true;
@@ -169,13 +178,26 @@
case 'n':
nzc = atoi(optarg);
break;
+ case 'm':
+ M = atoi(optarg);
+ break;
case 'l':
- nz = atoi(optarg);
+ M = 1 << atoi(optarg);
break;
case 'f':
- if (!strcmp(optarg,"fixed")) format = 0;
- else if (!strcmp(optarg,"float")) format = 1;
- else usage(argv[0]);
+ 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);
@@ -193,11 +215,14 @@
// 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;
+ // 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: ");
@@ -207,7 +232,7 @@
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_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);
@@ -224,7 +249,7 @@
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 x = (2.0 * M_PI * ix * Fcr) / M;
double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;
y *= atten;
@@ -232,11 +257,13 @@
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));
+ 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,");
}
@@ -254,9 +281,12 @@
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));
+ 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");
}
@@ -277,5 +307,3 @@
}
// http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html
-
-