/*
Code to generate wavedata for bandlimited waveforms.
Copyright 2011 David Robillard
Copyright 2003 Mike Rawes
This is free software: you can redistribute it and/or modify it
under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This software is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this software. If not, see .
*/
#include
#include
#include
#include "common.h"
#include "math_func.h"
#include "wavedata.h"
#include "wdatutil.h"
#ifdef __cplusplus
extern "C" {
#endif
void generate_sine(float* samples,
uint32_t sample_count);
void generate_sawtooth(float* samples,
uint32_t sample_count,
unsigned long harmonics,
float gibbs_comp);
void generate_square(float* samples,
uint32_t sample_count,
unsigned long harmonics,
float gibbs_comp);
void generate_parabola(float* samples,
uint32_t sample_count,
unsigned long harmonics,
float gibbs_comp);
#ifdef __cplusplus
} /* extern "C" { */
#endif
char* wave_names[] = {
"saw",
"square",
"parabola"
};
char* wave_descriptions[] = {
"Sawtooth Wave",
"Square Wave",
"Parabola Wave"
};
unsigned long wave_first_harmonics[] = {
1,
1,
1
};
unsigned long wave_harmonic_intervals[] = {
1,
2,
1
};
Wavedata*
wavedata_new(double sample_rate)
{
Wavedata* w;
w = (Wavedata*)malloc(sizeof(Wavedata));
if (!w) {
return 0;
}
w->data_handle = 0;
w->table_count = 0;
w->tables = 0;
w->lookup = 0;
w->lookup_max = 0;
w->sample_rate = (float)sample_rate;
w->nyquist = w->sample_rate * 0.5f;
return w;
}
void
wavedata_cleanup(Wavedata* w)
{
unsigned long ti;
Wavetable* t;
for (ti = 0; ti < w->table_count; ti++) {
t = w->tables[ti];
if (t) {
if (t->samples_hf) {
free(t->samples_hf);
}
if (t->samples_lf) {
free(t->samples_lf);
}
free(t);
}
}
free(w);
}
int
wavedata_add_table(Wavedata* w,
uint32_t sample_count,
unsigned long harmonics)
{
Wavetable** tables;
Wavetable* t;
size_t bytes;
t = (Wavetable*)malloc(sizeof(Wavetable));
if (!t) {
return -1;
}
/* Extra 3 samples for interpolation */
bytes = (sample_count + 3) * sizeof(float);
t->samples_lf = (float*)malloc(bytes);
if (!t->samples_lf) {
free(t);
return -1;
}
t->samples_hf = (float*)malloc(bytes);
if (!t->samples_hf) {
free(t->samples_lf);
free(t);
return -1;
}
bytes = (w->table_count + 1) * sizeof(Wavetable*);
if (w->table_count == 0) {
tables = (Wavetable**)malloc(bytes);
} else {
tables = (Wavetable**)realloc(w->tables, bytes);
}
if (!tables) {
free(t);
return -1;
}
t->sample_count = sample_count;
t->harmonics = harmonics;
if (w->lookup_max < harmonics) {
w->lookup_max = harmonics;
}
tables[w->table_count] = t;
w->tables = tables;
w->table_count++;
return 0;
}
void
wavedata_generate_tables(Wavedata* w,
Wavetype wavetype,
float gibbs_comp)
{
Wavetable* t;
float* samples_lf;
float* samples_hf;
unsigned long h_lf;
unsigned long h_hf;
unsigned long i;
for (i = 0; i < w->table_count; i++) {
t = w->tables[i];
h_lf = t->harmonics;
if (i < w->table_count - 1) {
h_hf = w->tables[i + 1]->harmonics;
} else {
h_hf = 1;
}
samples_lf = t->samples_lf;
samples_hf = t->samples_hf;
samples_lf++;
samples_hf++;
switch (wavetype) {
case SAW:
generate_sawtooth(samples_lf, t->sample_count, h_lf, gibbs_comp);
generate_sawtooth(samples_hf, t->sample_count, h_hf, gibbs_comp);
break;
case SQUARE:
generate_square(samples_lf, t->sample_count, h_lf, gibbs_comp);
generate_square(samples_hf, t->sample_count, h_hf, gibbs_comp);
break;
case PARABOLA:
generate_parabola(samples_lf, t->sample_count, h_lf, gibbs_comp);
generate_parabola(samples_hf, t->sample_count, h_hf, gibbs_comp);
break;
}
/* Basic denormalization */
for (uint32_t s = 0; s < t->sample_count; s++) {
samples_lf[s] = FABSF(samples_lf[s]) < SMALLEST_FLOAT ? 0.0 : samples_lf[s];
}
samples_lf--;
samples_lf[0] = samples_lf[t->sample_count];
samples_lf[t->sample_count + 1] = samples_hf[1];
samples_lf[t->sample_count + 2] = samples_hf[2];
for (uint32_t s = 0; s < t->sample_count; s++) {
samples_hf[s] = FABSF(samples_hf[s]) < SMALLEST_FLOAT ? 0.0 : samples_hf[s];
}
samples_hf--;
samples_hf[0] = samples_hf[t->sample_count];
samples_hf[t->sample_count + 1] = samples_hf[1];
samples_hf[t->sample_count + 2] = samples_hf[2];
}
}
int
wavedata_write(Wavedata* w,
FILE* wdat_fp,
const char* data_name)
{
Wavetable* t = 0;
unsigned long table_count;
unsigned long i;
unsigned long j;
unsigned long s;
int column;
/*
* Extra table at end
*/
table_count = w->table_count + 1;
fprintf(wdat_fp, "#include \"lv2/lv2plug.in/ns/lv2core/lv2.h\"\n");
fprintf(wdat_fp, "#include \n");
fprintf(wdat_fp, "#include \"wavedata.h\"\n");
fprintf(wdat_fp, "\n");
/*
* Fixed data and tables
*/
fprintf(wdat_fp, "unsigned long ref_count = 0;\n");
fprintf(wdat_fp, "unsigned long first_sample_rate = 0;\n");
fprintf(wdat_fp, "unsigned long table_count = %ld;\n", table_count);
fprintf(wdat_fp, "Wavetable tables[%ld];\n", table_count);
fprintf(wdat_fp, "Wavetable * ptables[%ld];\n", table_count);
fprintf(wdat_fp, "unsigned long lookup[%ld];\n", w->lookup_max + 1);
fprintf(wdat_fp, "unsigned long lookup_max = %ld;\n", w->lookup_max);
fprintf(wdat_fp, "\n");
/*
* Sample data
* Each table has an extra 3 samples for interpolation
*/
for (i = 0; i < w->table_count; i++) {
t = w->tables[i];
fprintf(wdat_fp, "static float samples_lf_%ld[%ld] = {\n", i, t->sample_count + 3);
column = 0;
for (s = 0; s < t->sample_count + 3 - 1; s++, column++) {
if (column == 5) {
fprintf(wdat_fp, "\n");
column = 0;
}
fprintf(wdat_fp, "%+.8ef,", t->samples_lf[s]);
}
if (column == 5) {
fprintf(wdat_fp, "\n");
}
fprintf(wdat_fp, "%+.8ef\n", t->samples_lf[s]);
fprintf(wdat_fp, "};\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "static float samples_hf_%ld[%ld] = {\n", i, t->sample_count + 3);
column = 0;
for (s = 0; s < t->sample_count + 3 - 1; s++, column++) {
if (column == 5) {
fprintf(wdat_fp, "\n");
column = 0;
}
fprintf(wdat_fp, "%+.8ef,", t->samples_hf[s]);
}
if (column == 5) {
fprintf(wdat_fp, "\n");
}
fprintf(wdat_fp, "%+.8ef\n", t->samples_hf[s]);
fprintf(wdat_fp, "};\n");
fprintf(wdat_fp, "\n");
}
fprintf(wdat_fp, "float samples_zero[%ld];\n", t->sample_count + 3);
fprintf(wdat_fp, "\n");
/*
* Function to get Wavedata - the sample rate is needed to calculate
* frequencies and related things
*/
fprintf(wdat_fp, "int\n");
fprintf(
wdat_fp,
"blop_get_%s (Wavedata * w, double sample_rate, const char* bundle_path, const LV2_Feature* const* features)\n",
data_name);
fprintf(wdat_fp, "{\n");
fprintf(wdat_fp, "\tWavetable * t;\n");
fprintf(wdat_fp, "\tunsigned long ti;\n");
fprintf(wdat_fp, "\n");
/*
* Sample rate must be > 0
*/
fprintf(wdat_fp, "\tif (sample_rate == 0)\n");
fprintf(wdat_fp, "\t\treturn -1;\n");
fprintf(wdat_fp, "\n");
/*
* First time call - set up all sample rate dependent data
*/
fprintf(wdat_fp, "\tif (first_sample_rate == 0)\n");
fprintf(wdat_fp, "\t{\n");
fprintf(wdat_fp, "\t\tfirst_sample_rate = sample_rate;\n");
fprintf(wdat_fp, "\t\tw->sample_rate = (float) sample_rate;\n");
fprintf(wdat_fp, "\t\tw->nyquist = w->sample_rate / 2.0f;\n");
fprintf(wdat_fp, "\t\tw->table_count = table_count;\n");
fprintf(wdat_fp, "\t\tw->tables = ptables;\n");
fprintf(wdat_fp, "\t\tw->lookup = lookup;\n");
fprintf(wdat_fp, "\t\tw->lookup_max = lookup_max;\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\t\tfor (ti = 1; ti < table_count - 1; ti++)\n");
fprintf(wdat_fp, "\t\t{\n");
fprintf(wdat_fp, "\t\t\tt = ptables[ti];\n");
fprintf(wdat_fp,
"\t\t\tt->min_frequency = w->nyquist / (float) (ptables[ti - 1]->harmonics);\n");
fprintf(wdat_fp, "\t\t\tt->max_frequency = w->nyquist / (float) (t->harmonics);\n");
fprintf(wdat_fp, "\t\t}\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\t\tt = w->tables[0];\n");
fprintf(wdat_fp, "\t\tt->min_frequency = 0.0f;\n");
fprintf(wdat_fp, "\t\tt->max_frequency = ptables[1]->min_frequency;\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\t\tt = ptables[table_count - 1];\n");
fprintf(wdat_fp, "\t\tt->min_frequency = ptables[w->table_count - 2]->max_frequency;\n");
fprintf(wdat_fp, "\t\tt->max_frequency = w->nyquist;\n");
fprintf(wdat_fp, "\t\n");
fprintf(wdat_fp, "\t\tfor (ti = 0; ti < w->table_count; ti++)\n");
fprintf(wdat_fp, "\t\t{\n");
fprintf(wdat_fp, "\t\t\tt = w->tables[ti];\n");
fprintf(wdat_fp, "\t\t\tt->phase_scale_factor = (float) (t->sample_count) / w->sample_rate;\n");
fprintf(wdat_fp,
"\t\t\tt->range_scale_factor = 1.0f / (t->max_frequency - t->min_frequency);\n");
fprintf(wdat_fp, "\t\t}\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\t\treturn 0;\n");
fprintf(wdat_fp, "\t}\n");
/*
* Already called at least once, so just set up wavedata
*/
fprintf(wdat_fp, "\telse if (sample_rate == first_sample_rate)\n");
fprintf(wdat_fp, "\t{\n");
fprintf(wdat_fp, "\t\tw->sample_rate = (float) sample_rate;\n");
fprintf(wdat_fp, "\t\tw->nyquist = w->sample_rate / 2.0f;\n");
fprintf(wdat_fp, "\t\tw->table_count = table_count;\n");
fprintf(wdat_fp, "\t\tw->tables = ptables;\n");
fprintf(wdat_fp, "\t\tw->lookup = lookup;\n");
fprintf(wdat_fp, "\t\tw->lookup_max = lookup_max;\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\t\treturn 0;\n");
fprintf(wdat_fp, "\t}\n");
/*
* Sample rate does not match, so fail
*
* NOTE: This means multiple sample rates are not supported
* This should not present any problems
*/
fprintf(wdat_fp, "\telse\n");
fprintf(wdat_fp, "\t{\n");
fprintf(wdat_fp, "\t\treturn -1;\n");
fprintf(wdat_fp, "\t}\n");
fprintf(wdat_fp, "}\n");
fprintf(wdat_fp, "\n");
/*
* _init()
* Assemble tables and lookup
*/
fprintf(wdat_fp, "void\n");
fprintf(wdat_fp, "__attribute__ ((constructor))\n");
fprintf(wdat_fp, "init (void)\n");
fprintf(wdat_fp, "{\n");
fprintf(wdat_fp, "\tunsigned long max_harmonic;\n");
fprintf(wdat_fp, "\tunsigned long ti;\n");
fprintf(wdat_fp, "\tunsigned long li;\n");
fprintf(wdat_fp, "\n");
for (i = 0; i < w->table_count; i++) {
t = w->tables[i];
fprintf(wdat_fp, "\ttables[%ld].sample_count = %ld;\n", i, t->sample_count);
fprintf(wdat_fp, "\ttables[%ld].samples_lf = samples_lf_%ld;\n", i, i);
fprintf(wdat_fp, "\ttables[%ld].samples_hf = samples_hf_%ld;\n", i, i);
fprintf(wdat_fp, "\ttables[%ld].harmonics = %ld;\n", i, t->harmonics);
fprintf(wdat_fp, "\n");
}
/*
* Last table - uses same sample data as previous table for lf data,
* and zeroes for hf data
*/
i = w->table_count - 1;
j = i + 1;
t = w->tables[i];
/*
* Zero silent samples
*/
fprintf(wdat_fp, "\tfor (uint32_t s = 0; s < %ld; s++)\n", t->sample_count + 3);
fprintf(wdat_fp, "\t\tsamples_zero[s] = 0.0f;\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\ttables[%ld].sample_count = %ld;\n", j, t->sample_count);
fprintf(wdat_fp, "\ttables[%ld].samples_lf = samples_hf_%ld;\n", j, i);
fprintf(wdat_fp, "\ttables[%ld].samples_hf = samples_zero;\n", j);
fprintf(wdat_fp, "\ttables[%ld].harmonics = 1;\n", j);
fprintf(wdat_fp, "\n");
/*
* Get pointers to each wavetable and put them in the pointer array
*/
fprintf(wdat_fp, "\tfor (ti = 0; ti < table_count; ti++)\n");
fprintf(wdat_fp, "\t\tptables[ti] = &tables[ti];\n");
fprintf(wdat_fp, "\n");
/*
* Shift all sample offsets forward by one sample
* !!! NO! Don't!
fprintf (wdat_fp, "\tfor (ti = 0; ti < table_count; ti++)\n");
fprintf (wdat_fp, "\t{\n");
fprintf (wdat_fp, "\t\tptables[ti]->samples_lf++;\n");
fprintf (wdat_fp, "\t\tptables[ti]->samples_hf++;\n");
fprintf (wdat_fp, "\t}\n");
fprintf (wdat_fp, "\n");
*/
/*
* Table lookup vector indexed by harmonic
* Add lookup data to vector
*/
fprintf(wdat_fp, "\tli = 0;");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\tfor (ti = table_count - 1; ti > 0; ti--)\n");
fprintf(wdat_fp, "\t{\n");
fprintf(wdat_fp, "\t\tmax_harmonic = ptables[ti]->harmonics;\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\t\tfor ( ; li <= max_harmonic; li++)\n");
fprintf(wdat_fp, "\t\t\tlookup[li] = ti;\n");
fprintf(wdat_fp, "\t}\n");
fprintf(wdat_fp, "\n");
fprintf(wdat_fp, "\tfor ( ; li <= lookup_max; li++)\n");
fprintf(wdat_fp, "\t\tlookup[li] = 0;\n");
fprintf(wdat_fp, "}\n");
return 0;
}
void
generate_sawtooth(float* samples,
uint32_t sample_count,
unsigned long harmonics,
float gibbs_comp)
{
double phase_scale = 2.0 * M_PI / (double)sample_count;
float scale = 2.0f / M_PI;
unsigned long i;
unsigned long h;
double mhf;
double hf;
double k;
double m;
double phase;
double partial;
if (gibbs_comp > 0.0f) {
/* Degree of Gibbs Effect compensation */
mhf = (double)harmonics;
k = M_PI * (double)gibbs_comp / mhf;
for (i = 0; i < sample_count; i++) {
samples[i] = 0.0f;
}
for (h = 1; h <= harmonics; h++) {
hf = (double)h;
/* Gibbs Effect compensation - Hamming window */
/* Modified slightly for smoother fade at highest frequencies */
m = 0.54 + 0.46 * cos((hf - 0.5 / mhf) * k);
for (i = 0; i < sample_count; i++) {
phase = (double)i * phase_scale;
partial = (m / hf) * sin(phase * hf);
samples[i] += (float)partial;
}
}
for (i = 0; i < sample_count; i++) {
samples[i] *= scale;
}
} else {
/* Allow overshoot */
for (i = 0; i < sample_count; i++) {
phase = (double)i * phase_scale;
samples[i] = 0.0f;
for (h = 1; h <= harmonics; h++) {
hf = (double)h;
partial = (1.0 / hf) * sin(phase * hf);
samples[i] += (float)partial;
}
samples[i] *= scale;
}
}
}
void
generate_square(float* samples,
uint32_t sample_count,
unsigned long harmonics,
float gibbs_comp)
{
double phase_scale = 2.0 * M_PI / (double)sample_count;
float scale = 4.0f / M_PI;
unsigned long i;
unsigned long h;
double mhf;
double hf;
double k;
double m;
double phase;
double partial;
if (gibbs_comp > 0.0f) {
/* Degree of Gibbs Effect compensation */
mhf = (double)harmonics;
k = M_PI * (double)gibbs_comp / mhf;
for (i = 0; i < sample_count; i++) {
samples[i] = 0.0f;
}
for (h = 1; h <= harmonics; h += 2) {
hf = (double)h;
/* Gibbs Effect compensation - Hamming window */
/* Modified slightly for smoother fade at highest frequencies */
m = 0.54 + 0.46 * cos((hf - 0.5 / pow(mhf, 2.2)) * k);
for (i = 0; i < sample_count; i++) {
phase = (double)i * phase_scale;
partial = (m / hf) * sin(phase * hf);
samples[i] += (float)partial;
}
}
for (i = 0; i < sample_count; i++) {
samples[i] *= scale;
}
} else {
/* Allow overshoot */
for (i = 0; i < sample_count; i++) {
phase = (double)i * phase_scale;
samples[i] = 0.0f;
for (h = 1; h <= harmonics; h += 2) {
hf = (double)h;
partial = (1.0 / hf) * sin(phase * hf);
samples[i] += (float)partial;
}
samples[i] *= scale;
}
}
}
void
generate_parabola(float* samples,
uint32_t sample_count,
unsigned long harmonics,
float gibbs_comp)
{
double phase_scale = 2.0 * M_PI / (double)sample_count;
float scale = 2.0f / (M_PI * M_PI);
unsigned long i;
unsigned long h;
//double mhf;
double hf;
//double k;
//double m;
double phase;
double partial;
double sign;
if (gibbs_comp > 0.0f) {
/* Degree of Gibbs Effect compensation */
//mhf = (double)harmonics;
//k = M_PI * (double)gibbs_comp / mhf;
for (i = 0; i < sample_count; i++) {
samples[i] = 0.0f;
}
sign = -1.0;
for (h = 1; h <= harmonics; h++) {
hf = (double)h;
/* Gibbs Effect compensation - Hamming window */
/* Modified slightly for smoother fade at highest frequencies */
//m = 0.54 + 0.46 * cos((hf - 0.5 / mhf) * k);
for (i = 0; i < sample_count; i++) {
phase = (double)i * phase_scale;
partial = (sign * 4.0 / (hf * hf)) * cos(phase * hf);
samples[i] += (float)partial;
}
sign = -sign;
}
for (i = 0; i < sample_count; i++) {
samples[i] *= scale;
}
} else {
/* Allow overshoot */
for (i = 0; i < sample_count; i++) {
phase = (double)i * phase_scale;
samples[i] = 0.0f;
sign = -1.0;
for (h = 1; h <= harmonics; h++) {
hf = (double)h;
partial = (sign * 4.0 / (hf * hf)) * cos(phase * hf);
samples[i] += (float)partial;
sign = -sign;
}
samples[i] *= scale;
}
}
}