#include "synth_engine.h"
#include "synth_math.h"
#include "lowpass.h"
#include "filter.h"
#include "control.h"
#include "sound.h"
#include <string.h>
float
sin_sample(float amp, float freq, unsigned long long phase, unsigned int sample_rate)
{
return amp * sinf(2.0 * M_PI * freq * ((float)phase / (float)sample_rate));
}
float
saw_sample(float amp, float freq, unsigned long long phase, unsigned int sample_rate)
{
return amp * (0.17 * (1.0 - (2.0 * M_PI * freq * fmod((float)phase, (float)(sample_rate / (freq)))) / (float)sample_rate));
}
float
sawX_sample(float amp, float freq, float sm, unsigned long long phase, unsigned int sample_rate)
{
float dOutput = 0.0;
for (float n = 1.0; n < sm; n++)
dOutput += (sinf(n * 2.0 * M_PI * freq * ((float)phase / (float)sample_rate))) / n;
return 0.5 * amp * dOutput;
}
float
sqr_sample(float amp, float freq, float duty_cycle, unsigned long long phase, unsigned int sample_rate)
{
if (duty_cycle < 0.0001 || duty_cycle > 0.9999) {
duty_cycle = 0.5;
}
return (fmod((float)phase / (float)sample_rate, 1.0 / freq) < duty_cycle * (1.0 / freq)) ? amp : -amp;
}
float
gen10(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
return fmodf(phase, sample_rate/f) / (sample_rate/f);
}
float
gen1110(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
float amplitude = 0.5; // Set the initial amplitude to 0.5
float period = 1.0 / f; // Calculate the period of the waveform
float angular_frequency = 2.0 * M_PI * f; // Calculate the angular frequency
// Calculate the phase angle
float phase_angle = fmodf(angular_frequency * phase / sample_rate, 2.0 * M_PI);
// Generate the sample value using the sine function
float sample = amplitude * sinf(phase_angle);
return sample;
}
float
gen110(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
float a = fmodf(phase, sample_rate/f);
return a > (sample_rate/f) / 2.0f ? 0.9999f : -0.9999f;
}
float
gen0(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
return sin_sample(1, f, phase, sample_rate);
//return chirp(phase, f);
/* return sqr_sample(0.1, f, 0.3, phase, sample_rate) */
/* + sqr_sample(0.1, f * 3.0 / 2.0 , 0.5, phase, sample_rate) */
/* + saw_sample(0.3, f, phase, sample_rate) */
/* + sin_sample(0.1, f, phase, sample_rate) */
/* + sin_sample(0.1, f * 5, phase, sample_rate) */
/* /\* + sin_sample(0.1, freq * 50 * 1021, phase, sample_rate) *\/ */
/* /\* + sin_sample(0.1, freq * 50 * 3531021, phase, sample_rate) *\/ */
/* + sin_sample(0.1, f * 7, phase, sample_rate); */
}
float
gen1(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
//return saw_sample(1, f, phase, sample_rate);
return sawX_sample(0.5, f, 5, phase, sample_rate)
+ saw_sample(0.3, 2 * f / 5, phase, sample_rate)
+ sin_sample(0.2, f * 5.0 / 7.0 , phase, sample_rate);
}
float
gen2(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
/* return sin_sample(0.5, f * sqrt(2) , phase, sample_rate) */
/* + sin_sample(0.5, f, phase, sample_rate); */
return sawX_sample(1, f, 15, phase, sample_rate);
}
float
gen3(float f, unsigned long long phase, float x, unsigned int sample_rate)
{
return sqr_sample(1, f, .5, phase, sample_rate);
/* return sawX_sample(0.7, f, 5, phase, sample_rate) */
/* + sin_sample(0.3, 4.0/17.0*f, phase, sample_rate); */
/* return saw_sample(0.5, f * (1 + sqrt(5)) / 2, phase, sample_rate) */
/* + sin_sample(0.3, f * x, phase, sample_rate) */
/* + sqr_sample(0.2, f * x, 0.2 * x * x, phase, sample_rate); */
}
float
lfo(float f, unsigned long long phase, unsigned int sample_rate) {
return sin_sample(1, f, phase, sample_rate);
}
int
notes_active(synth_t *synth)
{
int flag = 0;
for (int i = 0; i < MIDI_NOTES; i++) {
if (!synth->midi_note[i].active)
continue;
if (!adsr_amplitude(&synth->adsr,
(float)synth->midi_note[i].noteOn,
(float)synth->midi_note[i].noteOff,
(float)synth->midi_note[i].elapsed)
&& synth->midi_note[i].noteOff != 0) {
synth->midi_note[i].freq = 0;
synth->midi_note[i].channel = -1;
synth->midi_note[i].noteOn = -1;
synth->midi_note[i].noteOff = -1;
synth->midi_note[i].velocity = -1;
synth->midi_note[i].elapsed = -1;
synth->midi_note[i].active = 0;
} else {
flag++;
}
}
return flag;
}
float cur_freq, next_freq;
float
make_sample(void *synthData, unsigned int sample_rate, int frame)
{
synth_t *synth = (synth_t*)synthData;
float sample = 0.0f;
float n = notes_active(synth);
if (n == 0) return sample;
float rms = 0;
for (int i = 0; i < MIDI_NOTES; i++) {
if (!synth->midi_note[i].active)
continue;
rms += synth->midi_note[i].velocity * synth->midi_note[i].velocity;
}
rms = sqrt(rms / n);
for (int i = 0; i < MIDI_NOTES; i++) {
if (!synth->midi_note[i].active)
continue;
float adsr = adsr_amplitude(&synth->adsr,
synth->midi_note[i].noteOn,
synth->midi_note[i].noteOff,
synth->midi_note[i].elapsed);
float targ_freq = synth->midi_note[i].freq * cc_iget(&synth->cc_pitch, frame, FRAMES_PER_BUFFER);
float targ_freq_with_repitch = synth->midi_note[i].freq * cc_iget(&synth->cc_pitch, frame, FRAMES_PER_BUFFER);
float synth_sample = synth->gen[synth->geni](targ_freq,
synth->midi_note[i].elapsed,
synth->x,
sample_rate);
/* if (targ_freq != targ_freq_with_repitch) { */
/* unsigned long long tmp = synth->midi_note[i].elapsed; */
/* float next_sample = synth->gen[synth->geni](targ_freq_with_repitch, */
/* tmp, */
/* synth->x, */
/* sample_rate); */
/* while (synth_sample - next_sample > 0.001 && next_sample - synth_sample > 0.001) { */
/* next_sample = synth->gen[synth->geni](targ_freq_with_repitch, */
/* ++tmp, */
/* synth->x, */
/* sample_rate); */
/* } */
/* synth_sample = next_sample; */
/* } */
sample += 0.2 * rms * adsr * synth_sample;
}
/* filter */
if (synth->filter) {
// ALLL THE FILTERS
float cutoff = cc_iget(&synth->cc_cutoff, frame, FRAMES_PER_BUFFER);
float reso = round(cc_iget(&synth->cc_resonance, frame, FRAMES_PER_BUFFER));
//cutoff = cutoff + cutoff * (synth->lfo.amp) * lfo(cc_iget(&synth->cc_lfo_freq, frame, FRAMES_PER_BUFFER), synth->lfo.elapsed, sample_rate);
if (cutoff == 0) cutoff = 0.001;
LowPass_Update(reso, cutoff, sample_rate);
sample = LowPass_Filter(sample);
update_bw_low_pass_filter(synth->fff, SAMPLE_RATE,cutoff, reso);
sample = bw_low_pass(synth->fff, sample);
}
sample = synth->gain * sample;
// band stop for high freqs
sample = bw_band_stop(synth->fff2, sample);
if (synth->clamp) sample = clamp(sample, -1, 1);
return sample;
}
void
add_to_viz(synth_t *synth, float sample)
{
synth->viz.wave[synth->viz.wi++] = sample;
if (synth->viz.wi >= VIZ_BUF) {
synth->viz.wi = 0;
}
}
void
increment_synth(synth_t *synth)
{
synth->lfo.elapsed++;
synth->adsr.elapsed++;
for (int i = 0; i < MIDI_NOTES; i++) {
if (!synth->midi_note[i].active)
continue;
synth->midi_note[i].elapsed++;
}
}
void
get_frame(void *outputBuffer, synth_t *synth, int i)
{
float *out = (float*)outputBuffer + 2 * i;
float s = 0.0f;
if (!synth->active) {
*out++ = 0.0f;
*out++ = 0.0f;
add_to_viz(synth, 0.0f);
synth->lfo.elapsed = 0;
return;
}
if (!notes_active(synth)) {
synth->active = 0;
*out++ = 0.0f;
*out++ = 0.0f;
add_to_viz(synth, 0.0f);
return;
}
s = make_sample(synth, SAMPLE_RATE, i);
*out++ = s;
*out++ = s;
// move time
increment_synth(synth);
// viz
add_to_viz(synth, s);
}
#include "fftw3.h"
int
process(float * buffer)
{
int len = FRAMES_PER_BUFFER;
float buf[FRAMES_PER_BUFFER];
for( unsigned long i=0; i < len * 2; i += 2 ) {
buf[i / 2] = buffer[i];
}
fftwf_complex* frequency_signal = (fftwf_complex*)fftwf_malloc(sizeof(fftwf_complex) * (len / 2 + 1));
//double* time_signal = (double*)fftwf_malloc(sizeof(double) * len);
fftwf_plan forward_plan = fftwf_plan_dft_r2c_1d(len, buf, frequency_signal, FFTW_ESTIMATE);
fftwf_plan backward_plan = fftwf_plan_dft_c2r_1d(len, frequency_signal, buf, FFTW_ESTIMATE);
fftwf_execute(forward_plan);
/* for (int i=0; i < (len / 2 + 1); i++ ) { */
/* if (frequency_signal[i][0] > 1) { */
/* frequency_signal[i][0] = 0; */
/* } */
/* if (frequency_signal[i][1] > 1) { */
/* frequency_signal[i][1] = 0; */
/* } */
/* printf("%f %f | ", sqrt(frequency_signal[i][0] * frequency_signal[i][0] + frequency_signal[i][1] * frequency_signal[i][1]), atan2(frequency_signal[i][1], frequency_signal[i][0])); */
/* float f = atan2(frequency_signal[i][1], frequency_signal[i][0]); */
/* if (f > M_PI / 2 && f < 2 * M_PI / 3) { */
/* frequency_signal[i][0] = 0.0; */
/* frequency_signal[i][1] = 0.0; */
/* } */
/* } */
fftwf_execute(backward_plan);
for (unsigned long i=0; i < len * 2; i += 2) {
buffer[i] = buf[i / 2] / len ;
buffer[i + 1] = buf[i / 2] / len ;
}
fftwf_destroy_plan(forward_plan);
fftwf_destroy_plan(backward_plan);
fftwf_free(frequency_signal);
//fftwf_free(time_signal);
/* for( unsigned long i=0; i < len * 2; i += 2 ) { */
/* buf[i / 2] = buffer[i]; */
/* } */
/* float kernel[7] = {.01, .88, .21, .36, .21, .13, .01}; */
/* float res[len + 7 - 1]; */
/* convole(buf, kernel, len, 7, res); */
/* int N = (len + 7 - 1); */
/* int M = len; */
/* float ratio = (float) N / M; */
/* for (int i = 0; i < M; i++) { */
/* int index = (int)(i * ratio); */
/* buf[i] = res[index]; */
/* } */
/* for (unsigned long i=0; i < len * 2; i += 2) { */
/* buffer[i] = buf[i / 2] ; */
/* buffer[i + 1] = buf[i / 2] ; */
/* } */
return 0;
}
int
sound_gen(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *synthData)
{
synth_t *synth = (synth_t*)synthData;
float *out = (float*)outputBuffer;
float buffer[2 * FRAMES_PER_BUFFER];
(void) timeInfo;
(void) statusFlags;
(void) inputBuffer;
// get_changes();
for (int i = 0; i < synth->cci; i++) {
cc_prep(synth->ccs[i]);
}
// fill buffer
for( unsigned long i=0; i<framesPerBuffer; i++ ) {
// use iget inside
get_frame(buffer, synth, i);
}
// process buffer
if (synth->multi) process(buffer);
// output buffer
for( unsigned long i=0; i<framesPerBuffer * 2; i += 2 ) {
// use iget inside
*out++ = buffer[i];
*out++ = buffer[i+1];
}
// finalize_changes();
for (int i = 0; i < synth->cci; i++) {
cc_fix(synth->ccs[i]);
}
return paContinue;
}
void
init_synth(synth_t * synth)
{
synth->cci = 0;
CC(synth->cc_cutoff, "cutoff", 10, 5000, 30, 5000);
CC(synth->cc_resonance, "resonance", 1, 10, .02, 1);
CC(synth->cc_lfo_freq, "lfo_freq", .01, 10, .02, .1);
CC(synth->cc_pitch, "pitch", -3, 4, 0.01, 1);
synth->modi = 0;
synth->freq_offset = 0;
synth->gain = 1;
synth->x = 1;
synth->adsr.a = 0.0;
synth->adsr.peak = 1.0f;
synth->adsr.d = 0.3;
synth->adsr.s = 0.7;
synth->adsr.r = 0.4;
synth->adsr.elapsed = 0;
synth->lfo.freq = 1.0f;
synth->lfo.amp = 0.0f;
synth->lfo.elapsed = 0;
for (int i =0; i<MIDI_NOTES; i++) {
synth->midi_note[i].freq = 0;
synth->midi_note[i].channel = -1;
synth->midi_note[i].noteOn = -1;
synth->midi_note[i].noteOff = -1;
synth->midi_note[i].velocity = -1;
synth->midi_note[i].elapsed = -1;
synth->midi_note[i].active = 0;
synth->midi_note[i].adsr = &(synth->adsr);
}
synth->octave = 3;
synth->poly = 0;
synth->multi = 0;
synth->filter = 1;
synth->cutoff = 22000.0f;
synth->clamp = 1;
synth->gen[0] = gen0;
synth->gen[1] = gen1;
synth->gen[2] = gen2;
synth->gen[3] = gen3;
synth->geni = 2;
synth->active = 0;
synth->viz.sample_rate_divider = 1;
synth->viz.wi = 0;
LowPass_Init();
synth->fff = create_bw_low_pass_filter(2, SAMPLE_RATE, 400);
synth->fff2 = create_bw_band_stop_filter(8, SAMPLE_RATE, 15000, 22000);
init_sound(synth, sound_gen);
}
void
free_synth(synth_t * synth)
{
destroy_sound(synth);
free_bw_low_pass(synth->fff);
free_bw_band_stop(synth->fff2);
}