add lfo, pitch bend, key stack (for unison trills), filter lfo, filter EG intensity, 24db/oct filter
vampirefrog
6 years ago
parent
7c84b45a03
commit
693332aa07
@ -1,10 +1,11 @@
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CFLAGS=$(shell pkg-config alsa jack --cflags)
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LDFLAGS=$(shell pkg-config alsa jack --libs)
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PKGS=alsa jack gtk+-3.0
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CFLAGS=$(shell pkg-config $(PKGS) --cflags)
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LDFLAGS=$(shell pkg-config $(PKGS) --libs)
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ifeq ($(DEBUG),true)
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CFLAGS+=-Wall -g
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endif
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programs=vampisynth
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vampisynth_SRCS=synth.c envelope.c filter.c oscillator.c voice.c main.c
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vampisynth_SRCS=synth.c envelope.c filter.c oscillator.c voice.c sineosc.c main.c
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include common.mk/common.mk
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@ -1,30 +1,72 @@
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#include <math.h>
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#include <string.h>
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#include <stdlib.h>
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#include "filter.h"
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void filter_init(struct Filter *filter, float cutoff, float resonance) {
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filter->v0 = filter->v1 = 0;
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memset(filter, 0, sizeof(struct Filter));
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filter->cutoff = cutoff;
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filter->resonance = resonance;
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}
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void filter_set_cutoff(struct Filter *filter, float cutoff) {
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filter->cutoff = cutoff;
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if(filter->cutoff > 0.9) filter->cutoff = 0.9;
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else if(filter->cutoff < 0.0) filter->cutoff = 0.0;
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}
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void filter_set_resonance(struct Filter *filter, float resonance) {
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filter->resonance = resonance;
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}
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// http://www.musicdsp.org/showone.php?id=25
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// float filter_sample(struct Filter *filter, float input) {
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// float q = 1.0f - filter->cutoff;
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// float p = filter->cutoff + 0.8f * filter->cutoff * q;
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// float f = p + p - 1.0f;
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// q = filter->resonance * (1.0f + 0.5f * q * (1.0f - q + 5.6f * q * q));
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// #define OVERSAMPLE 2
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// float output = 0;
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// for(int i = 0; i < OVERSAMPLE; i++) {
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// input -= q * filter->b4; //feedback
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// float t1 = filter->b1; filter->b1 = (input + filter->b0) * p - filter->b1 * f;
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// float t2 = filter->b2; filter->b2 = (filter->b1 + t1) * p - filter->b2 * f;
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// t1 = filter->b3; filter->b3 = (filter->b2 + t2) * p - filter->b3 * f;
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// filter->b4 = (filter->b3 + t1) * p - filter->b4 * f;
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// filter->b4 = filter->b4 - filter->b4 * filter->b4 * filter->b4 * 0.166667f; //clipping
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// filter->b0 = input;
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// output += filter->b4;
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// }
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// return output / (float)OVERSAMPLE;
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// }
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// mad's filter code
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float filter_sample(struct Filter *filter, float input) {
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float c = pow(0.5, (128-filter->cutoff) / 16.0);
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float r = pow(0.5, (filter->resonance+24) / 16.0);
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if(++filter->last_calc >= 16) {
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filter->last_calc = 0;
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filter->cutoff_calc = pow (0.5, 8.5 - (filter->cutoff)*8 );
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filter->resonance_calc = filter->resonance;
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}
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float cutoff = filter->cutoff_calc;
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float resonance = filter->resonance_calc;
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float outputs[2]; // oversample
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for(int i = 0; i < 2; i++) {
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filter->v0 = (1-r*c)*filter->v0 - (c)*filter->v1 + (c)*input;
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outputs[i] = filter->v1 = (1-r*c)*filter->v1 + (c)*filter->v0;
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#define OVERSAMPLE 2
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float output = 0;
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for(int i = 0; i < OVERSAMPLE; i++) {
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float in = filter->resonance_input;
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if(in > 1) in = 1;
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if(in < -1) in = -1;
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in = 1.5*in - 0.5*in*in*in;
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in = input - in;
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filter->stage_1 += (in - filter->stage_1) * cutoff;
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filter->stage_2 += (filter->stage_1 - filter->stage_2) * cutoff;
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filter->stage_3 += (filter->stage_2 - filter->stage_3) * cutoff;
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filter->stage_4 += (filter->stage_3 - filter->stage_4) * cutoff;
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output += filter->stage_4;
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filter->resonance_input = output * resonance;
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}
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return (outputs[0] + outputs[1]) / 2;
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return output / OVERSAMPLE;
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}
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@ -0,0 +1,19 @@
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#include <math.h>
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#include "sineosc.h"
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#include "config.h"
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void sine_osc_init(struct SineOsc *osc) {
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osc->s0 = 1.0;
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osc->s1 = 0.0;
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}
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void sine_osc_set_freq(struct SineOsc *osc, float freq) {
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osc->a = 2.0 * sinf(M_PI * freq / SAMPLE_RATE);
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}
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float sine_osc_sample(struct SineOsc *osc) {
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osc->s0 = osc->s0 - osc->a * osc->s1;
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osc->s1 = osc->s1 + osc->a * osc->s0;
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return osc->s0;
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}
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@ -0,0 +1,12 @@
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#ifndef SINEOSC_H_
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#define SINEOSC_H_
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struct SineOsc {
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float a, s0, s1;
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};
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void sine_osc_init(struct SineOsc *osc);
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void sine_osc_set_freq(struct SineOsc *osc, float freq);
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float sine_osc_sample(struct SineOsc *osc);
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#endif /* SINEOSC_H_ */
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