very nuclear, rendering engine gone
This commit is contained in:
Gordon JC Pearce
parent
7f43f8987b
commit
28b6dcb1ef
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@ -22,7 +22,6 @@
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#include "nekobee-src/nekobee_synth.c"
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#include "nekobee-src/nekobee_voice.c"
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#include "nekobee-src/nekobee_voice_render.c"
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#include "nekobee-src/minblep_tables.c"
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// -----------------------------------------------------------------------
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// nekobee_handle_raw_event
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File diff suppressed because it is too large
Load Diff
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@ -1,414 +1,20 @@
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/* nekobee DSSI software synthesizer plugin
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*/
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#define _BSD_SOURCE 1
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#define _SVID_SOURCE 1
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#define _ISOC99_SOURCE 1
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#include <math.h>
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#include "nekobee.h"
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#include "nekobee_synth.h"
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#include "nekobee_voice.h"
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#ifndef M_PI
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#define M_PI 3.14159265358979323846
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#endif
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#define M_2PI_F (2.0f * (float)M_PI)
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#define M_PI_F (float)M_PI
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#define VCF_FREQ_MAX (0.825f) /* original filters only stable to this frequency */
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static int tables_initialized = 0;
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float nekobee_pitch[128];
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#define pitch_ref_note 69
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#define volume_to_amplitude_scale 128
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static float volume_to_amplitude_table[4 + volume_to_amplitude_scale + 2];
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static float velocity_to_attenuation[128];
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static float qdB_to_amplitude_table[4 + 256 + 0];
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void
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nekobee_init_tables(void)
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{
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int i;
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float pexp;
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float volume, volume_exponent;
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float ol, amp;
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if (tables_initialized)
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return;
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/* MIDI note to pitch */
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for (i = 0; i < 128; ++i) {
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pexp = (float)(i - pitch_ref_note) / 12.0f;
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nekobee_pitch[i] = powf(2.0f, pexp);
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}
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/* volume to amplitude
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*
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* This generates a curve which is:
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* volume_to_amplitude_table[128 + 4] = 0.25 * 3.16... ~= -2dB
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* volume_to_amplitude_table[64 + 4] = 0.25 * 1.0 ~= -12dB
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* volume_to_amplitude_table[32 + 4] = 0.25 * 0.316... ~= -22dB
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* volume_to_amplitude_table[16 + 4] = 0.25 * 0.1 ~= -32dB
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* etc.
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*/
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volume_exponent = 1.0f / (2.0f * log10f(2.0f));
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for (i = 0; i <= volume_to_amplitude_scale; i++) {
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volume = (float)i / (float)volume_to_amplitude_scale;
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volume_to_amplitude_table[i + 4] = powf(2.0f * volume, volume_exponent) / 4.0f;
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}
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volume_to_amplitude_table[ -1 + 4] = 0.0f;
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volume_to_amplitude_table[129 + 4] = volume_to_amplitude_table[128 + 4];
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/* velocity to attenuation
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*
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* Creates the velocity to attenuation lookup table, for converting
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* velocities [1, 127] to full-velocity-sensitivity attenuation in
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* quarter decibels. Modeled after my TX-7's velocity response.*/
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velocity_to_attenuation[0] = 253.9999f;
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for (i = 1; i < 127; i++) {
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if (i >= 10) {
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ol = (powf(((float)i / 127.0f), 0.32f) - 1.0f) * 100.0f;
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amp = powf(2.0f, ol / 8.0f);
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} else {
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ol = (powf(((float)10 / 127.0f), 0.32f) - 1.0f) * 100.0f;
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amp = powf(2.0f, ol / 8.0f) * (float)i / 10.0f;
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}
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velocity_to_attenuation[i] = log10f(amp) * -80.0f;
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}
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velocity_to_attenuation[127] = 0.0f;
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/* quarter-decibel attenuation to amplitude */
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qdB_to_amplitude_table[-1 + 4] = 1.0f;
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for (i = 0; i <= 255; i++) {
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qdB_to_amplitude_table[i + 4] = powf(10.0f, (float)i / -80.0f);
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}
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tables_initialized = 1;
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}
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static inline float
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volume(float level)
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{
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unsigned char segment;
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float fract;
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level *= (float)volume_to_amplitude_scale;
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segment = lrintf(level - 0.5f);
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fract = level - (float)segment;
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return volume_to_amplitude_table[segment + 4] + fract *
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(volume_to_amplitude_table[segment + 5] -
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volume_to_amplitude_table[segment + 4]);
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}
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static inline float
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qdB_to_amplitude(float qdB)
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{
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int i = lrintf(qdB - 0.5f);
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float f = qdB - (float)i;
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return qdB_to_amplitude_table[i + 4] + f *
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(qdB_to_amplitude_table[i + 5] -
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qdB_to_amplitude_table[i + 4]);
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}
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void blosc_place_step_dd(float *buffer, int index, float phase, float w, float scale){
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float r;
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int i;
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r = MINBLEP_PHASES * phase / w;
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i = lrintf(r - 0.5f);
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r -= (float)i;
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i &= MINBLEP_PHASE_MASK; /* port changes can cause i to be out-of-range */
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/* This would be better than the above, but more expensive:
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* while (i < 0) {
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* i += MINBLEP_PHASES;
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* index++;
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* }
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*/
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while (i < MINBLEP_PHASES * STEP_DD_PULSE_LENGTH) {
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buffer[index] += scale * (step_dd_table[i].value + r * step_dd_table[i].delta);
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i += MINBLEP_PHASES;
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index++;
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}
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}
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void vco(unsigned long sample_count, nekobee_voice_t *voice, struct blosc *osc,
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int index, float w)
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{
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unsigned long sample;
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float pos = osc->pos;
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float pw, gain, halfgain, out;
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pw=0.46f;
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gain=1.0f;
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halfgain=gain*0.5f;
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int bp_high = osc->bp_high;
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out=(bp_high ? halfgain : -halfgain);
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switch (osc->waveform)
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{
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default:
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case 0: {
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for (sample = 0; sample < sample_count; sample++) {
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pos += w;
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if (bp_high) {
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if (pos >= pw) {
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blosc_place_step_dd(voice->osc_audio, index, pos - pw, w, -gain);
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bp_high = 0;
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out = -halfgain;
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}
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if (pos >= 1.0f) {
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pos -= 1.0f;
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blosc_place_step_dd(voice->osc_audio, index, pos, w, gain);
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bp_high = 1;
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out = halfgain;
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}
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} else {
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if (pos >= 1.0f) {
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pos -= 1.0f;
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blosc_place_step_dd(voice->osc_audio, index, pos, w, gain);
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bp_high = 1;
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out = halfgain;
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}
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if (bp_high && pos >= pw) {
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blosc_place_step_dd(voice->osc_audio, index, pos - pw, w, -gain);
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bp_high = 0;
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out = -halfgain;
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}
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}
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voice->osc_audio[index + DD_SAMPLE_DELAY] += out;
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index++;
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}
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osc->pos = pos;
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osc->bp_high = bp_high;
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break;
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}
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case 1: // sawtooth wave
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{
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for (sample=0; sample < sample_count; sample++) {
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pos += w;
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if (pos >= 1.0f) {
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pos -= 1.0f;
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blosc_place_step_dd(voice->osc_audio, index, pos, w, gain);
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}
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voice->osc_audio[index + DD_SAMPLE_DELAY] += gain * (0.5f - pos);
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index++;
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}
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break;
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}
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}
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osc->pos=pos;
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}
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static inline void
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vcf_4pole(nekobee_voice_t *voice, unsigned long sample_count,
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float *in, float *out, float *cutoff, float qres, float *amp)
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{
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unsigned long sample;
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float freqcut, freqcut2, highpass,
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delay1 = voice->delay1,
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delay2 = voice->delay2,
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delay3 = voice->delay3,
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delay4 = voice->delay4;
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qres = 2.0f - qres * 1.995f;
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for (sample = 0; sample < sample_count; sample++) {
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/* Hal Chamberlin's state variable filter */
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freqcut = cutoff[sample] * 2.0f;
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freqcut2 = cutoff[sample] * 4.0f;
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if (freqcut > VCF_FREQ_MAX) freqcut = VCF_FREQ_MAX;
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if (freqcut2 > VCF_FREQ_MAX) freqcut2 = VCF_FREQ_MAX;
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delay2 = delay2 + freqcut * delay1; /* delay2/4 = lowpass output */
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highpass = in[sample] - delay2 - qres * delay1;
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delay1 = freqcut * highpass + delay1; /* delay1/3 = bandpass output */
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delay4 = delay4 + freqcut2 * delay3;
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highpass = delay2 - delay4 - qres * delay3;
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delay3 = freqcut2 * highpass + delay3;
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/* mix filter output into output buffer */
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out[sample] += 0.1*atan(3*delay4 * amp[sample]);
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}
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voice->delay1 = delay1;
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voice->delay2 = delay2;
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voice->delay3 = delay3;
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voice->delay4 = delay4;
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voice->c5 = 0.0f;
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}
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/*
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* nekobee_voice_render
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*
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* generate the actual sound data for this voice
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* Copyright (C) 2023 Gordonjcp, with attributions inline
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License as
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* published by the Free Software Foundation; either version 2 of
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* the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be
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* useful, but WITHOUT ANY WARRANTY; without even the implied
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* warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public
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* License along with this program; if not, write to the Free
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* Software Foundation, Inc., 59 Temple Place, Suite 330, Boston,
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* MA 02111-1307, USA.
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*/
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void
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nekobee_voice_render(nekobee_synth_t *synth, nekobee_voice_t *voice,
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float *out, unsigned long sample_count,
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int do_control_update)
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{
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unsigned long sample;
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/* state variables saved in voice */
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float lfo_pos = voice->lfo_pos,
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vca_eg = voice->vca_eg,
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vcf_eg = voice->vcf_eg;
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unsigned char vca_eg_phase = voice->vca_eg_phase,
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vcf_eg_phase = voice->vcf_eg_phase;
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int osc_index = voice->osc_index;
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/* temporary variables used in calculating voice */
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float fund_pitch;
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float deltat = synth->deltat;
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float freq, cutoff, vcf_amt;
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float vcf_acc_amt;
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/* set up synthesis variables from patch */
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float omega;
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float vca_eg_amp = qdB_to_amplitude(velocity_to_attenuation[voice->velocity] * 0);
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float vca_eg_rate_level[3], vca_eg_one_rate[3];
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float vcf_eg_amp = qdB_to_amplitude(velocity_to_attenuation[voice->velocity] * 0);
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float vcf_eg_rate_level[3], vcf_eg_one_rate[3];
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float qres = synth->resonance;
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float vol_out = volume(synth->volume);
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float velocity = (voice->velocity);
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float vcf_egdecay = synth->decay;
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fund_pitch = 0.1f*voice->target_pitch +0.9 * voice->prev_pitch; /* glide */
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if (do_control_update) {
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voice->prev_pitch = fund_pitch; /* save pitch for next time */
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}
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fund_pitch *= 440.0f;
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omega = synth->tuning * fund_pitch;
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// if we have triggered ACCENT
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// we need a shorter decay
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// we should probably have something like this in the note on code
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// that could trigger an ACCENT light
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if (velocity>90) {
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vcf_egdecay=.0005;
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}
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// VCA - In a real 303, it is set for around 2 seconds
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vca_eg_rate_level[0] = 0.1f * vca_eg_amp; // instant on attack
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vca_eg_one_rate[0] = 0.9f; // very fast
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vca_eg_rate_level[1] = 0.0f; // sustain is zero
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vca_eg_one_rate[1] = 1.0f - 0.00001f; // decay time is very slow
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vca_eg_rate_level[2] = 0.0f; // decays to zero
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vca_eg_one_rate[2] = 0.975f; // very fast release
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// VCF - funny things go on with the accent
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vcf_eg_rate_level[0] = 0.1f * vcf_eg_amp;
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vcf_eg_one_rate[0] = 1-0.1f; //0.9f;
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vcf_eg_rate_level[1] = 0.0f; // vcf_egdecay * *(synth->vcf_eg_sustain_level) * vcf_eg_amp;
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vcf_eg_one_rate[1] = 1.0f - vcf_egdecay;
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vcf_eg_rate_level[2] = 0.0f;
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vcf_eg_one_rate[2] = 0.9995f; // 1.0f - *(synth->vcf_eg_release_time);
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vca_eg_amp *= 0.99f;
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vcf_eg_amp *= 0.99f;
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freq = M_PI_F * deltat * fund_pitch * synth->mod_wheel; /* now (0 to 1) * pi */
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cutoff = 0.008f * synth->cutoff;
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// 303 always has slight VCF mod
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vcf_amt = 0.05f+(synth->envmod*0.75);
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/* copy some things so oscillator functions can see them */
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voice->osc1.waveform = lrintf(synth->waveform);
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// work out how much the accent will affect the filter
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vcf_acc_amt=.333f+ (synth->resonance/1.5f);
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for (sample = 0; sample < sample_count; sample++) {
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vca_eg = vca_eg_rate_level[vca_eg_phase] + vca_eg_one_rate[vca_eg_phase] * vca_eg;
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vcf_eg = vcf_eg_rate_level[vcf_eg_phase] + vcf_eg_one_rate[vcf_eg_phase] * vcf_eg;
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voice->freqcut_buf[sample] = (cutoff + (vcf_amt * vcf_eg/2.0f) + (synth->vcf_accent * synth->accent*0.5f));
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voice->vca_buf[sample] = vca_eg * vol_out*(1.0f + synth->accent*synth->vca_accent);
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if (!vca_eg_phase && vca_eg > vca_eg_amp) vca_eg_phase = 1; /* flip from attack to decay */
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if (!vcf_eg_phase && vcf_eg > vcf_eg_amp) vcf_eg_phase = 1; /* flip from attack to decay */
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}
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// oscillator
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vco(sample_count, voice, &voice->osc1, osc_index, deltat * omega);
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// VCF and VCA
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vcf_4pole(voice, sample_count, voice->osc_audio + osc_index, out, voice->freqcut_buf, qres, voice->vca_buf);
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osc_index += sample_count;
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if (do_control_update || osc_index > MINBLEP_BUFFER_LENGTH - (XSYNTH_NUGGET_SIZE + LONGEST_DD_PULSE_LENGTH)) {
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/* do those things should be done only once per control-calculation
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* interval ("nugget"), such as voice check-for-dead, pitch envelope
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* calculations, volume envelope phase transition checks, etc. */
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/* check if we've decayed to nothing, turn off voice if so */
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if (vca_eg_phase == 2 && voice->vca_buf[sample_count - 1] < 6.26e-6f) {
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// sound has completed its release phase (>96dB below volume '5' max)
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XDB_MESSAGE(XDB_NOTE, " nekobee_voice_render check for dead: killing note id %d\n", voice->note_id);
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nekobee_voice_off(voice);
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return; // we're dead now, so return
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}
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/* already saved prev_pitch above */
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/* check oscillator audio buffer index, shift buffer if necessary */
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if (osc_index > MINBLEP_BUFFER_LENGTH - (XSYNTH_NUGGET_SIZE + LONGEST_DD_PULSE_LENGTH)) {
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memcpy(voice->osc_audio, voice->osc_audio + osc_index,
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LONGEST_DD_PULSE_LENGTH * sizeof (float));
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memset(voice->osc_audio + LONGEST_DD_PULSE_LENGTH, 0,
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(MINBLEP_BUFFER_LENGTH - LONGEST_DD_PULSE_LENGTH) * sizeof (float));
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osc_index = 0;
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}
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}
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/* save things for next time around */
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voice->lfo_pos = lfo_pos;
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voice->vca_eg = vca_eg;
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voice->vca_eg_phase = vca_eg_phase;
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voice->vcf_eg = vcf_eg;
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voice->vcf_eg_phase = vcf_eg_phase;
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voice->osc_index = osc_index;
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return;
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(void)freq;
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(void)vcf_acc_amt;
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}
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||||
|
|
|
|||
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Reference in New Issue