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livetrax/libs/plugins/reasonablesynth.lv2/rsynth.c

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/*
* Copyright (C) 2013-2019 Robin Gareus <robin@gareus.org>
*
* This program 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 2 of the License, or
* (at your option) any later version.
*
* This program 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 program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef _GNU_SOURCE
#define _GNU_SOURCE // needed for M_PI
#endif
#include <math.h>
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#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#ifndef COMPILER_MSVC
#include <stdbool.h>
#endif
#include <assert.h>
#ifndef BUFFER_SIZE_SAMPLES
#define BUFFER_SIZE_SAMPLES 64
#endif
#ifndef MIN
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#define MIN(A, B) ((A) < (B) ? (A) : (B))
#endif
/* internal MIDI event abstraction */
enum RMIDI_EV_TYPE {
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INVALID = 0,
NOTE_ON,
NOTE_OFF,
PROGRAM_CHANGE,
CONTROL_CHANGE,
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PITCH_BEND,
};
struct rmidi_event_t {
enum RMIDI_EV_TYPE type;
uint8_t channel; /**< the MIDI channel number 0-15 */
union {
struct {
uint8_t note;
uint8_t velocity;
} tone;
struct {
uint8_t param;
uint8_t value;
} control;
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struct {
uint8_t lo;
uint8_t hi;
} bend;
} d;
};
typedef struct {
uint32_t tme[3]; // attack, decay, release times [settings:ms || internal:samples]
float vol[2]; // attack, sustain volume [0..1]
uint32_t off[3]; // internal use (added attack,decay,release times)
} ADSRcfg;
typedef struct _RSSynthChannel {
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uint32_t keycomp;
uint32_t adsr_cnt[128];
float adsr_amp[128];
float phase[128]; // various use, zero'ed on note-on
int8_t miditable[128]; // internal, note-on/off velocity
int8_t midimsgs[128]; // internal, note-off + on in same cycle, sustained-off
int8_t sustain; // sustain pedal pressed
ADSRcfg adsr;
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float pitch_bend;
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void (*synthesize) (struct _RSSynthChannel* sc,
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const uint8_t note, const float vol, float pc,
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const size_t n_samples, float* left, float* right);
} RSSynthChannel;
typedef void (*SynthFunction) (RSSynthChannel* sc,
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const uint8_t note,
const float vol,
const float pc,
const size_t n_samples,
float* left, float* right);
typedef struct {
uint32_t boffset;
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float buf[2][BUFFER_SIZE_SAMPLES];
RSSynthChannel sc[16];
float freqs[128];
float kcgain;
float kcfilt;
double rate;
uint32_t xmas_on;
uint32_t xmas_off;
} RSSynthesizer;
/* initialize ADSR values
*
* @param rate sample-rate
* @param a attack time in seconds
* @param d decay time in seconds
* @param r release time in seconds
* @param avol attack gain [0..1]
* @param svol sustain volume level [0..1]
*/
static void
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init_adsr (ADSRcfg* adsr, const double rate,
const uint32_t a, const uint32_t d, const uint32_t r,
const float avol, const float svol)
{
adsr->vol[0] = avol;
adsr->vol[1] = svol;
adsr->tme[0] = a * rate / 1000.0;
adsr->tme[1] = d * rate / 1000.0;
adsr->tme[2] = r * rate / 1000.0;
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assert (adsr->tme[0] > 32);
assert (adsr->tme[1] > 32);
assert (adsr->tme[2] > 32);
assert (adsr->vol[0] >= 0 && adsr->vol[1] <= 1.0);
assert (adsr->vol[1] >= 0 && adsr->vol[1] <= 1.0);
adsr->off[0] = adsr->tme[0];
adsr->off[1] = adsr->tme[1] + adsr->off[0];
adsr->off[2] = adsr->tme[2] + adsr->off[1];
}
/* calculate per-sample, per-key envelope */
static inline float
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adsr_env (RSSynthChannel* sc, const uint8_t note)
{
if (sc->adsr_cnt[note] < sc->adsr.off[0]) {
// attack
const uint32_t p = ++sc->adsr_cnt[note];
if (p == sc->adsr.tme[0]) {
sc->adsr_amp[note] = sc->adsr.vol[0];
return sc->adsr.vol[0];
} else {
const float d = sc->adsr.vol[0] - sc->adsr_amp[note];
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return sc->adsr_amp[note] + (p / (float)sc->adsr.tme[0]) * d;
}
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} else if (sc->adsr_cnt[note] < sc->adsr.off[1]) {
// decay
const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[0];
if (p == sc->adsr.tme[1]) {
sc->adsr_amp[note] = sc->adsr.vol[1];
return sc->adsr.vol[1];
} else {
const float d = sc->adsr.vol[1] - sc->adsr_amp[note];
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return sc->adsr_amp[note] + (p / (float)sc->adsr.tme[1]) * d;
}
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} else if (sc->adsr_cnt[note] == sc->adsr.off[1]) {
// sustain
return sc->adsr.vol[1];
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} else if (sc->adsr_cnt[note] < sc->adsr.off[2]) {
// release
const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[1];
if (p == sc->adsr.tme[2]) {
sc->adsr_amp[note] = 0;
return 0;
} else {
const float d = 0 - sc->adsr_amp[note];
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return sc->adsr_amp[note] + (p / (float)sc->adsr.tme[2]) * d;
}
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} else {
sc->adsr_cnt[note] = 0;
return 0;
}
}
/*****************************************************************************/
/* piano like sound w/slight stereo phase */
static void
synthesize_sineP (RSSynthChannel* sc,
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const uint8_t note, const float vol, float fq,
const size_t n_samples, float* left, float* right)
{
size_t i;
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float phase = sc->phase[note];
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if (sc->pitch_bend != 1) {
fq *= sc->pitch_bend;
}
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for (i = 0; i < n_samples; ++i) {
float env = adsr_env (sc, note);
if (sc->adsr_cnt[note] == 0) {
break;
}
const float amp = vol * env;
if (amp > 1e-10) {
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left[i] += amp * sinf (2.0 * M_PI * phase);
left[i] += .300 * amp * sinf (2.0 * M_PI * phase * 2.0);
left[i] += .150 * amp * sinf (2.0 * M_PI * phase * 3.0);
left[i] += .080 * amp * sinf (2.0 * M_PI * phase * 4.0);
//left[i] -= .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
//left[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
left[i] += .020 * amp * sinf (2.0 * M_PI * phase * 7.0);
phase += fq;
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right[i] += amp * sinf (2.0 * M_PI * phase);
right[i] += .300 * amp * sinf (2.0 * M_PI * phase * 2.0);
right[i] += .150 * amp * sinf (2.0 * M_PI * phase * 3.0);
right[i] -= .080 * amp * sinf (2.0 * M_PI * phase * 4.0);
//right[i] += .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
//right[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
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right[i] -= .020 * amp * sinf (2.0 * M_PI * phase * 7.0);
} else {
phase += fq;
}
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if (phase > 1.0)
phase -= 2.0;
}
sc->phase[note] = phase;
}
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static const ADSRcfg piano_adsr = { { 5, 800, 100 }, { 1.0, 0.0 }, { 0, 0, 0 } };
/*****************************************************************************/
/* process note - move through ADSR states, count active keys,.. */
static void
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process_key (void* synth,
const uint8_t chn, const uint8_t note,
const size_t n_samples,
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float* left, float* right)
{
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RSSynthesizer* rs = (RSSynthesizer*)synth;
RSSynthChannel* sc = &rs->sc[chn];
const int8_t vel = sc->miditable[note];
const int8_t msg = sc->midimsgs[note];
const float vol = /* master_volume */ 0.1f * abs (vel) / 127.f;
const float phase = sc->phase[note];
const int8_t sus = sc->sustain;
sc->midimsgs[note] &= ~3;
if (phase == -10 && vel > 0) {
// new note on
sc->midimsgs[note] &= ~4;
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assert (sc->adsr_cnt[note] == 0);
sc->adsr_amp[note] = 0;
sc->adsr_cnt[note] = 0;
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sc->phase[note] = 0;
sc->keycomp++;
//printf("[On] Now %d keys active on chn %d\n", sc->keycomp, chn);
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} else if (phase >= -1.0 && phase <= 1.0 && vel > 0) {
// sustain note or re-start note while adsr in progress:
if (sc->adsr_cnt[note] > sc->adsr.off[1] || msg == 3 || msg == 5 || msg == 7) {
sc->midimsgs[note] &= ~4;
// x-fade to attack
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sc->adsr_amp[note] = adsr_env (sc, note);
sc->adsr_cnt[note] = 0;
}
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} else if (phase >= -1.0 && phase <= 1.0 && vel < 0) {
sc->midimsgs[note] |= 4;
// note off
if (sc->adsr_cnt[note] <= sc->adsr.off[1] && !sus) {
if (sc->adsr_cnt[note] != sc->adsr.off[1]) {
// x-fade to release
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sc->adsr_amp[note] = adsr_env (sc, note);
}
sc->adsr_cnt[note] = sc->adsr.off[1] + 1;
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} else if (sus && sc->adsr_cnt[note] == sc->adsr.off[1]) {
sc->adsr_cnt[note] = sc->adsr.off[1] + 1;
}
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} else {
//printf("FORCE NOTE OFF: %d %d\n", vel, sus);
/* note-on + off in same cycle */
sc->miditable[note] = 0;
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sc->adsr_cnt[note] = 0;
sc->phase[note] = -10;
return;
}
//printf("NOTE: %d (%d %d %d)\n", sc->adsr_cnt[note], sc->adsr.off[0], sc->adsr.off[1], sc->adsr.off[2]);
// synthesize actual sound
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sc->synthesize (sc, note, vol, rs->freqs[note], n_samples, left, right);
if (sc->adsr_cnt[note] == 0) {
//printf("Note %d,%d released\n", chn, note);
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sc->midimsgs[note] = 0;
sc->miditable[note] = 0;
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sc->adsr_amp[note] = 0;
sc->phase[note] = -10;
sc->keycomp--;
//printf("[off] Now %d keys active on chn %d\n", sc->keycomp, chn);
}
}
/* synthesize a BUFFER_SIZE_SAMPLES's of audio-data */
static void
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synth_fragment (void* synth, const size_t n_samples, float* left, float* right)
{
RSSynthesizer* rs = (RSSynthesizer*)synth;
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memset (left, 0, n_samples * sizeof (float));
memset (right, 0, n_samples * sizeof (float));
uint8_t keycomp = 0;
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int c, k;
size_t i;
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for (c = 0; c < 16; ++c) {
for (k = 0; k < 128; ++k) {
if (rs->sc[c].miditable[k] == 0) {
continue;
}
process_key (synth, c, k, n_samples, left, right);
}
keycomp += rs->sc[c].keycomp;
}
#if 1 // key-compression
float kctgt = 8.0 / (float)(keycomp + 7.0);
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if (kctgt < .5) kctgt = .5;
if (kctgt > 1.0) kctgt = 1.0;
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const float _w = rs->kcfilt;
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for (i = 0; i < n_samples; ++i) {
rs->kcgain += _w * (kctgt - rs->kcgain);
left[i] *= rs->kcgain;
right[i] *= rs->kcgain;
}
rs->kcgain += 1e-12;
#endif
}
static void
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synth_reset_channel (RSSynthChannel* sc)
{
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int k;
for (k = 0; k < 128; ++k) {
sc->adsr_cnt[k] = 0;
sc->adsr_amp[k] = 0;
sc->phase[k] = -10;
sc->miditable[k] = 0;
sc->midimsgs[k] = 0;
}
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sc->keycomp = 0;
sc->pitch_bend = 1.0;
}
static void
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synth_reset (void* synth)
{
RSSynthesizer* rs = (RSSynthesizer*)synth;
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int c;
for (c = 0; c < 16; ++c) {
synth_reset_channel (&(rs->sc[c]));
}
rs->kcgain = 0;
}
static void
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synth_load (RSSynthChannel* sc, const double rate,
SynthFunction synthesize,
ADSRcfg const* const adsr)
{
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synth_reset_channel (sc);
init_adsr (&sc->adsr, rate,
adsr->tme[0], adsr->tme[1], adsr->tme[2],
adsr->vol[0], adsr->vol[1]);
sc->synthesize = synthesize;
}
/**
* internal abstraction of MIDI data handling
*/
static void
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synth_process_midi_event (void* synth, struct rmidi_event_t* ev)
{
RSSynthesizer* rs = (RSSynthesizer*)synth;
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switch (ev->type) {
case NOTE_ON:
rs->sc[ev->channel].midimsgs[ev->d.tone.note] |= 1;
if (rs->sc[ev->channel].miditable[ev->d.tone.note] <= 0)
rs->sc[ev->channel].miditable[ev->d.tone.note] = ev->d.tone.velocity;
break;
case NOTE_OFF:
rs->sc[ev->channel].midimsgs[ev->d.tone.note] |= 2;
if (rs->sc[ev->channel].miditable[ev->d.tone.note] > 0)
rs->sc[ev->channel].miditable[ev->d.tone.note] *= -1.0;
break;
case PROGRAM_CHANGE:
break;
case CONTROL_CHANGE:
if (ev->d.control.param == 0x00 || ev->d.control.param == 0x20) {
/* 0x00 and 0x20 are used for BANK select */
} else if (ev->d.control.param == 64) {
/* damper pedal*/
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rs->sc[ev->channel].sustain = ev->d.control.value < 64 ? 0 : 1;
} else if (ev->d.control.param == 121) {
/* reset all controllers */
} else if (ev->d.control.param == 120 || ev->d.control.param == 123) {
/* Midi panic: 120: all sound off, 123: all notes off*/
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synth_reset_channel (&(rs->sc[ev->channel]));
} else if (ev->d.control.param >= 120) {
/* params 122-127 are reserved - skip them. */
}
break;
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case PITCH_BEND:
{
float st = ((((int)ev->d.bend.hi << 7) | ev->d.bend.lo) - 8192) / 4096.0; // +/- 2 semitones
rs->sc[ev->channel].pitch_bend = powf (2, st / 12.0);
}
break;
default:
break;
}
}
/******************************************************************************
* PUBLIC API (used by lv2.c)
*/
/**
* align LV2 and internal synth buffers
* call synth_fragment as often as needed for the given LV2 buffer size
*
* @param synth synth-handle
* @param written samples written so far (offset in \ref out)
* @param nframes total samples to synthesize and write to the \out buffer
* @param out pointer to stereo output buffers
* @return end of buffer (written + nframes)
*/
static uint32_t
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synth_sound (void* synth, uint32_t written, const uint32_t nframes, float** out)
{
RSSynthesizer* rs = (RSSynthesizer*)synth;
while (written < nframes) {
uint32_t nremain = nframes - written;
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if (rs->boffset >= BUFFER_SIZE_SAMPLES) {
const uint32_t tosynth = MIN (BUFFER_SIZE_SAMPLES, nremain);
rs->boffset = BUFFER_SIZE_SAMPLES - tosynth;
synth_fragment (rs, tosynth, &(rs->buf[0][rs->boffset]), &(rs->buf[1][rs->boffset]));
}
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uint32_t nread = MIN (nremain, (BUFFER_SIZE_SAMPLES - rs->boffset));
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memcpy (&out[0][written], &rs->buf[0][rs->boffset], nread * sizeof (float));
memcpy (&out[1][written], &rs->buf[1][rs->boffset], nread * sizeof (float));
written += nread;
rs->boffset += nread;
}
return written;
}
/**
* parse raw midi-data.
*
* @param synth synth-handle
* @param data 8bit midi message
* @param size number of bytes in the midi-message
*/
static void
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synth_parse_midi (void* synth, const uint8_t* data, const size_t size)
{
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if (size < 2 || size > 3)
return;
// All messages need to be 3 bytes; except program-changes: 2bytes.
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if (size == 2 && (data[0] & 0xf0) != 0xC0)
return;
struct rmidi_event_t ev;
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ev.channel = data[0] & 0x0f;
switch (data[0] & 0xf0) {
case 0x80:
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ev.type = NOTE_OFF;
ev.d.tone.note = data[1] & 0x7f;
ev.d.tone.velocity = data[2] & 0x7f;
break;
case 0x90:
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ev.type = NOTE_ON;
ev.d.tone.note = data[1] & 0x7f;
ev.d.tone.velocity = data[2] & 0x7f;
if (ev.d.tone.velocity == 0) {
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ev.type = NOTE_OFF;
}
break;
case 0xB0:
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ev.type = CONTROL_CHANGE;
ev.d.control.param = data[1] & 0x7f;
ev.d.control.value = data[2] & 0x7f;
break;
case 0xC0:
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ev.type = PROGRAM_CHANGE;
ev.d.control.value = data[1] & 0x7f;
break;
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case 0xE0:
ev.type = PITCH_BEND;
ev.d.bend.lo = data[1] & 0x7f;
ev.d.bend.hi = data[2] & 0x7f;
break;
default:
return;
}
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synth_process_midi_event (synth, &ev);
}
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static const uint8_t jingle[] = { 71, 71, 71, 71, 71, 71, 71, 74, 67, 69, 71, 72, 72, 72, 72, 72, 71, 71, 71, 71, 71, 69, 69, 71, 69, 74, 71, 71, 71, 71, 71, 71, 71, 74, 67, 69, 71, 72, 72, 72, 72, 72, 71, 71, 71, 71, 74, 74, 72, 69, 67, 62, 62, 71, 69, 67, 62, 62, 62, 62, 71, 69, 67, 64, 64, 64, 72, 71, 69, 66, 74, 76, 74, 72, 69, 71, 62, 62, 71, 69, 67, 62, 62, 62, 62, 71, 69, 67, 64, 64, 64, 72, 71, 69, 74, 74, 74, 74, 76, 74, 72, 69, 67, 74, 71, 71, 71, 71, 71, 71, 71, 74, 67, 69, 71, 72, 72, 72, 72, 72, 71, 71, 71, 71, 71, 69, 69, 71, 69, 74, 71, 71, 71, 71, 71, 71, 71, 74, 67, 69, 71, 72, 72, 72, 72, 72, 71, 71, 71, 71, 74, 74, 72, 69, 67 };
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static void
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synth_parse_xmas (void* synth, const uint8_t* data, const size_t size)
{
RSSynthesizer* rs = (RSSynthesizer*)synth;
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if (size < 2 || size > 3)
return;
// All messages need to be 3 bytes; except program-changes: 2bytes.
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if (size == 2 && (data[0] & 0xf0) != 0xC0)
return;
struct rmidi_event_t ev;
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ev.channel = data[0] & 0x0f;
switch (data[0] & 0xf0) {
case 0x80:
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ev.type = NOTE_OFF;
ev.d.tone.note = jingle[rs->xmas_off++];
ev.d.tone.velocity = data[2] & 0x7f;
if (rs->xmas_off >= sizeof (jingle))
rs->xmas_off = 0;
break;
case 0x90:
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ev.type = NOTE_ON;
ev.d.tone.note = jingle[rs->xmas_on++];
ev.d.tone.velocity = data[2] & 0x7f;
if (rs->xmas_on >= sizeof (jingle))
rs->xmas_on = 0;
break;
case 0xB0:
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ev.type = CONTROL_CHANGE;
ev.d.control.param = data[1] & 0x7f;
ev.d.control.value = data[2] & 0x7f;
break;
case 0xC0:
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ev.type = PROGRAM_CHANGE;
ev.d.control.value = data[1] & 0x7f;
break;
default:
return;
}
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synth_process_midi_event (synth, &ev);
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}
/**
* initialize the synth
* This should be called after synth_alloc()
* as soon as the sample-rate is known
*
* @param synth synth-handle
* @param rate sample-rate
*/
static void
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synth_init (void* synth, double rate)
{
RSSynthesizer* rs = (RSSynthesizer*)synth;
rs->rate = rate;
rs->boffset = BUFFER_SIZE_SAMPLES;
const float tuning = 440;
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int c, k;
for (k = 0; k < 128; k++) {
rs->freqs[k] = (tuning / 32.0f) * powf (2, (k - 9.0) / 12.0) / rate;
assert (rs->freqs[k] < M_PI / 2); // otherwise spatialization may phase out..
}
rs->kcfilt = 12.0 / rate;
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synth_reset (synth);
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for (c = 0; c < 16; c++) {
synth_load (&rs->sc[c], rate, &synthesize_sineP, &piano_adsr);
}
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rs->xmas_on = 0;
rs->xmas_off = 0;
}
/**
* Allocate data-structure, create a handle for all other synth_* functions.
*
* This data should be freeded with \ref synth_free when the synth is no
* longer needed.
*
* The synth can only be used after calling \rev synth_init as well.
*
* @return synth-handle
*/
static void*
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synth_alloc (void)
{
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return calloc (1, sizeof (RSSynthesizer));
}
/**
* release synth data structure
* @param synth synth-handle
*/
static void
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synth_free (void* synth)
{
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free (synth);
}