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easySynth.ino
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easySynth.ino
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/*
* Copyright (c) 2022 Marcel Licence
*
* 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 3 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, see <http://www.gnu.org/licenses/>.
*
* Dieses Programm ist Freie Software: Sie können es unter den Bedingungen
* der GNU General Public License, wie von der Free Software Foundation,
* Version 3 der Lizenz oder (nach Ihrer Wahl) jeder neueren
* veröffentlichten Version, weiter verteilen und/oder modifizieren.
*
* Dieses Programm wird in der Hoffnung bereitgestellt, dass es nützlich sein wird, jedoch
* OHNE JEDE GEWÄHR,; sogar ohne die implizite
* Gewähr der MARKTFÄHIGKEIT oder EIGNUNG FÜR EINEN BESTIMMTEN ZWECK.
* Siehe die GNU General Public License für weitere Einzelheiten.
*
* Sie sollten eine Kopie der GNU General Public License zusammen mit diesem
* Programm erhalten haben. Wenn nicht, siehe <https://www.gnu.org/licenses/>.
*/
/**
* @file easySynth.ino
* @author Marcel Licence
* @date 17.12.2021
*
* @brief This file contains a simple implementation for a polyphonic synthesizer
*/
#ifdef __CDT_PARSER__
#include "cdt.h"
#endif
#include <ml_osc.h>
#include <ml_env.h>
#include <ml_filter.h>
#include <ml_waveform.h>
static struct notePlayerT *getFreeVoice(void);
/*
* Param indices for Synth_SetParam function
*/
#define SYNTH_PARAM_VEL_ENV_ATTACK 0
#define SYNTH_PARAM_VEL_ENV_DECAY 1
#define SYNTH_PARAM_VEL_ENV_SUSTAIN 2
#define SYNTH_PARAM_VEL_ENV_RELEASE 3
#define SYNTH_PARAM_FIL_ENV_ATTACK 4
#define SYNTH_PARAM_FIL_ENV_DECAY 5
#define SYNTH_PARAM_FIL_ENV_SUSTAIN 6
#define SYNTH_PARAM_FIL_ENV_RELEASE 7
#define SYNTH_PARAM_MAIN_FILT_CUTOFF 10
#define SYNTH_PARAM_MAIN_FILT_RESO 11
#define SYNTH_PARAM_VOICE_FILT_RESO 12
#define SYNTH_PARAM_VOICE_NOISE_LEVEL 13
#define SYNTH_PARAM_VOICE_PORT_TIME 14
#define SYNTH_PARAM_PITCH_BEND_RANGE 15
#define SYNTH_PARAM_MODULATION_SPEED 16
#define SYNTH_PARAM_MODULATION_PITCH 17
#define SYNTH_PARAM_OSCILLATOR_SEL 18
#define SYNTH_PARAM_VOICE_MORPH 19
#define SYNTH_PARAM_VOICE_MORPH_WAVEFORM 20
#define SYNTH_PARAM_VOICE_MORPH_LFO 21
#define SYNTH_OSC_PARAM_VOLUME 0
#define SYNTH_OSC_PARAM_PITCH 1
#define SYNTH_OSC_PARAM_WAVEFORM 2
/*
* following are the envelope indexes for easier selection
*/
#define ENV_IDX_VOLUME 0 /* envelope for the voice volume */
#define ENV_IDX_FILTER 1 /* envelope for the voice filter */
#define ENV_IDX_PITCH 2 /* envelope for the voice pitch */
#define ENV_IDX_MOD_PITCH 3 /* envelope for the voice modulation depth of pitch */
#define ENV_IDX_MOD_FILTER 4 /* envelope for the voice modulation depth of filter */
#define ENV_IDX_MOD_MORPH 5 /* envelope for the voice modulation depth of filter */
/*
* Following defines can be changed for different puprposes
*/
#ifdef ARDUINO_DISCO_F407VG
#define MAX_POLY_VOICE 6 /* max single voices, can use multiple osc */
#else
#define MAX_POLY_VOICE 8 /* max single voices, can use multiple osc */
#endif
#define MAX_POLY_OSC (3*MAX_POLY_VOICE) /* osc polyphony, always active reduces single voices max poly */
#define MIDI_NOTE_CNT 128
static uint32_t midi_note_to_add[MIDI_NOTE_CNT]; /* lookup to playback waveforms with correct frequency */
#define CHANNEL_MAX 16
/*
* set the correct count of available waveforms
*/
#define WAVEFORM_TYPE_COUNT 7
/*
* add here your waveforms
*/
//float *sine = NULL;
float static_sine[WAVEFORM_CNT];
float *sine = static_sine;
#if 0
float *saw = NULL;
#else
float saw[WAVEFORM_CNT];
#endif
//float *square = NULL;
float square[WAVEFORM_CNT];
#ifdef ESP32
float *pulse = NULL;
float *tri = NULL;
float *crappy_noise = NULL;
float *silence = NULL;
#else
float pulse[WAVEFORM_CNT];
float tri[WAVEFORM_CNT];
float crappy_noise[WAVEFORM_CNT];
float silence[WAVEFORM_CNT];
#endif
/*
* do not forget to enter the waveform pointer addresses here
*/
float *waveFormLookUp[WAVEFORM_TYPE_COUNT];
struct synth_ctrl_s
{
uint8_t sel_osc_mask;
} synthCtrl =
{
1,
};
static uint8_t lastCh = 0;
struct synth_channel_cfg_s
{
struct synth_osc_cfg_s oscCfg[3];
float glideTime;
float cutoffFreq;
struct adsrT filEnv;
struct adsrT volEnv;
};
static struct filterCoeffT filterGlobalC;
static struct filterProcT mainFilterL, mainFilterR;
#define NOTE_STACK_MAX 8
struct channelSetting_s
{
float soundFiltReso;
float soundNoiseLevel;
struct adsrT adsr_vol;
struct adsrT adsr_fil;
struct adsrT adsr_pit;
struct adsrT adsr_mod;
struct adsrT adsr_mof;
struct adsrT adsr_mph;
float pitchEnv;
/* modulation */
float modulationDepth;
float modulationSpeed;
float modulationPitch;
#ifdef PRESSURE_SENSOR_ENABLED
float pressureValue;
#endif
/* pitchbend */
float pitchBendValue;
float pitchMultiplier;
/* morph */
float morph;
float morph_lfo;
/* mono mode variables */
bool mono;
float portAdd;
float port;
float noteA;
float noteB;
float modulation;
uint32_t noteCnt;
uint32_t noteStack[NOTE_STACK_MAX];
/*
* store configuration for three different oscillators
*/
struct synth_osc_cfg_s oscCfg[3];
};
static struct channelSetting_s chCfg[CHANNEL_MAX];
static struct channelSetting_s *curChCfg = &chCfg[0];
static uint8_t selectedOsc = 0;
float voiceSink[SAMPLE_BUFFER_SIZE];
struct oscillatorT oscPlayer[MAX_POLY_OSC];
static uint32_t osc_act = 0;
struct notePlayerT
{
float lastSample[2][SAMPLE_BUFFER_SIZE];
float velocity;
bool active;
// adsr_phaseT phase;
uint8_t midiCh;
uint8_t midiNote;
// float control_sign;
float out_level;
float pitch_mod;
float morph; /* current morph value per voice */
struct filterCoeffT filterC;
struct filterProcT filterL;
struct filterProcT filterR;
float f_control_sign_slow;
struct adsr_ctrl_t adsr_volume;
struct adsr_ctrl_t adsr_filter;
struct adsr_ctrl_t adsr_pitch;
struct adsr_ctrl_t adsr_mod;
struct adsr_ctrl_t adsr_mod_f;
struct adsr_ctrl_t adsr_mph;
struct channelSetting_s *cfg;
};
struct notePlayerT voicePlayer[MAX_POLY_VOICE];
uint32_t voc_act = 0;
static float justOne = 1.0f;
void Synth_Init()
{
#ifdef ESP32
randomSeed(34547379);
#endif
/*
* we do not check if malloc was successful
* if there is not enough memory left the application will crash
*/
//sine = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
//saw = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
//square = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
#ifdef ESP32
pulse = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
tri = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
crappy_noise = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
silence = (float *)malloc(sizeof(float) * WAVEFORM_CNT);
#endif
if (sine == NULL)
{
Serial.printf("Not enough memory (1)\n");
}
#if 0
if (saw == NULL)
{
Serial.printf("Not enough memory (2)\n");
}
else
#endif
{
Serial.printf("saw ready\n");
}
#ifdef ESP32
if (square == NULL)
{
Serial.printf("Not enough memory (3)\n");
}
if (pulse == NULL)
{
Serial.printf("Not enough memory (4)\n");
}
if (tri == NULL)
{
Serial.printf("Not enough memory (5)\n");
}
if (crappy_noise == NULL)
{
Serial.printf("Not enough memory (6)\n");
}
if (silence == NULL)
{
Serial.printf("Not enough memory (7)\n");
}
#endif
/*
* let us calculate some waveforms
* - using lookup tables can save a lot of processing power later
* - but it does consume memory
*/
for (int i = 0; i < WAVEFORM_CNT; i++)
{
float val = (float)sin(((float)i) * 2.0 * M_PI * (1.0f / ((float)WAVEFORM_CNT)));
sine[i] = val;
saw[i] = (2.0f * ((float)i) / ((float)WAVEFORM_CNT)) - 1.0f;
square[i] = (i > (WAVEFORM_CNT / 2)) ? 1 : -1;
pulse[i] = (i > (WAVEFORM_CNT / 4)) ? 1.0f / 4.0f : -3.0f / 4.0f;
tri[i] = ((i > (WAVEFORM_CNT / 2)) ? (((4.0f * (float)i) / ((float)WAVEFORM_CNT)) - 1.0f) : (3.0f - ((4.0f * (float)i) / ((float)WAVEFORM_CNT)))) - 2.0f;
crappy_noise[i] = (random(1024) / 512.0f) - 1.0f;
silence[i] = 0;
}
waveFormLookUp[0] = sine;
waveFormLookUp[1] = saw;
waveFormLookUp[2] = square;
waveFormLookUp[3] = pulse;
waveFormLookUp[4] = tri;
waveFormLookUp[5] = crappy_noise;
waveFormLookUp[6] = silence;
/*
* initialize all oscillators
*/
for (int i = 0; i < MAX_POLY_OSC; i++)
{
oscillatorT *osc = &oscPlayer[i];
osc->waveForm = &waveFormLookUp[WAVEFORM_TYPE_COUNT - 1];
osc->dest[0] = voiceSink;
osc->dest[1] = voiceSink;
osc->cfg = &chCfg[0].oscCfg[0]; /* link to a configuration */
osc->pitchMod = &justOne;
}
/*
* initialize all voices
*/
for (int i = 0; i < MAX_POLY_VOICE; i++)
{
notePlayerT *voice = &voicePlayer[i];
voice->active = false;
for (int n = 0; n < SAMPLE_BUFFER_SIZE; n++)
{
voice->lastSample[0][n] = 0.0f;
voice->lastSample[1][n] = 0.0f;
}
voice->filterL.filterCoeff = &voice->filterC;
voice->filterR.filterCoeff = &voice->filterC;
voice->cfg = &chCfg[0];
}
/*
* prepare lookup for constants to drive oscillators
*/
for (int i = 0; i < MIDI_NOTE_CNT; i++)
{
float f = ((pow(2.0f, (float)(i - 69) / 12.0f) * 440.0f));
uint32_t add = (uint32_t)(f * ((float)(1ULL << 32ULL) / ((float)SAMPLE_RATE)));
midi_note_to_add[i] = add;
}
/*
* assign main filter
*/
mainFilterL.filterCoeff = &filterGlobalC;
mainFilterR.filterCoeff = &filterGlobalC;
Filter_Proc_Init(&mainFilterL);
Filter_Proc_Init(&mainFilterR);
Filter_Coeff_Init(mainFilterL.filterCoeff);
Filter_Calculate(1.0f, 1.0f, &filterGlobalC);
for (int i = 0; i < CHANNEL_MAX; i++)
{
Synth_ChannelSettingInit(&chCfg[i]);
}
}
static float filtCutoff = 1.0f;
static float filtReso = 0.5f;
static float constZero = 0.0f;
static void Synth_OscCfgInit(struct synth_osc_cfg_s *cfg)
{
cfg->pitch = 1.0f;
cfg->pitchOctave = 1;
cfg->volume = 0.0f;
cfg->selectedWaveForm = saw;
cfg->morphWaveForm = sine;
cfg->morph = &constZero;
}
static void Synth_ChannelSettingInit(struct channelSetting_s *setting)
{
setting->soundFiltReso = 0.5f;
setting->soundNoiseLevel = 0.0f;
ADSR_SetSamplerate(SAMPLE_RATE);
ASDR_Init(&setting->adsr_vol, 1.0f, 0.25f, 1.0f, 0.01f);
ASDR_Init(&setting->adsr_fil, 1.0f, 0.25f, 1.0f, 0.01f);
ASDR_Init(&setting->adsr_pit, 1.0f, 0.25f, 1.0f, 0.01f);
ASDR_Init(&setting->adsr_mod, 1.0f, 0.25f, 1.0f, 0.01f);
ASDR_Init(&setting->adsr_mof, 1.0f, 0.25f, 1.0f, 0.01f);
ASDR_Init(&setting->adsr_mph, 1.0f, 0.25f, 1.0f, 0.01f);
setting->pitchEnv = 0.5f;
setting->modulationDepth = 0.0f;
setting->modulationSpeed = 5.0f;
setting->modulationPitch = 1.0f;
setting->pitchBendValue = 0.0f;
setting->pitchMultiplier = 1.0f;
setting->morph = 0.0f;
setting->morph_lfo = 0.0f;
setting->mono = false;
setting->portAdd = 0.01f; /*!< speed of portamento */
setting->port = 1.0f;
setting->noteA = 0;
setting->noteB = 0;
setting->noteCnt = 0;
/* setting->noteStack[NOTE_STACK_MAX]; can be left uninitialized */
for (int i = 0; i < 3; i++)
{
Synth_OscCfgInit(&setting->oscCfg[i]);
setting->oscCfg->pitchMultiplier = &setting->pitchMultiplier; /* link pitch multiplier */
}
setting->oscCfg[0].volume = 1.0f;
}
void Voice_Off(uint32_t i)
{
notePlayerT *voice = &voicePlayer[i];
for (int f = 0; f < MAX_POLY_OSC; f++)
{
oscillatorT *osc = &oscPlayer[f];
if (osc->dest[0] == voice->lastSample[0])
{
osc->dest[0] = voiceSink;
osc->dest[1] = voiceSink;
osc_act -= 1;
}
}
voc_act -= 1;
}
inline
float SineNorm(float alpha_div2pi)
{
uint32_t index = ((uint32_t)(alpha_div2pi * ((float)WAVEFORM_CNT))) % WAVEFORM_CNT;
return sine[index];
}
inline
float GetModulation(uint8_t ch)
{
float modSpeed = chCfg[ch].modulationSpeed;
return chCfg[ch].modulationPitch * (SineNorm((modSpeed * ((float)millis()) / 1000.0f)));
}
const float postGain = 1.0f / ((float)MAX_POLY_OSC);
//[[gnu::noinline, gnu::optimize ("fast-math")]]
inline void Synth_Process(float *left, float *right, uint32_t len)
{
/*
* update pitch bending / modulation
*/
{
for (int i = 0; i < CHANNEL_MAX; i++)
{
chCfg[i].modulation = GetModulation(i) ;
chCfg[i].port += chCfg[i].portAdd; /* active portamento */
chCfg[i].port = chCfg[i].port > 1.0f ? 1.0f : chCfg[i].port; /* limit value to max of 1.0f */
float portVal = (((float)(chCfg[i].noteA)) * (1.0f - chCfg[i].port) + ((float)(chCfg[i].noteB)) * chCfg[i].port);
float pitchVar = chCfg[i].pitchBendValue + chCfg[i].modulationDepth * chCfg[i].modulation + portVal;
chCfg[i].pitchMultiplier = pow(2.0f, pitchVar / 12.0f);
}
}
/*
* prepare all voices
*/
for (int i = 0; i < MAX_POLY_VOICE; i++) /* one loop is faster than two loops */
{
notePlayerT *voice = &voicePlayer[i];
for (uint32_t n = 0; n < len; n++)
{
voice->lastSample[0][n] = 0;
voice->lastSample[1][n] = 0;
}
voice->pitch_mod = 1.0f + (voice->adsr_pitch.ctrl * voice->cfg->adsr_pit.w);
voice->pitch_mod *= pow(2.0f, voice->cfg->modulation * voice->adsr_mod.ctrl * voice->cfg->adsr_mod.s);
voice->morph = voice->cfg->morph;
voice->morph += voice->adsr_mph.ctrl * voice->cfg->adsr_mph.w;
voice->morph += voice->cfg->morph_lfo * voice->cfg->modulation * 2;
#ifdef PRESSURE_SENSOR_ENABLED
voice->morph += voice->cfg->pressureValue;
#endif
}
/*
* oscillator processing -> mix to voice
*/
OscProcess(oscPlayer, MAX_POLY_OSC, len);
#ifdef NOISE_ENABLED
float noise_signal[len];
for (uint32_t n = 0; n < len; n++)
{
/* generate a noise signal */
noise_signal[n] = ((random(1024) / 512.0f) - 1.0f) * soundNoiseLevel;
}
#endif
for (int i = 0; i < MAX_POLY_VOICE; i++) /* one loop is faster than two loops */
{
notePlayerT *voice = &voicePlayer[i];
if (voice->active)
{
#ifdef NOISE_ENABLED
for (uint32_t n = 0; n < len; n++)
{
/* add some noise to the voice */
voice->lastSample[0][n] += noise_signal[n];
voice->lastSample[1][n] += noise_signal[n];
}
#endif
for (uint32_t n = 0; n < len; n++)
{
if (n % 4 == 0)
{
voice->active = ADSR_Process(&voice->cfg->adsr_vol, &voice->adsr_volume);
if (voice->active == false)
{
Voice_Off(i);
}
/*
* make is slow to avoid bad things .. or crying ears
*/
(void)ADSR_Process(&voice->cfg->adsr_fil, &voice->adsr_filter);
(void)ADSR_Process(&voice->cfg->adsr_pit, &voice->adsr_pitch);
(void)ASRM_Process(&voice->cfg->adsr_mod, &voice->adsr_mod);
(void)ASRM_Process(&voice->cfg->adsr_mof, &voice->adsr_mod_f);
(void)ASRM_Process(&voice->cfg->adsr_mph, &voice->adsr_mph);
}
voice->lastSample[0][n] *= voice->adsr_volume.ctrl * voice->velocity;
voice->lastSample[1][n] *= voice->adsr_volume.ctrl * voice->velocity;
if (n % 32 == 0)
{
float ff = voice->adsr_filter.ctrl;
ff += voice->cfg->modulation * voice->adsr_mod_f.ctrl * voice->cfg->adsr_mof.s;
if (ff > 1.0f)
{
ff = 1.0f;
}
if (ff < 0.0f)
{
ff = 0.0;
}
voice->f_control_sign_slow = 0.05 * ff + 0.95 * voice->f_control_sign_slow;
Filter_Calculate(voice->f_control_sign_slow, voice->cfg->soundFiltReso, &voice->filterC);
}
}
Filter_Process_Buffer(voice->lastSample[0], &voice->filterL, len);
Filter_Process_Buffer(voice->lastSample[1], &voice->filterR, len);
/*
* push voice audio to main signal
*/
for (uint32_t n = 0; n < len; n++)
{
left[n] += voice->lastSample[0][n] * postGain;
right[n] += voice->lastSample[1][n] * postGain;
}
}
}
/*
* process main filter
*/
Filter_Process_Buffer(left, &mainFilterL, len);
Filter_Process_Buffer(right, &mainFilterR, len);
}
struct oscillatorT *getFreeOsc()
{
for (int i = 0; i < MAX_POLY_OSC ; i++)
{
//if (oscPlayer[i].isFree())
if (oscPlayer[i].dest[0] == voiceSink)
{
return &oscPlayer[i];
}
}
return NULL;
}
static struct notePlayerT *getFreeVoice(void)
{
for (int i = 0; i < MAX_POLY_VOICE ; i++)
{
if (voicePlayer[i].active == false)
{
return &voicePlayer[i];
}
}
return NULL;
}
inline void Synth_NoteOn(uint8_t ch, uint8_t note, float vel __attribute__((unused)))
{
struct notePlayerT *voice = getFreeVoice();
struct oscillatorT *osc = getFreeOsc();
lastCh = ch;
/* put note onto stack */
if (chCfg[ch].mono)
{
if (chCfg[ch].noteCnt < (NOTE_STACK_MAX - 1))
{
chCfg[ch].noteStack[chCfg[ch].noteCnt] = note;
chCfg[ch].noteCnt++;
//Status_ValueChangedIntArr("noteCnt", chCfg[ch].noteCnt, ch);
}
if (chCfg[ch].noteCnt > 1)
{
for (int i = 0; i < MAX_POLY_VOICE ; i++)
{
if ((voicePlayer[i].active) && (voicePlayer[i].midiCh == ch))
{
float diff = note - voicePlayer[i].midiNote;
voicePlayer[i].cfg->noteA = voicePlayer[i].cfg->port * ((float)voicePlayer[i].cfg->noteB) + (1.0f - voicePlayer[i].cfg->port) * voicePlayer[i].cfg->noteA;
voicePlayer[i].cfg->port = 0.0f;
voicePlayer[i].cfg->noteB += diff;
voicePlayer[i].midiNote = note;
return;
}
}
}
}
/*
* No free voice found, return otherwise crash xD
*/
if ((voice == NULL) || (osc == NULL))
{
Serial.printf("NotFree, voc: %" PRIu32 ", osc: %" PRIu32 "\n", voc_act, osc_act);
return ;
}
voice->cfg = &chCfg[ch];
voice->midiCh = ch;
voice->midiNote = note;
#ifdef MIDI_USE_CONST_VELOCITY
voice->velocity = 1.0f;
#else
voice->velocity = vel;
#endif
for (int n = 0; n < SAMPLE_BUFFER_SIZE; n++)
{
voice->lastSample[0][n] = 0.0f;
voice->lastSample[1][n] = 0.0f;
}
/* default values to avoid portamento */
voice->cfg->port = 1.0f;
voice->cfg->noteB = 0;
/* pitch mod */
voice->pitch_mod = 1.0f;
voice->morph = 0.0f;
voice->active = true;
ADSR_Start(&voice->cfg->adsr_vol, &voice->adsr_volume);
ADSR_Start(&voice->cfg->adsr_fil, &voice->adsr_filter);
ADSR_Start(&voice->cfg->adsr_pit, &voice->adsr_pitch);
ADSR_Start(&voice->cfg->adsr_mod, &voice->adsr_mod);
ADSR_Start(&voice->cfg->adsr_mof, &voice->adsr_mod_f);
ADSR_Start(&voice->cfg->adsr_mph, &voice->adsr_mph);
voice->f_control_sign_slow = voice->adsr_filter.ctrl;
/* update all values to avoid audible artifacts */
ADSR_Process(&voice->cfg->adsr_vol, &voice->adsr_volume);
ADSR_Process(&voice->cfg->adsr_fil, &voice->adsr_filter);
ADSR_Process(&voice->cfg->adsr_pit, &voice->adsr_pitch);
ASRM_Process(&voice->cfg->adsr_mod, &voice->adsr_mod);
ASRM_Process(&voice->cfg->adsr_mof, &voice->adsr_mod_f);
ASRM_Process(&voice->cfg->adsr_mph, &voice->adsr_mph);
Filter_Calculate(voice->f_control_sign_slow, voice->cfg->soundFiltReso, &voice->filterC);
voc_act += 1;
/*
* add oscillator
*/
for (int i = 0; i < 3; i++)
{
osc = getFreeOsc();
if (osc != NULL)
{
{
osc->addVal = midi_note_to_add[note];
}
osc->samplePos = 0;
osc->dest[0] = voice->lastSample[0];
osc->dest[1] = voice->lastSample[1];
osc->pan_l = 1;
osc->pan_r = 1;
osc->cfg = &chCfg[ch].oscCfg[i];
osc->cfg->pitchMultiplier = &chCfg[ch].pitchMultiplier;
osc->pitchMod = &voice->pitch_mod;
osc->cfg->morph = &voice->morph;
osc_act += 1;
}
else
{
Serial.printf("not enough free osc!\n");
break;
}
}
/*
* trying to avoid audible suprises
*/
Filter_Reset(&voice->filterL);
Filter_Reset(&voice->filterR);
Filter_Process(&voice->lastSample[0][0], &voice->filterL);
Filter_Process(&voice->lastSample[0][0], &voice->filterL);
Filter_Process(&voice->lastSample[0][0], &voice->filterL);
Filter_Process(&voice->lastSample[1][0], &voice->filterR);
Filter_Process(&voice->lastSample[1][0], &voice->filterR);
Filter_Process(&voice->lastSample[1][0], &voice->filterR);
}
inline void Synth_NoteOff(uint8_t ch, uint8_t note)
{
for (uint32_t j = 0; j < chCfg[ch].noteCnt; j++)
{
if (chCfg[ch].noteStack[j] == note)
{
for (int k = j; k < NOTE_STACK_MAX - 1; k++)
{
chCfg[ch].noteStack[k] = chCfg[ch].noteStack[k + 1];
}
chCfg[ch].noteCnt = (chCfg[ch].noteCnt > 0) ? (chCfg[ch].noteCnt - 1) : 0;
//Status_ValueChangedIntArr("noteCnt-", chCfg[ch].noteCnt, ch);
}
}
for (int i = 0; i < MAX_POLY_VOICE ; i++)
{
if ((voicePlayer[i].active) && (voicePlayer[i].midiNote == note) && (voicePlayer[i].midiCh == ch))
{
if ((voicePlayer[i].cfg->noteCnt > 0) && (voicePlayer[i].cfg->mono))
{
uint8_t midiNote = voicePlayer[i].cfg->noteStack[voicePlayer[i].cfg->noteCnt - 1];
float diff = midiNote - voicePlayer[i].midiNote;
voicePlayer[i].cfg->noteA = voicePlayer[i].cfg->port * ((float)voicePlayer[i].cfg->noteB) + (1.0f - voicePlayer[i].cfg->port) * voicePlayer[i].cfg->noteA;
voicePlayer[i].cfg->port = 0.0f;
voicePlayer[i].cfg->noteB += diff;
voicePlayer[i].midiNote = midiNote;
}
else
{
voicePlayer[i].adsr_volume.phase = release;
}
}
}
}
void Synth_ModulationWheel(uint8_t ch, float value)
{
chCfg[ch].modulationDepth = value;
}
#ifdef PRESSURE_SENSOR_ENABLED
void Synth_Pressure(uint8_t ch, float value)
{
chCfg[ch].pressureValue = value;
}
#endif
void Synth_ModulationSpeed(uint8_t ch, float value)
{
chCfg[ch].modulationSpeed = value * 10;
//Status_ValueChangedFloat("ModulationSpeed", modulationSpeed);
}
void Synth_ModulationPitch(uint8_t ch, float value)
{
chCfg[ch].modulationPitch = value * 5;
//Status_ValueChangedFloat("ModulationDepth", modulationPitch);
}
void Synth_PitchBend(uint8_t ch, float bend)
{
chCfg[ch].pitchBendValue = bend;
//Serial.printf("pitchBendValue: %0.3f\n", chCfg[ch].pitchBendValue);
}
void Synth_PortTime(float value)
{
float min = 0.02f; /* 1/(0.02 * 1000) -> 0.05s */
float max = 0.0002f; /* 1/(0.0002 * 1000) -> 5s */
curChCfg->portAdd = (pow(2.0f, value) - 1.0f) * (max - min) + min;
}
void Synth_SetCurCh(uint8_t ch, float value)
{
if (value > 0)
{
if (ch < 16)
{
curChCfg = &chCfg[ch];
//Status_ValueChangedInt("Current ch", ch);
}
}
}
void Synth_ToggleMono(uint8_t ch, float value)
{
if (value > 0)
{
curChCfg->mono = !curChCfg->mono;
//Status_LogMessage(curChCfg->mono ? "Mono" : "Poly");
}
}
void Synth_SetCfgOsc(uint8_t oscIdx, float value)
{
if (value > 0)
{
if (oscIdx < 3)
{
selectedOsc = oscIdx;
}
}
}
void Synth_SetOscParam(uint8_t slider, float value)
{
struct synth_osc_cfg_s *oscCfg = &curChCfg->oscCfg[selectedOsc];
switch (slider)
{
case SYNTH_OSC_PARAM_WAVEFORM:
{
uint8_t selWaveForm = (value) * (WAVEFORM_TYPE_COUNT);
oscCfg->selectedWaveForm = waveFormLookUp[selWaveForm];
Serial.printf("osc[%d].selWaveForm: %d\n", selectedOsc, selWaveForm);
}
break;
case SYNTH_OSC_PARAM_VOLUME:
oscCfg->volume = value;
break;
case SYNTH_OSC_PARAM_PITCH:
oscCfg->pitch = pow(2, (14.0f / 12.0f) * (value - 0.5f));
break;
case SYNTH_PARAM_VOICE_MORPH_WAVEFORM:
{
uint8_t selWaveForm = (value) * (WAVEFORM_TYPE_COUNT);
oscCfg->morphWaveForm = waveFormLookUp[selWaveForm];
Serial.printf("osc[%d].morphWaveForm: %d\n", selectedOsc, selWaveForm);
}
break;
}
}
void Synth_SetOctave(uint8_t octave, float value)
{
struct synth_osc_cfg_s *oscCfg = &curChCfg->oscCfg[selectedOsc];
if (value > 0)
{
oscCfg->pitchOctave = pow(2, octave);
}
}
struct adsrT *adsrCfg1 = &chCfg[0].adsr_mph;
struct adsrT *adsrCfg2 = &chCfg[0].adsr_fil;
struct adsrT **adsrCfg = &adsrCfg1;
void Synth_Sel_Env(uint8_t env, float value)
{
if (value > 0)
{
switch (env)
{
case ENV_IDX_VOLUME:
(*adsrCfg) = &curChCfg->adsr_vol;
break;
case ENV_IDX_FILTER:
(*adsrCfg) = &curChCfg->adsr_fil;
break;
case ENV_IDX_PITCH:
(*adsrCfg) = &curChCfg->adsr_pit;
break;
case ENV_IDX_MOD_PITCH:
(*adsrCfg) = &curChCfg->adsr_mod;
break;
case ENV_IDX_MOD_FILTER:
(*adsrCfg) = &curChCfg->adsr_mof;
break;
case ENV_IDX_MOD_MORPH:
(*adsrCfg) = &curChCfg->adsr_mph;
break;
}
}
}
void Synth_SetParam(uint8_t slider, float value)
{
switch (slider)
{
case SYNTH_PARAM_VEL_ENV_ATTACK:
if (value == 0)
{
adsrCfg1->a = 1.0f;
}
else if (value >= 0.999f)
{
adsrCfg1->a = 0.0f;
}
else
{
adsrCfg1->a = (0.00005 * pow(5000, 1.0f - value));
}