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m68_Delay.ino
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m68_Delay.ino
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// --------------------------------------------------------------------------
// This file is part of the NOZORI firmware.
//
// NOZORI firmware 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.
//
// NOZORI firmware 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 NOZORI firmware. If not, see <http://www.gnu.org/licenses/>.
// --------------------------------------------------------------------------
// sinus oscillator with a waveshapper
// Pot 1 : Delay time
// Pot 2 : Mod time
// Pot 3 : audio IN gain
// Pot 4 : audio in gain modulation
// Pot 5 : FeedBack
// Pot 6 : Mod FB
// CV 1 : syncro
// CV 2 : Delay time
// CV 3 : In
// CV 4 : FB
// IN 1 : In
// IN 2 : Pan
// Selecteur3 : overflow type : clip / fold / wave shape
// OUT 1 : OUT L
// OUT 2 : OUT R
inline void Delay_init_() {
// Switch to 48KHz
send_dac(0x08,0b000000001); // sampling control (usb , 250fs, 48K, clock div 2, clk out, active)
init_chaos();
}
inline void Delay_loop_() {
uint32_t tmp, delay_timeU, freq;
int32_t tmpS, CV2_value, CV3_value, CV4_value, toggle, gain, delay_time, FB;
filter16_nozori_68
test_connect_loop_68();
chaos(15); // for default mod values
toggle = get_toggle();
toggle_global = toggle;
if (CV2_connect < 60) CV2_value = CV_filter16_out[index_filter_cv2] - CV2_0V; else CV2_value = chaos_dx>>16;
if (CV3_connect < 60) CV3_value = CV_filter16_out[index_filter_cv3] - CV3_0V; else CV3_value = chaos_dy>>16;
if (CV4_connect < 60) CV4_value = CV_filter16_out[index_filter_cv4] - CV4_0V; else CV4_value = chaos_dz>>16;
CV2_value = min(0x7FFF,max(-0x7FFF,CV2_value));
CV3_value = min(0x7FFF,max(-0x7FFF,CV3_value));
CV4_value = min(0x7FFF,max(-0x7FFF,CV4_value));
// delay time
if (CV1_connect < 60) { // connect mode
freq = (CV_filter16_out[index_filter_pot1] + 4095) / 8192;
tmp = (CV_filter16_out[index_filter_pot2] + 5461)/10923; // from 0.5 to 6.5 // i.e: 0..6 // i.e :7 values
clock_diviseur = tab_diviseur[freq] * tab_diviseur2[tmp];;
clock_multiplieur = tab_multiplieur[freq] * tab_multiplieur2[tmp];;
}
else
{
delay_time = CV_filter16_out[index_filter_pot1];
tmpS = CV2_value>>1;
tmpS *= CV_filter16_out[index_filter_pot2];
tmpS >>= 17;
delay_time += tmpS;
delay_time = min(max(0, delay_time), 0xFFFF);
delay_timeU = delay_time;
delay_timeU *= delay_timeU;
delay_time_global = delay_timeU >> 8; // on reste sur 24 bits
}
// gain
gain = CV_filter16_out[index_filter_pot3];
tmpS = CV3_value>>1;
tmpS *= CV_filter16_out[index_filter_pot4];
tmpS >>= 15;
gain += tmpS;
gain = min(max(0, gain), 0xFFFF);
gain_global = gain << 8; // 24 bits
// feedback
FB = CV_filter16_out[index_filter_pot5];
tmpS = CV4_value>>1;
tmpS *= CV_filter16_out[index_filter_pot6];
tmpS >>= 15;
FB += tmpS;
FB = min(max(0, FB), 0xFFFF);
if (toggle == 2) { FB >> 2; }
FB_global = FB << 8; // 24 bits
led2((CV2_value+0x7FFF)>>7);
led4((CV3_value+0x7FFF)>>7);
}
inline void Delay_audio_() {
int32_t audio_in, out, tmpS;
int32_t audio_out, delay_out, out1, out2;
uint32_t delay_time, FB, gain, delay_time_LSB, read_point, toggle;
toggle = toggle_global;
// syncro
nb_tick = min(0x0007FFFF, nb_tick+1); // to prvent overflow with multiplier
if( (last_clock_ == 0) && (CV1_connect < 60) && (CV_filter16_out[index_filter_cv1] > 0xB000) ) { // mode syncro, on a une syncro
last_clock_ = 1;
nb_tick /= clock_multiplieur; // reversed because we use the period, not the frequency
nb_tick *= clock_diviseur;
delay_time_global = min(Max_Delay, nb_tick)<<9;
nb_tick = 0;
}
else if ( (CV1_connect < 60) && (CV_filter16_out[index_filter_cv1] < 0xA000) ) {
last_clock_ = 0;
}
/////////////////////
if (IN1_connect < 60)
audio_in = audio_inL ^0x80000000;
else
audio_in = 0;
audio_in >>= 9;
filter_audio_in = ((filter_audio_in * 255) + audio_in) >> 8;
audio_in -= filter_audio_in; // hip on audio in
index_ecriture = (index_ecriture+1) & Max_Delay;
delay_time = delay_time_global;
delay_time = filter(delay_time, delay_time_save, 8);
delay_time_save = delay_time;
gain = filter(gain_global, gain_save, 8);
gain_save = gain;
FB = filter(FB_global, FB_save, 8);
FB_save = FB;
FB = fast_sin(FB<<6); // sinusoidal curve
FB = max(0x7FFFFFFF, FB) - 0x7FFFFFFF;
FB <<= 1;
FB = min(0xFF000000, FB);
FB += FB>>8;
if(toggle == 2) FB >>= 1;
delay_time_LSB = delay_time & 0x1FF; // on garde les 9 bits de poinds faible pour interpoler
delay_time >>= 9; // on les suprime pour ne garder que l'index sur 15 bit (taille du buffer)
read_point = (index_ecriture - delay_time) & Max_Delay;
out1 = delay_line.S16[read_point];
out2 = delay_line.S16[(read_point-1) & Max_Delay];
out2 -= out1;
out2 *= delay_time_LSB;
out1 += out2 >> 9;
delay_out = out1 * (FB>>16);
delay_out >>= 16;
audio_in >>= 8;
audio_in *= (gain>>8);
audio_in >>= 15;
audio_out = audio_in-delay_out;
switch(toggle) { //clip type
case 0 :
audio_out = max(-0x7FFF, min(0x7FFF, audio_out));
break;
case 1 :
if(audio_out > 0x7FFF) audio_out = 0xFFFE - audio_out;
if(audio_out < -0x7FFF) audio_out = -0xFFFE - audio_out;
break;
case 2 :
audio_out = ((fast_sin(audio_out<<15)^0x80000000)>>16);
break;
}
out = (audio_out<<16)^0x80000000;
macro_out_pan
audio_out = audio_in-delay_out;
switch(toggle) { //clip type
case 0 :
audio_out = max(-0x7FFF, min(0x7FFF, audio_out));
break;
case 1 :
if(audio_out > 0x7FFF) audio_out = 0xFFFE - audio_out;
if(audio_out < -0x7FFF) audio_out = -0xFFFE - audio_out;
break;
case 2 :
audio_out = ((fast_sin(audio_out<<16)^0x80000000)>>16);
break;
}
delay_line.S16[index_ecriture] = audio_out;
}