From ba84fbc9d61a9529e25be0c788630ab842242845 Mon Sep 17 00:00:00 2001
From: Holger Wirtz <wirtz@parasitstudio.de>
Date: Thu, 25 Apr 2019 14:52:27 +0200
Subject: [PATCH] Exchanged double to float data types in PluginFX. Removed
 Teensy filter (is replaced by the Dexed internal filter) and antialias
 filter.

---
 MicroDexed.ino |  34 ++---
 PluginFx.cpp   | 358 ++++++++++++++++++++++++-------------------------
 PluginFx.h     |   6 +-
 UI.cpp         |   9 +-
 UI.h           |   1 -
 5 files changed, 198 insertions(+), 210 deletions(-)

diff --git a/MicroDexed.ino b/MicroDexed.ino
index deedcd4..6ebf1cd 100644
--- a/MicroDexed.ino
+++ b/MicroDexed.ino
@@ -51,21 +51,14 @@ uint8_t ui_main_state = UI_MAIN_VOICE;
 
 AudioPlayQueue           queue1;
 AudioAnalyzePeak         peak1;
-AudioFilterStateVariable filter1;
 AudioEffectDelay         delay1;
 AudioMixer4              mixer1;
 AudioMixer4              mixer2;
-AudioFilterBiquad        antialias;
 AudioConnection          patchCord0(queue1, peak1);
-AudioConnection          patchCord1(queue1, antialias);
-AudioConnection          patchCord2(antialias, 0, filter1, 0);
-AudioConnection          patchCord3(filter1, 0, delay1, 0);
-AudioConnection          patchCord4(filter1, 0, mixer1, 0);
-AudioConnection          patchCord5(filter1, 0, mixer2, 0);
+AudioConnection          patchCord9(queue1, 0, mixer1, 0);
 AudioConnection          patchCord6(delay1, 0, mixer1, 1);
 AudioConnection          patchCord7(delay1, 0, mixer2, 2);
 AudioConnection          patchCord8(mixer1, delay1);
-AudioConnection          patchCord9(queue1, 0, mixer1, 3); // for disabling the filter
 AudioConnection          patchCord10(mixer1, 0, mixer2, 1);
 #if defined(TEENSY_AUDIO_BOARD)
 AudioOutputI2S           i2s1;
@@ -225,25 +218,18 @@ void setup()
 #endif
 
     // Init effects
-    antialias.setLowpass(0, 6000, 0.707);
-    filter1.frequency(EXP_FUNC((float)map(effect_filter_frq, 0, ENC_FILTER_FRQ_STEPS, 0, 1024) / 150.0) * 10.0 + 80.0);
-    //filter1.resonance(mapfloat(effect_filter_resonance, 0, ENC_FILTER_RES_STEPS, 0.7, 5.0));
-    filter1.resonance(EXP_FUNC(mapfloat(effect_filter_resonance, 0, ENC_FILTER_RES_STEPS, 0.7, 5.0)) * 0.044 + 0.61);
-    filter1.octaveControl(mapfloat(effect_filter_octave, 0, ENC_FILTER_OCT_STEPS, 0.0, 7.0));
     delay1.delay(0, mapfloat(effect_delay_feedback, 0, ENC_DELAY_TIME_STEPS, 0.0, DELAY_MAX_TIME));
     // mixer1 is the feedback-adding mixer, mixer2 the whole delay (with/without feedback) mixer
     mixer1.gain(0, 1.0); // original signal
     mixer1.gain(1, mapfloat(effect_delay_feedback, 0, ENC_DELAY_FB_STEPS, 0.0, 1.0)); // amount of feedback
-    mixer1.gain(0, 0.0); // filtered signal off
-    mixer1.gain(3, 1.0); // original signal on
     mixer2.gain(0, 1.0 - mapfloat(effect_delay_volume, 0, ENC_DELAY_VOLUME_STEPS, 0.0, 1.0)); // original signal
     mixer2.gain(1, mapfloat(effect_delay_volume, 0, ENC_DELAY_VOLUME_STEPS, 0.0, 1.0)); // delayed signal (including feedback)
     mixer2.gain(2, mapfloat(effect_delay_volume, 0, ENC_DELAY_VOLUME_STEPS, 0.0, 1.0)); // only delayed signal (without feedback)
 
     // just for testing:
-    dexed->fx.uiReso = 0.5;
-    dexed->fx.uiGain = 0.5;
-    dexed->fx.uiCutoff = 0.5;
+    dexed->fx.Reso = 0.5;
+    dexed->fx.Gain = 0.5;
+    dexed->fx.Cutoff = 0.5;
 
     // load default SYSEX data
     load_sysex(configuration.bank, configuration.voice);
@@ -433,7 +419,7 @@ void handleControlChange(byte inChannel, byte inCtrl, byte inValue)
         }
         break;
       case 102:  // CC 102: filter frequency
-        effect_filter_frq = map(inValue, 0, 127, 0, ENC_FILTER_FRQ_STEPS);
+/*        effect_filter_frq = map(inValue, 0, 127, 0, ENC_FILTER_FRQ_STEPS);
         if (effect_filter_frq == ENC_FILTER_FRQ_STEPS)
         {
           // turn "off" filter
@@ -447,17 +433,17 @@ void handleControlChange(byte inChannel, byte inCtrl, byte inValue)
           mixer1.gain(3, 0.0); // original signal off
         }
         filter1.frequency(EXP_FUNC((float)map(effect_filter_frq, 0, ENC_FILTER_FRQ_STEPS, 0, 1024) / 150.0) * 10.0 + 80.0);
-        handle_ui();
+        handle_ui(); */
         break;
       case 103:  // CC 103: filter resonance
-        effect_filter_resonance = map(inValue, 0, 127, 0, ENC_FILTER_RES_STEPS);
+/*        effect_filter_resonance = map(inValue, 0, 127, 0, ENC_FILTER_RES_STEPS);
         filter1.resonance(EXP_FUNC(mapfloat(effect_filter_resonance, 0, ENC_FILTER_RES_STEPS, 0.7, 5.0)) * 0.044 + 0.61);
-        handle_ui();
+        handle_ui(); */
         break;
       case 104:  // CC 104: filter octave
-        effect_filter_octave = map(inValue, 0, 127, 0, ENC_FILTER_OCT_STEPS);
+/*      effect_filter_octave = map(inValue, 0, 127, 0, ENC_FILTER_OCT_STEPS);
         filter1.octaveControl(mapfloat(effect_filter_octave, 0, ENC_FILTER_OCT_STEPS, 0.0, 7.0));
-        handle_ui();
+        handle_ui(); */
         break;
       case 105:  // CC 105: delay time
         effect_delay_time = map(inValue, 0, 127, 0, ENC_DELAY_TIME_STEPS);
diff --git a/PluginFx.cpp b/PluginFx.cpp
index c474565..2616ffd 100644
--- a/PluginFx.cpp
+++ b/PluginFx.cpp
@@ -1,26 +1,26 @@
 /**
- *
- * Copyright (c) 2013-2014 Pascal Gauthier.
- * Copyright (c) 2013-2014 Filatov Vadim.
- * 
- * Filter taken from the Obxd project :
- *   https://github.com/2DaT/Obxd
- *
- * 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, write to the Free Software Foundation,
- * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
- *
- */
+
+   Copyright (c) 2013-2014 Pascal Gauthier.
+   Copyright (c) 2013-2014 Filatov Vadim.
+
+   Filter taken from the Obxd project :
+     https://github.com/2DaT/Obxd
+
+   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, write to the Free Software Foundation,
+   Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+
+*/
 
 #define _USE_MATH_DEFINES
 #include <math.h>
@@ -28,189 +28,189 @@
 
 const float dc = 1e-18;
 
-inline static float tptpc(float& state,float inp,float cutoff) {
-	double v = (inp - state) * cutoff / (1 + cutoff);
-	double res = v + state;
-	state = res + v;
-	return res;
+inline static float tptpc(float& state, float inp, float cutoff) {
+  float v = (inp - state) * cutoff / (1 + cutoff);
+  float res = v + state;
+  state = res + v;
+  return res;
 }
 
 inline static float tptlpupw(float & state , float inp , float cutoff , float srInv) {
-	cutoff = (cutoff * srInv)*M_PI;
-	double v = (inp - state) * cutoff / (1 + cutoff);
-	double res = v + state;
-	state = res + v;
-	return res;
+  cutoff = (cutoff * srInv) * M_PI;
+  float v = (inp - state) * cutoff / (1 + cutoff);
+  float res = v + state;
+  state = res + v;
+  return res;
 }
 
 //static float linsc(float param,const float min,const float max) {
 //    return (param) * (max - min) + min;
 //}
 
-static float logsc(float param, const float min,const float max,const float rolloff = 19.0f) {
-	return ((expf(param * logf(rolloff+1)) - 1.0f) / (rolloff)) * (max-min) + min;
+static float logsc(float param, const float min, const float max, const float rolloff = 19.0f) {
+  return ((expf(param * logf(rolloff + 1)) - 1.0f) / (rolloff)) * (max - min) + min;
 }
 
 PluginFx::PluginFx() {
-    uiCutoff = 1;
-    uiReso = 0;
-    uiGain = 1;
+  Cutoff = 1;
+  Reso = 0;
+  Gain = 1;
 }
 
 void PluginFx::init(int sr) {
-    mm=0;
-    s1=s2=s3=s4=c=d=0;
-    R24=0;
-    
-    mmch = (int)(mm * 3);
-    mmt = mm*3-mmch;
-
-    sampleRate = sr;
-    sampleRateInv = 1/sampleRate;
-    float rcrate =sqrt((44000/sampleRate));
-    rcor24 = (970.0 / 44000)*rcrate;
-    rcor24Inv = 1 / rcor24;
-    
-    bright = tan((sampleRate*0.5f-10) * M_PI * sampleRateInv);
-    
-    R = 1;
-	rcor = (480.0 / 44000)*rcrate;
-    rcorInv = 1 / rcor;
-    bandPassSw = false;
-
-    pCutoff = -1;
-    pReso = -1;
-    
-    dc_r = 1.0-(126.0/sr);
-    dc_id = 0;
-    dc_od = 0;
+  mm = 0;
+  s1 = s2 = s3 = s4 = c = d = 0;
+  R24 = 0;
+
+  mmch = (int)(mm * 3);
+  mmt = mm * 3 - mmch;
+
+  sampleRate = sr;
+  sampleRateInv = 1 / sampleRate;
+  float rcrate = sqrt((44000 / sampleRate));
+  rcor24 = (970.0 / 44000) * rcrate;
+  rcor24Inv = 1 / rcor24;
+
+  bright = tanf((sampleRate * 0.5f - 10) * M_PI * sampleRateInv);
+
+  R = 1;
+  rcor = (480.0 / 44000) * rcrate;
+  rcorInv = 1 / rcor;
+  bandPassSw = false;
+
+  pCutoff = -1;
+  pReso = -1;
+
+  dc_r = 1.0 - (126.0 / sr);
+  dc_id = 0;
+  dc_od = 0;
 }
 
-inline float PluginFx::NR24(float sample,float g,float lpc) {
-    float ml = 1 / (1+g);
-    float S = (lpc*(lpc*(lpc*s1 + s2) + s3) +s4)*ml;
-    float G = lpc*lpc*lpc*lpc;
-    float y = (sample - R24 * S) / (1 + R24*G);
-    return y + 1e-8;
+inline float PluginFx::NR24(float sample, float g, float lpc) {
+  float ml = 1 / (1 + g);
+  float S = (lpc * (lpc * (lpc * s1 + s2) + s3) + s4) * ml;
+  float G = lpc * lpc * lpc * lpc;
+  float y = (sample - R24 * S) / (1 + R24 * G);
+  return y + 1e-8;
 };
 
 inline float PluginFx::NR(float sample, float g) {
-    float y = ((sample- R * s1*2 - g*s1  - s2)/(1+ g*(2*R + g))) + dc;
-    return y;
+  float y = ((sample - R * s1 * 2 - g * s1  - s2) / (1 + g * (2 * R + g))) + dc;
+  return y;
 }
 
 void PluginFx::process(float *work, int sampleSize) {
-    // very basic DC filter
-    float t_fd = work[0];
-    work[0] = work[0] - dc_id + dc_r * dc_od;
+  // very basic DC filter
+  float t_fd = work[0];
+  work[0] = work[0] - dc_id + dc_r * dc_od;
+  dc_id = t_fd;
+  for (int i = 1; i < sampleSize; i++) {
+    t_fd = work[i];
+    work[i] = work[i] - dc_id + dc_r * work[i - 1];
     dc_id = t_fd;
-    for (int i=1; i<sampleSize; i++) {
-        t_fd = work[i];
-        work[i] = work[i] - dc_id + dc_r * work[i-1];
-        dc_id = t_fd;
-        
-    }
-    dc_od = work[sampleSize-1];
-    
-    if ( uiGain != 1 ) {
-        for(int i=0; i < sampleSize; i++ )
-            work[i] *= uiGain;
-    }
-    
-    // don't apply the LPF if the cutoff is to maximum
-    if ( uiCutoff == 1 )
-        return;
-    
-    if ( uiCutoff != pCutoff || uiReso != pReso ) {
-        rReso = (0.991-logsc(1-uiReso,0,0.991));
-        R24 = 3.5 * rReso;
-        
-        float cutoffNorm = logsc(uiCutoff,60,19000);
-        rCutoff = (float)tan(cutoffNorm * sampleRateInv * M_PI);
-            
-        pCutoff = uiCutoff;
-        pReso = uiReso;
-        
-        R = 1 - rReso;
-    }
-        
-    // THIS IS MY FAVORITE 4POLE OBXd filter
-        
-    // maybe smooth this value
-    float g = rCutoff;
-    float lpc = g / (1 + g);
-    
-    for(int i=0; i < sampleSize; i++ ) {
-        float s = work[i];
-        s = s - 0.45*tptlpupw(c,s,15,sampleRateInv);
-        s = tptpc(d,s,bright);
-        
-        float y0 = NR24(s,g,lpc);
-        
-        //first low pass in cascade
-        double v = (y0 - s1) * lpc;
-        double res = v + s1;
-        s1 = res + v;
-        
-        //damping
-        s1 =atan(s1*rcor24)*rcor24Inv;
-        float y1= res;
-        float y2 = tptpc(s2,y1,g);
-        float y3 = tptpc(s3,y2,g);
-        float y4 = tptpc(s4,y3,g);
-        float mc=0.0;
-    
-        switch(mmch) {
-            case 0:
-                mc = ((1 - mmt) * y4 + (mmt) * y3);
-                break;
-            case 1:
-                mc = ((1 - mmt) * y3 + (mmt) * y2);
-                break;
-            case 2:
-                mc = ((1 - mmt) * y2 + (mmt) * y1);
-                break;
-            case 3:
-                mc = y1;
-                break;
-        }
-        
-        //half volume comp
-        work[i] = mc * (1 + R24 * 0.45);
+
+  }
+  dc_od = work[sampleSize - 1];
+
+  if ( Gain != 1 ) {
+    for (int i = 0; i < sampleSize; i++ )
+      work[i] *= Gain;
+  }
+
+  // don't apply the LPF if the cutoff is to maximum
+  if ( Cutoff == 1 )
+    return;
+
+  if ( Cutoff != pCutoff || Reso != pReso ) {
+    rReso = (0.991 - logsc(1 - Reso, 0, 0.991));
+    R24 = 3.5 * rReso;
+
+    float cutoffNorm = logsc(Cutoff, 60, 19000);
+    rCutoff = (float)tanf(cutoffNorm * sampleRateInv * M_PI);
+
+    pCutoff = Cutoff;
+    pReso = Reso;
+
+    R = 1 - rReso;
+  }
+
+  // THIS IS MY FAVORITE 4POLE OBXd filter
+
+  // maybe smooth this value
+  float g = rCutoff;
+  float lpc = g / (1 + g);
+
+  for (int i = 0; i < sampleSize; i++ ) {
+    float s = work[i];
+    s = s - 0.45 * tptlpupw(c, s, 15, sampleRateInv);
+    s = tptpc(d, s, bright);
+
+    float y0 = NR24(s, g, lpc);
+
+    //first low pass in cascade
+    float v = (y0 - s1) * lpc;
+    float res = v + s1;
+    s1 = res + v;
+
+    //damping
+    s1 = atanf(s1 * rcor24) * rcor24Inv;
+    float y1 = res;
+    float y2 = tptpc(s2, y1, g);
+    float y3 = tptpc(s3, y2, g);
+    float y4 = tptpc(s4, y3, g);
+    float mc = 0.0;
+
+    switch (mmch) {
+      case 0:
+        mc = ((1 - mmt) * y4 + (mmt) * y3);
+        break;
+      case 1:
+        mc = ((1 - mmt) * y3 + (mmt) * y2);
+        break;
+      case 2:
+        mc = ((1 - mmt) * y2 + (mmt) * y1);
+        break;
+      case 3:
+        mc = y1;
+        break;
     }
+
+    //half volume comp
+    work[i] = mc * (1 + R24 * 0.45);
+  }
 }
 
 /*
- 
- // THIS IS THE 2POLE FILTER
- 
- for(int i=0; i < sampleSize; i++ ) {
- float s = work[i];
- s = s - 0.45*tptlpupw(c,s,15,sampleRateInv);
- s = tptpc(d,s,bright);
- 
- //float v = ((sample- R * s1*2 - g2*s1 - s2)/(1+ R*g1*2 + g1*g2));
- float v = NR(s,g);
- float y1 = v*g + s1;
- //damping
- s1 = atan(s1 * rcor) * rcorInv;
- 
- float y2 = y1*g + s2;
- s2 = y2 + y1*g;
- 
- double mc;
- if(!bandPassSw)
- mc = (1-mm)*y2 + (mm)*v;
- else
- {
- 
- mc =2 * ( mm < 0.5 ?
- ((0.5 - mm) * y2 + (mm) * y1):
- ((1-mm) * y1 + (mm-0.5) * v)
- );
- }
- 
- work[i] = mc;
- }
- 
+
+  // THIS IS THE 2POLE FILTER
+
+  for(int i=0; i < sampleSize; i++ ) {
+  float s = work[i];
+  s = s - 0.45*tptlpupw(c,s,15,sampleRateInv);
+  s = tptpc(d,s,bright);
+
+  //float v = ((sample- R * s1*2 - g2*s1 - s2)/(1+ R*g1*2 + g1*g2));
+  float v = NR(s,g);
+  float y1 = v*g + s1;
+  //damping
+  s1 = atanf(s1 * rcor) * rcorInv;
+
+  float y2 = y1*g + s2;
+  s2 = y2 + y1*g;
+
+  float mc;
+  if(!bandPassSw)
+  mc = (1-mm)*y2 + (mm)*v;
+  else
+  {
+
+  mc =2 * ( mm < 0.5 ?
+  ((0.5 - mm) * y2 + (mm) * y1):
+  ((1-mm) * y1 + (mm-0.5) * v)
+  );
+  }
+
+  work[i] = mc;
+  }
+
 */
diff --git a/PluginFx.h b/PluginFx.h
index be169a5..7946493 100644
--- a/PluginFx.h
+++ b/PluginFx.h
@@ -61,9 +61,9 @@ class PluginFx {
     PluginFx();
 
     // this is set directly by the ui / parameter
-    float uiCutoff;
-    float uiReso;
-    float uiGain;
+    float Cutoff;
+    float Reso;
+    float Gain;
 
     void init(int sampleRate);
     void process(float *work, int sampleSize);
diff --git a/UI.cpp b/UI.cpp
index 2055816..a1bfd2d 100644
--- a/UI.cpp
+++ b/UI.cpp
@@ -292,7 +292,7 @@ void handle_ui(void)
               switch (ui_main_state)
               {
                 case UI_MAIN_FILTER_FRQ:
-                  if (enc[i].read() <= 0)
+/*                  if (enc[i].read() <= 0)
                     enc[i].write(0);
                   else if (enc[i].read() > ENC_FILTER_FRQ_STEPS)
                     enc[i].write(ENC_FILTER_FRQ_STEPS);
@@ -314,9 +314,10 @@ void handle_ui(void)
                   Serial.print(F("Setting filter frequency to: "));
                   Serial.println(EXP_FUNC((float)map(effect_filter_frq, 0, ENC_FILTER_FRQ_STEPS, 0, 1024) / 150.0) * 10.0 + 80.0, DEC);
 #endif
+*/
                   break;
                 case UI_MAIN_FILTER_RES:
-                  if (enc[i].read() <= 0)
+/*                  if (enc[i].read() <= 0)
                     enc[i].write(0);
                   else if (enc[i].read() > ENC_FILTER_RES_STEPS)
                     enc[i].write(ENC_FILTER_RES_STEPS);
@@ -328,9 +329,10 @@ void handle_ui(void)
                   Serial.print(F("Setting filter resonance to: "));
                   Serial.println(EXP_FUNC(mapfloat(effect_filter_resonance, 0, ENC_FILTER_RES_STEPS, 0.7, 5.0)) * 0.044 + 0.61, 2);
 #endif
+*/
                   break;
                 case UI_MAIN_FILTER_OCT:
-                  if (enc[i].read() <= 0)
+/*                  if (enc[i].read() <= 0)
                     enc[i].write(0);
                   else if (enc[i].read() > ENC_FILTER_OCT_STEPS)
                     enc[i].write(ENC_FILTER_OCT_STEPS);
@@ -340,6 +342,7 @@ void handle_ui(void)
                   Serial.print(F("Setting filter octave control to: "));
                   Serial.println(mapfloat(effect_filter_octave, 0, ENC_FILTER_OCT_STEPS, 0.0, 7.0), 2);
 #endif
+*/
                   break;
               }
               ui_show_effects_filter();
diff --git a/UI.h b/UI.h
index 3d5dd35..47fcc9e 100644
--- a/UI.h
+++ b/UI.h
@@ -54,7 +54,6 @@ extern uint8_t effect_delay_time;
 extern uint8_t effect_delay_feedback;
 extern uint8_t effect_delay_volume;
 extern bool effect_delay_sync;
-extern AudioFilterStateVariable filter1;
 extern AudioEffectDelay delay1;
 extern AudioMixer4 mixer1;
 extern AudioMixer4 mixer2;