Added function for LP coefficient calculation based on frequency and q.

Small fixes for calculating the delay offset.

effect_modulated_delay.cpp

@@ -26,9 +26,6 @@
 #include "arm_math.h"
 #include "effect_modulated_delay.h"
 #include "config.h"
-#include "DCfilter.h"
-#include "OnePoleLP.h"
-#include "limits.h"
 
 extern config_t configuration;
 
@@ -52,7 +49,6 @@ boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
   _delay_length = 0;
   _delay_offset = 0.0;
   _cb_index = 0;
-  z1 = 0;
 
   if (delayline == NULL) {
     return (false);
@@ -63,13 +59,11 @@ boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
 
   _delayline = delayline;
   _delay_length = d_length;
+  memset(_delayline, 0, d_length);
 
   set_modulator_filter_coeffs();
   modulator_filter_data = {1, modulator_filter_state, modulator_filter_coeffs};
 
-  lp = new OnePoleLP();
-  dcFilter = new DCfilter(0.9);
-
   return (true);
 }
 
@@ -80,12 +74,31 @@ void AudioEffectModulatedDelay::set_modulator_filter_coeffs(void)
   // Example: https://web.fhnw.ch/technik/projekte/eit/Fruehling2016/MuelZum/html/parametric_equalizer_example_8c-example.html
   // Coeeficients calculated with https://arachnoid.com/BiQuadDesigner/index.html
 
-  // SR = 44110, Fc = 20 Hz; Q=0.707
-  modulator_filter_coeffs[0] = 5.06973332e-7;  // b0
-  modulator_filter_coeffs[1] = 1.01394666e-6;  // b1
-  modulator_filter_coeffs[2] = modulator_filter_coeffs[0];  // b2
-  modulator_filter_coeffs[3] = 1.99798478;  // -a1
-  modulator_filter_coeffs[4] = -0.99798681;  // -a2
+  double frequency = 20.0;
+  double q = 0.1;
+
+  // low-pass
+  double w0 = frequency * (2 * M_PI / AUDIO_SAMPLE_RATE_EXACT);
+  double sinW0 = sin(w0);
+  double alpha = sinW0 / ((double)q * 2.0);
+  double cosW0 = cos(w0);
+  //double scale = 1073741824.0 / (1.0 + alpha); // for integers
+  double scale = 1.0 / (1.0 + alpha);
+
+  modulator_filter_coeffs[0] = ((1.0 - cosW0) / 2.0) * scale;  // b0
+  modulator_filter_coeffs[1] = (1.0 - cosW0) * scale;          // b1
+  modulator_filter_coeffs[2] = modulator_filter_coeffs[0];     // b2
+  modulator_filter_coeffs[3] = (2.0 * cosW0) * scale;          // -a1
+  modulator_filter_coeffs[4] = (-1.0 - alpha) * scale;         // -a2
+
+  /*
+    // SR = 44110, Fc = 20 Hz; Q=0.707
+    modulator_filter_coeffs[0] = 5.06973332e-7;  // b0
+    modulator_filter_coeffs[1] = 1.01394666e-6;  // b1
+    modulator_filter_coeffs[2] = modulator_filter_coeffs[0];  // b2
+    modulator_filter_coeffs[3] = 1.99798478;  // -a1
+    modulator_filter_coeffs[4] = -0.99798681;  // -a2
+  */
 }
 
 void AudioEffectModulatedDelay::update(void)
@@ -135,14 +148,7 @@ void AudioEffectModulatedDelay::update(void)
       else
         cb_mod_index_neighbor = cb_mod_index - 1;
 
-      //*bp = round(float(_delayline[cb_mod_index]) * mod_fraction + float(_delayline[cb_mod_index_neighbor]) * (1.0 - mod_fraction));
-      *bp = (round(float(_delayline[cb_mod_index]) * mod_fraction + float(_delayline[cb_mod_index_neighbor]) * (1.0 - mod_fraction))+z1)/2;
-      z1 = *bp;
-
-      float bp_f = *bp / float(SHRT_MAX);
-      lp->tick(&bp_f, 0.95f);
-      *bp = round(bp_f * SHRT_MAX);
-      *bp = dcFilter->next(*bp);
+      *bp = round(float(_delayline[cb_mod_index]) * mod_fraction + float(_delayline[cb_mod_index_neighbor]) * (1.0 - mod_fraction));
 
       // push the pointers forward
       bp++; // next audio data
@@ -163,13 +169,13 @@ void AudioEffectModulatedDelay::update(void)
 
 float AudioEffectModulatedDelay::offset(float offset_value) // in ms
 {
-  uint16_t offset_frames = (offset_value / 1000) * AUDIO_SAMPLE_RATE;
-  if (offset_frames > _delay_length * MODULATION_MAX_FACTOR)
-    _delay_offset = _delay_length * MODULATION_MAX_FACTOR;
-  else if (offset_frames <= _delay_length * (1 - MODULATION_MAX_FACTOR))
-    _delay_offset = _delay_length * (1 - MODULATION_MAX_FACTOR);
+  uint16_t offset_frames = floor((offset_value / 1000) * AUDIO_SAMPLE_RATE);
+  if (offset_frames > round( _delay_length * MODULATION_MAX_FACTOR))
+    _delay_offset = floor(_delay_length * MODULATION_MAX_FACTOR);
+  else if (offset_frames <= round(_delay_length * (1 - MODULATION_MAX_FACTOR)))
+    _delay_offset = floor(_delay_length * (1 - MODULATION_MAX_FACTOR));
   else
     _delay_offset = offset_frames;
 
-  return (offset_frames / AUDIO_SAMPLE_RATE * 1000);
+  return (floor(offset_frames / AUDIO_SAMPLE_RATE * 1000));
 }

effect_modulated_delay.h

@@ -27,8 +27,6 @@
 #include "Arduino.h"
 #include "AudioStream.h"
 #include "config.h"
-#include "DCfilter.h"
-#include "OnePoleLP.h"
 
 #define MODULATION_MAX_FACTOR 0.75
 
@@ -61,9 +59,6 @@ class AudioEffectModulatedDelay :
     arm_biquad_casd_df1_inst_f32 modulator_filter_data;
     float32_t modulator_filter_state[4];
     float32_t modulator_filter_coeffs[5];
-    int16_t z1;
-    OnePoleLP *lp;
-    DCfilter *dcFilter;
 };
 
 #endif

name.c

@@ -1,5 +1,5 @@
 #include "usb_names.h"
-#define MIDI_NAME   {'M','i','c','r','o','M','D','A','E','p','i','a','n','o'}
+#define MIDI_NAME   {'M','i','c','r','o','M','D','A','E','P','i','a','n','o'}
 #define MIDI_NAME_LEN 14
 
 // Do not change this part.  This exact format is required by USB.