Changed chorus mix-in.

Small fixes for interpolation.

UI.hpp

@@ -2068,7 +2068,9 @@ void set_chorus_level(uint8_t value)
   Serial.print(F("Set CHORUS_LEVEL "));
   Serial.println(value);
 #endif
-  float tmp = mapfloat(float(value), ENC_CHORUS_LEVEL_MIN, ENC_CHORUS_LEVEL_MAX, 0.0, 1.0);
+  float tmp = mapfloat(float(value), ENC_CHORUS_LEVEL_MIN, ENC_CHORUS_LEVEL_MAX, 0.0, 0.5);
+  mixer_r.gain(0, 1.0-tmp);
+  mixer_l.gain(0, 1.0-tmp);
   mixer_r.gain(2, tmp);
   mixer_l.gain(2, tmp);
   configuration.chorus_level = value;

config.h

@@ -55,9 +55,9 @@
 #define REDUCE_LOUDNESS 0
 #define USE_XFADE_DATA 1
 // CHORUS parameters
-#define CHORUS_DELAY_LENGTH_SAMPLES (15*AUDIO_BLOCK_SAMPLES) // one AUDIO_BLOCK_SAMPLES = 2.902ms; you need doubled length, e.g. delay point is 20ms, so you need up to 40ms delay!
+#define CHORUS_DELAY_LENGTH_SAMPLES (15*AUDIO_BLOCK_SAMPLES) // one AUDIO_BLOCK_SAMPLES = 2.902ms
 #define CHORUS_WAVEFORM WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
-#define CHORUS_MODULATOR_FILTER_FRQ 500
+#define CHORUS_MODULATOR_FILTER_FRQ 50 // see https://www.earlevel.com/main/2013/10/13/biquad-calculator-v2/
 #define CHORUS_MODULATOR_FILTER_Q 0.7
 
 //*************************************************************************************************

effect_modulated_delay.cpp

@@ -23,7 +23,7 @@
 
 #include <Arduino.h>
 #include <Audio.h>
-#include "limits.h"
+#include "arm_math.h"
 #include "effect_modulated_delay.h"
 #include "config.h"
 
@@ -73,50 +73,54 @@ void AudioEffectModulatedDelay::update(void)
   block = receiveWritable(0);
   modulation = receiveReadOnly(1);
 
-  if (block && modulation)
+  if (block && modulation && _modulation_intensity > 0.1)
   {
     int16_t *bp;
-    int16_t *mp;
+    float *mp;
     float mod_idx;
     float mod_number;
     float mod_fraction;
+    float mod_float[AUDIO_BLOCK_SAMPLES];
 
     bp = block->data;
-    mp = modulation->data;
+    arm_q15_to_float(modulation->data, mod_float, AUDIO_BLOCK_SAMPLES);
+    mp = mod_float;
 
     for (uint16_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++)
     {
-      // write data into circular buffer
+      // write data into circular buffer (delayline)
       if (_circ_idx >= _delay_length)
         _circ_idx = 0;
       _delayline[_circ_idx] = *bp;
 
-      // Calculate the modulation-index as a floating point number fo later interpolation
-      mod_idx = _delay_offset + ((float(*mp) / SHRT_MAX) * _modulation_intensity);
+      // Calculate the modulation-index as a floating point number for interpolation
+      mod_idx = _delay_offset + (*mp * _modulation_intensity);
       mod_fraction = modff(mod_idx, &mod_number);
 
-      // linear interpolation
-      c_mod_idx = _circ_idx - round(mod_idx);
-      c_mod_idx %= _delay_length;
+      // calculate modulation index into circular buffer
+      c_mod_idx = (_circ_idx - int(mod_number)) % _delay_length;
       if (c_mod_idx < 0) // check for negative offsets and correct them
         c_mod_idx += _delay_length;
-      if (c_mod_idx - 1 < 0)
+
+      // linear interpolation
+      if (c_mod_idx < 1)
       {
-        idx[0] = 0;
-        idx[1] = _delay_length - 1;
+        idx[0] = _delay_length - 1;
+        idx[1] = 0;
       }
       else if (c_mod_idx + 1 >= _delay_length)
       {
-        idx[0] = c_mod_idx;
-        idx[1] = 0;
+        idx[0] = 0;
+        idx[1] = c_mod_idx;
       }
       else
       {
-        idx[0] = c_mod_idx;
-        idx[1] = c_mod_idx + 1;
+        idx[0] = c_mod_idx + 1;
+        idx[1] = c_mod_idx;
       }
-      *bp = round(float(_delayline[idx[0]]) * mod_fraction + float(_delayline[idx[1]]) * (1.0 - mod_fraction));
+      *bp = round(float(_delayline[idx[0]]) * (mod_fraction) + float(_delayline[idx[1]]) * (1.0 - mod_fraction));
 
+      // push the pointers forward
       bp++; // next audio data
       mp++; // next modulation data
       _circ_idx++; // next circular buffer index
@@ -133,7 +137,7 @@ void AudioEffectModulatedDelay::update(void)
   }
 }
 
-float AudioEffectModulatedDelay::offset(float offset_value)
+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)