New try for linear interpolation.

Added a lowpass after chorus output.

MicroMDAEPiano.ino

@@ -58,6 +58,8 @@ AudioEffectModulatedDelay   modchorus_r;
 AudioEffectModulatedDelay   modchorus_l;
 AudioSynthWaveform       modulator;
 AudioFilterBiquad        modulator_filter;
+AudioFilterBiquad        chorus_filter_r;
+AudioFilterBiquad        chorus_filter_l;
 AudioConnection          patchCord0(queue_r, peak_r);
 AudioConnection          patchCord1(queue_l, peak_l);
 AudioConnection          patchCord2(queue_r, freeverb_r);
@@ -70,20 +72,24 @@ AudioConnection          patchCord8(modulator_filter, inverter);
 AudioConnection          patchCord9(inverter, 0, modchorus_l, 1);
 AudioConnection          patchCord10(queue_r, 0, mixer_r, 0);
 AudioConnection          patchCord11(queue_l, 0, mixer_l, 0);
-AudioConnection          patchCord12(modchorus_r, 0, mixer_r, 2);
+AudioConnection          patchCord12(modchorus_r, chorus_filter_r);
-AudioConnection          patchCord13(modchorus_l, 0, mixer_l, 2);
+AudioConnection          patchCord13(modchorus_l, chorus_filter_l);
-AudioConnection          patchCord14(freeverb_r, 0, mixer_r, 1);
+AudioConnection          patchCord14(chorus_filter_r, 0, mixer_r, 2);
-AudioConnection          patchCord15(freeverb_l, 0, mixer_l, 1);
+AudioConnection          patchCord15(chorus_filter_l, 0, mixer_l, 2);
-AudioConnection          patchCord16(mixer_r, volume_r);
+AudioConnection          patchCord16(modchorus_r, 0, mixer_r, 2);
-AudioConnection          patchCord17(mixer_l, volume_l);
+AudioConnection          patchCord17(modchorus_l, 0, mixer_l, 2);
+AudioConnection          patchCord18(freeverb_r, 0, mixer_r, 1);
+AudioConnection          patchCord19(freeverb_l, 0, mixer_l, 1);
+AudioConnection          patchCord20(mixer_r, volume_r);
+AudioConnection          patchCord21(mixer_l, volume_l);
 #ifdef USB_AUDIO
 AudioOutputUSB           usb1;
-AudioConnection          patchCord18(volume_r, 0, usb1, 0);
+AudioConnection          patchCord22(volume_r, 0, usb1, 0);
-AudioConnection          patchCord19(volume_l, 0, usb1, 1);
+AudioConnection          patchCord23(volume_l, 0, usb1, 1);
 #endif
 AudioOutputI2S           i2s1;
-AudioConnection          patchCord20(volume_r, 0, i2s1, 0);
+AudioConnection          patchCord24(volume_r, 0, i2s1, 0);
-AudioConnection          patchCord21(volume_l, 0, i2s1, 1);
+AudioConnection          patchCord25(volume_l, 0, i2s1, 1);
 AudioControlSGTL5000     sgtl5000_1;
 
 // Objects
@@ -238,8 +244,10 @@ void setup()
   inverter.gain(-1.0); // change phase for second modulated delay (faked stereo mode)
   modchorus_r.offset(15.0);
   modchorus_r.intensity(1.0);
+  chorus_filter_r.setLowpass(0, 10000.0, CHORUS_MODULATOR_FILTER_Q);
   modchorus_l.offset(15.0);
   modchorus_l.intensity(1.0);
+  chorus_filter_l.setLowpass(0, 10000.0, CHORUS_MODULATOR_FILTER_Q);
 
   // internal mixing of original signal(0), reverb(1) and chorus(2)
   mixer_r.gain(VOL_MAIN, 0.5);

config.h

@@ -58,12 +58,12 @@
 // 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_WAVEFORM WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
-#define CHORUS_MODULATOR_FILTER_FRQ 1000
+#define CHORUS_MODULATOR_FILTER_FRQ 500
 #define CHORUS_MODULATOR_FILTER_Q 0.7
 //#define CHORUS_INTERPOLATION_MODE 0       // no interpolation
-#define CHORUS_INTERPOLATION_MODE 1       // linear interpolation
+//#define CHORUS_INTERPOLATION_MODE 1       // linear interpolation
 //#define CHORUS_INTERPOLATION_MODE Catmull   // spline interpolation
-#define CHORUS_INTERPOLATION_WINDOW_SIZE 3  // only uneven numbers bigger than 3!!!
+#define CHORUS_INTERPOLATION_WINDOW_SIZE 5  // only uneven numbers bigger than 3!!!
 
 //*************************************************************************************************
 //* DEBUG OUTPUT SETTINGS

effect_modulated_delay.cpp

@@ -106,10 +106,10 @@ void AudioEffectModulatedDelay::update(void)
       // The index is located around the half of the delay length multiplied by the current amount of the modulator
       mod_idx = _delay_offset + ((float(*mp) / SHRT_MAX) * _modulation_intensity);
       mod_fraction = modff(mod_idx, &mod_number);
-#ifdef CHORUS_INTERPOLATION_MODE
-      // Spline interpolation
       // Generate a an array with the size of CHORUS_INTERPOLATION_WINDOW_SIZE of x/y values around mod_idx for interpolation
-      int16_t c_mod_idx = _circ_idx - int(round(mod_idx)); // This is the pointer to the value in the circular buffer at the current modulation index
+#ifdef CHORUS_INTERPOLATION_MODE
+      // spline interpolation
+      c_mod_idx = _circ_idx - int(mod_idx); // This is the pointer to the value in the circular buffer at the current modulation index
       for (j = (CHORUS_INTERPOLATION_WINDOW_SIZE / -2), c = 0; j <= (CHORUS_INTERPOLATION_WINDOW_SIZE / 2); j++, c++)
       {
         int16_t jc_mod_idx = (c_mod_idx + j) % _delay_length; // The modulation index pointer plus the value of the current window pointer
@@ -121,26 +121,29 @@ void AudioEffectModulatedDelay::update(void)
       }
       *bp = int(round(spline->value(mod_fraction))); // use spline interpolated value
 #else
-      // Simple interpolation
+      // linear interpolation
-      int16_t c_mod_idx = (_circ_idx + int(round(mod_idx))) % _delay_length;
+      c_mod_idx = _circ_idx - round(mod_idx);
-      float value1, value2;
+      if (mod_idx < 0.0) // c_mod_idx is the pointer to the value in the circular buffer at the current modulation index
-
+        c_mod_idx--;
-      if (c_mod_idx < 0)
+      c_mod_idx %= _delay_length; // The modulation index pointer plus the value of the current window pointer
+      if (c_mod_idx < 0) // check for negative offsets and correct them
+        c_mod_idx += _delay_length;
+      if (c_mod_idx - 1 < 0)
       {
-        value1 = _delayline[_delay_length + c_mod_idx - 1];
+        idx[0] = 0;
-        value2 = _delayline[_delay_length + c_mod_idx];
+        idx[1] = _delay_length - 1;
       }
-      else
+      else if (c_mod_idx + 1 >= _delay_length)
       {
-        value1 = _delayline[c_mod_idx - 1];
+        idx[0] = c_mod_idx;
-        value2 = _delayline[c_mod_idx];
+        idx[1] = 0;
       }
-
-      if (mod_fraction < 0)
-        *bp = int(round((1.0 + mod_fraction) * value1 + fabs(mod_fraction) * value2));
       else
-        *bp = int(round(mod_fraction * value1 + (1.0 - mod_fraction) * value2));
+      {
-
+        idx[0] = c_mod_idx;
+        idx[1] = c_mod_idx + 1;
+      }
+      *bp = round(float(_delayline[idx[0]]) * (1.0 - mod_fraction) + float(_delayline[idx[1]]) * mod_fraction);
 #endif
       bp++; // next audio data
       mp++; // next modulation data

effect_modulated_delay.h

@@ -56,10 +56,13 @@ class AudioEffectModulatedDelay :
     uint16_t _delay_offset;
     uint16_t _modulation_intensity;
     uint16_t _delay_length;
+    int16_t c_mod_idx;
 #ifdef CHORUS_INTERPOLATION_MODE
     float x[CHORUS_INTERPOLATION_WINDOW_SIZE];
     float y[CHORUS_INTERPOLATION_WINDOW_SIZE];
     Spline* spline;
+#else
+    uint16_t idx[2];
 #endif
 };