Trying again with Chamberlin filter for modulation to smooth the used

modulation waveform.
Renamed some vars for better reading.
Made the used core algorithm easier - hopefully it works and the sound is
better than before.

MicroMDAEPiano.ino

@@ -57,14 +57,16 @@ AudioAmplifier           inverter;
 AudioEffectModulatedDelay   modchorus_r;
 AudioEffectModulatedDelay   modchorus_l;
 AudioSynthWaveform       modulator;
+AudioFilterStateVariable modulator_filter;
 AudioConnection          patchCord0(queue_r, peak_r);
 AudioConnection          patchCord1(queue_l, peak_l);
 AudioConnection          patchCord2(queue_r, freeverb_r);
 AudioConnection          patchCord3(queue_l, freeverb_l);
 AudioConnection          patchCord4(queue_r, 0, modchorus_r, 0);
 AudioConnection          patchCord5(queue_l, 0, modchorus_l, 0);
-AudioConnection          patchCord6(modulator, 0, modchorus_r, 1);
+AudioConnection          patchCord6(modulator, 0, modulator_filter, 1);
-AudioConnection          patchCord8(modulator, inverter);
+AudioConnection          patchCord7(modulator_filter, 0, modchorus_r, 1);
+AudioConnection          patchCord8(modulator_filter, 0, inverter, 0);
 AudioConnection          patchCord9(inverter, 0, modchorus_l, 1);
 AudioConnection          patchCord10(queue_r, 0, mixer_r, 0);
 AudioConnection          patchCord11(queue_l, 0, mixer_l, 0);
@@ -234,6 +236,9 @@ void setup()
   modulator.amplitude(1.0);
   modulator.offset(0.0);
   inverter.gain(-1.0); // change phase for second modulated delay (faked stereo mode)
+  modulator_filter.frequency(50);
+  modulator_filter.resonance(0.6);
+  modulator_filter.octaveControl(1);
   modchorus_r.offset(15.0);
   modchorus_l.offset(15.0);
 

config.h

@@ -56,10 +56,7 @@
 #define USE_XFADE_DATA 1
 // CHORUS parameters
 #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_WAVEFORM WAVEFORM_SINE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
-//#define CHORUS_MODULATOR_FILTER_FRQ 10 // see https://www.earlevel.com/main/2013/10/13/biquad-calculator-v2/
-//#define CHORUS_MODULATOR_FILTER_Q 0.7
-#define CHORUS_MODULATOR_BIQUAD 1
 //*************************************************************************************************
 //* DEBUG OUTPUT SETTINGS
 //*************************************************************************************************

effect_modulated_delay.cpp

@@ -46,7 +46,7 @@ boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
   _delayline = NULL;
   _delay_length = 0;
   _delay_offset = 0.0;
-  _circ_idx = 0;
+  _cb_index = 0;
 
   if (delayline == NULL) {
     return (false);
@@ -58,17 +58,6 @@ boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
   _delayline = delayline;
   _delay_length = d_length;
 
-#ifdef CHORUS_MODULATOR_BIQUAD
-  filter_lp_mod.numStages = 1;
-  filter_lp_mod.pState =  filter_lp_state;
-  filter_lp_mod.pCoeffs = filter_lp_coeffs;
-
-  filter_lp_coeffs[0] = 0.072959657268266670;
-  filter_lp_coeffs[1] = 0.072959657268266670;
-  filter_lp_coeffs[2] = 0.0;
-  filter_lp_coeffs[3] = 0.854080685463466605;
-  filter_lp_coeffs[4] = 0.0;
-#endif
   return (true);
 }
 
@@ -87,7 +76,7 @@ void AudioEffectModulatedDelay::update(void)
   {
     int16_t *bp;
     float *mp;
-    float mod_idx;
+    float mod_index;
     float mod_number;
     float mod_fraction;
     float modulation_f32[AUDIO_BLOCK_SAMPLES];
@@ -95,51 +84,29 @@ void AudioEffectModulatedDelay::update(void)
     bp = block->data;
     arm_q15_to_float(modulation->data, modulation_f32, AUDIO_BLOCK_SAMPLES);
     mp = modulation_f32;
-#ifdef CHORUS_MODULATOR_BIQUAD
-    arm_biquad_cascade_df1_f32(&filter_lp_mod, modulation_f32, modulation_f32, AUDIO_BLOCK_SAMPLES);
-#endif
 
     for (uint16_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++)
     {
       // write data into circular buffer (delayline)
-      if (_circ_idx >= _delay_length)
+      if (_cb_index >= _delay_length)
-        _circ_idx = 0;
+        _cb_index = 0;
-      _delayline[_circ_idx] = *bp;
+      _delayline[_cb_index] = *bp;
 
       // Calculate the modulation-index as a floating point number for interpolation
-      mod_idx = *mp * (1 - MODULATION_MAX_FACTOR) * _delay_length;
+      mod_index = *mp * (1 - MODULATION_MAX_FACTOR) * _delay_length; // "(1 - MODULATION_MAX_FACTOR) * _delay_length" means: maximum bytes of modulation allowed by given delay length
-      mod_fraction = modff(mod_idx, &mod_number);
+      mod_fraction = modff(mod_index, &mod_number); // split float of mod_index into integer (= mod_number) and fraction part
 
       // calculate modulation index into circular buffer
-      c_mod_idx = (_circ_idx - _delay_offset - int(mod_number)) % _delay_length;
+      cb_mod_index = (_cb_index - (_delay_offset + int(mod_index))) % _delay_length;
-      if (c_mod_idx < 0) // check for negative offsets and correct them
+      if (cb_mod_index < 0) // check for negative offsets and correct them
-        c_mod_idx += _delay_length;
+        cb_mod_index += _delay_length;
-
-      // linear interpolation
-      if (c_mod_idx == 0)
-      {
-        idx[0] = _delay_length - 1;
-        idx[1] = 0;
-      }
-      else if (c_mod_idx == _delay_length - 1)
-      {
-        idx[0] = 0;
-        idx[1] = _delay_length - 1;
-      }
-      else
-      {
-        idx[0] = c_mod_idx;
-        idx[1] = c_mod_idx + 1;
-      }
-      if (mod_idx < 0.0)
-        *bp = round(float(_delayline[idx[0]]) * fabs(mod_fraction) + float(_delayline[idx[1]]) * (1.0 - fabs(mod_fraction)));
-      else
-        *bp = round(float(_delayline[idx[0]]) * (1.0 - mod_fraction) + float(_delayline[idx[1]]) * mod_fraction);
 
+      *bp = round(float(_delayline[cb_mod_index]) * mod_fraction + float(_delayline[cb_mod_index + 1]) * (1.0 - mod_fraction));
+      
       // push the pointers forward
       bp++; // next audio data
       mp++; // next modulation data
-      _circ_idx++; // next circular buffer index
+      _cb_index++; // next circular buffer index
     }
   }
 

effect_modulated_delay.h

@@ -34,7 +34,7 @@
 //                A u d i o E f f e c t M o d u l a t e d D e l a y
 // Written by Pete (El Supremo) Jan 2014
 // 140219 - correct storage class (not static)
-// 190527 - added modulation input handling (by Holger Wirtz)
+// 190527 - added modulation input handling (Aug 2019 by Holger Wirtz)
 
 class AudioEffectModulatedDelay :
   public AudioStream
@@ -50,18 +50,11 @@ class AudioEffectModulatedDelay :
 
   private:
     audio_block_t *inputQueueArray[2];
-    int16_t *_delayline;
+    int16_t *_delayline;  // pointer for the circular buffer
-    uint16_t _circ_idx;
+    uint16_t _cb_index;   // current write pointer of the circular buffer
-    uint16_t _delay_offset;
+    uint16_t _delay_offset; // number of samples for the read offset of the modulation inside the circular buffer
-    uint16_t _delay_length;
+    uint16_t _delay_length; // calculated number of samples of the delay
-    int16_t c_mod_idx;
+    int16_t cb_mod_index; // current read pointer with modulation for the circular buffer 
-    uint16_t idx[2];
-#ifdef CHORUS_MODULATOR_BIQUAD
-    arm_biquad_casd_df1_inst_f32 filter_lp_mod;
-    float filter_lp_state[4];
-    float filter_lp_coeffs[5];
-#endif
-
 };
 
 #endif