wirtz
/
MicroMDAEPiano
mirror of https://codeberg.org/dcoredump/MicroMDAEPiano.git
1
0
Fork 0
Code Issues Releases Wiki Activity

Save the current state.

Browse Source
master
Holger Wirtz 5 years ago
parent 449c7f72d1
commit 8d8ee66397
6 changed files with 142 additions and 57 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 89
      DCfilter.h
  2. 26
      MicroMDAEPiano.ino
  3. 20
      OnePoleLP.h
  4. 2
      config.h
  5. 55
      effect_modulated_delay.cpp
  6. 7
      effect_modulated_delay.h

89
DCfilter.h
Unescape View File

@ -0,0 +1,89 @@
/*
* DCfilter.h
*
* Copyright 2012 Tim Barrass.
*
* This file is part of Mozzi.
*
* Mozzi is licensed under a Creative Commons Attribution-NonCommercial-ShareAlike 4.0 International License.
*
*/
#ifndef DCFILTER_H
#define DCFILTER_H
/*
tb2010 adapted from:
robert bristow-johnson, DSP Trick: Fixed-Point DC Blocking Filter with Noise-Shaping
http://www.dspguru.com/book/export/html/126
y[n] = x[n] - x[n-1] + a * y[n-1]
Where y[n] is the output at the current time n, and x[n] is the input at the current time n.
also, see DC Blocker Algorithms, http://www.ingelec.uns.edu.ar/pds2803/materiales/articulos/04472252.pdf
*/
/**
A DC-blocking filter useful for highlighting changes in control signals.
The output of the filter settles to 0 if the incoming signal stays constant. If the input changes, the
filter output swings to track the change and eventually settles back to 0.
*/
class DCfilter
{
public:
/**
Instantiate a DC-blocking filter.
@param pole sets the responsiveness of the filter,
how long it takes to settle to 0 if the input signal levels out at a constant value.
*/
DCfilter(float pole):acc(0),prev_x(0),prev_y(0)
{
A = (int)(32768.0*(1.0 - pole));
}
/* almost original
// timing: 20us
int next(int x)
{
setPin13High();
acc -= prev_x;
prev_x = (long)x<<15;
acc += prev_x;
acc -= A*prev_y;
prev_y = acc>>15; // quantization happens here
int filtered = (int)prev_y;
// acc has y[n] in upper 17 bits and -e[n] in lower 15 bits
setPin13Low();
return filtered;
}
*/
/**
Filter the incoming value and return the result.
@param x the value to filter
@return filtered signal
*/
// timing :8us
inline
int next(int x)
{
acc += ((long)(x-prev_x)<<16)>>1;
prev_x = x;
acc -= (long)A*prev_y; // acc has y[n] in upper 17 bits and -e[n] in lower 15 bits
prev_y = (acc>>16)<<1; // faster than >>15 but loses bit 0
if (acc & 32784) prev_y += 1; // adds 1 if it was in the 0 bit position lost in the shifts above
return prev_y;
}
private:
long acc;
int prev_x, prev_y,A;
};
/**
@example 05.Control_Filters/DCFilter/DCFilter.ino
This example demonstrates the DCFilter class.
*/
#endif // #ifndef DCFILTER_H

26
MicroMDAEPiano.ino
Unescape View File

@ -57,8 +57,8 @@ AudioAmplifier inverter;
AudioEffectModulatedDelay modchorus_r;
AudioEffectModulatedDelay modchorus_l;
AudioSynthWaveform modulator;
AudioFilterStateVariable chorus_filter_r;
AudioFilterStateVariable chorus_filter_l;
//AudioFilterStateVariable chorus_filter_r;
//AudioFilterStateVariable chorus_filter_l;
AudioConnection patchCord0(queue_r, peak_r);
AudioConnection patchCord1(queue_l, peak_l);
AudioConnection patchCord2(queue_r, freeverb_r);
@ -70,10 +70,12 @@ AudioConnection patchCord9(modulator, inverter);
AudioConnection patchCord10(inverter, 0, modchorus_l, 1);
AudioConnection patchCord11(queue_r, 0, mixer_r, 0);
AudioConnection patchCord12(queue_l, 0, mixer_l, 0);
AudioConnection patchCord13(modchorus_r, chorus_filter_r);
AudioConnection patchCord14(modchorus_l, chorus_filter_l);
AudioConnection patchCord15(chorus_filter_r, 0, mixer_r, 2);
AudioConnection patchCord16(chorus_filter_l, 0, mixer_l, 2);
//AudioConnection patchCord13(modchorus_r, chorus_filter_r);
//AudioConnection patchCord14(modchorus_l, chorus_filter_l);
//AudioConnection patchCord15(chorus_filter_r, 0, mixer_r, 2);
//AudioConnection patchCord16(chorus_filter_l, 0, mixer_l, 2);
AudioConnection patchCord15(modchorus_r, 0, mixer_r, 2);
AudioConnection patchCord16(modchorus_l, 0, mixer_l, 2);
AudioConnection patchCord17(freeverb_r, 0, mixer_r, 1);
AudioConnection patchCord18(freeverb_l, 0, mixer_l, 1);
AudioConnection patchCord19(mixer_r, volume_r);
@ -238,12 +240,12 @@ void setup()
modulator.amplitude(1.0);
modulator.offset(0.0);
inverter.gain(-1.0); // change phase for second modulated delay (faked stereo mode)
chorus_filter_r.frequency(3500);
chorus_filter_r.resonance(0.7);
chorus_filter_r.octaveControl(1);
chorus_filter_l.frequency(3500);
chorus_filter_l.resonance(0.7);
chorus_filter_l.octaveControl(1);
//chorus_filter_r.frequency(3500);
//chorus_filter_r.resonance(0.7);
//chorus_filter_r.octaveControl(1);
//chorus_filter_l.frequency(3500);
//chorus_filter_l.resonance(0.7);
//chorus_filter_l.octaveControl(1);
modchorus_r.offset(15.0);
modchorus_l.offset(15.0);

20
DCBlock.h → OnePoleLP.h
Unescape View File

@ -1,4 +1,3 @@
// From "git clone --recursive https://github.com/SpotlightKid/ykchorus.git"
/*
==============================================================================
This file is part of Tal-NoiseMaker by Patrick Kunz.
@ -22,24 +21,23 @@
==============================================================================
*/
#if !defined(__DCBlock_h)
#define __DCBlock_h
#if !defined(__OnePoleLP_h)
#define __OnePoleLP_h
class DCBlock {
class OnePoleLP {
public:
float inputs, outputs, lastOutput;
DCBlock() {
OnePoleLP() {
lastOutput = inputs = outputs = 0.0f;
}
~DCBlock() {}
~OnePoleLP() {}
inline void tick(float *sample, float cutoff) {
outputs = *sample - inputs + (0.999f - cutoff * 0.4f) * outputs;
inputs = *sample;
lastOutput = outputs;
*sample = lastOutput;
void tick(float *sample, float cutoff) {
float p = (cutoff * 0.98f) * (cutoff * 0.98f) * (cutoff * 0.98f) * (cutoff * 0.98f);
outputs = (1.0f - p) * (*sample) + p * outputs;
*sample = outputs;
}
};

2
config.h
Unescape View File

@ -56,7 +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_MOD WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
#define CHORUS_WAVEFORM_MOD WAVEFORM_SINE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
//*************************************************************************************************
//* DEBUG OUTPUT SETTINGS
//*************************************************************************************************

55
effect_modulated_delay.cpp
Unescape View File

@ -26,7 +26,8 @@
#include "arm_math.h"
#include "effect_modulated_delay.h"
#include "config.h"
#include "DCBlock.h"
#include "DCfilter.h"
#include "OnePoleLP.h"
#include "limits.h"
extern config_t configuration;
@ -51,6 +52,7 @@ 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);
@ -65,7 +67,8 @@ boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
set_modulator_filter_coeffs();
modulator_filter_data = {1, modulator_filter_state, modulator_filter_coeffs};
dcBlock = new DCBlock();
lp = new OnePoleLP();
dcFilter = new DCfilter(0.9);
return (true);
}
@ -77,12 +80,12 @@ 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 = 400 Hz; Q=0.707
modulator_filter_coeffs[0] = 7.80321719e-4; // b0
modulator_filter_coeffs[1] = 0.00156064; // b1
modulator_filter_coeffs[2] = 7.80321719e-4; // b2
modulator_filter_coeffs[3] = 1.91943335; // -a1
modulator_filter_coeffs[4] = -0.92255463; // -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)
@ -124,33 +127,23 @@ void AudioEffectModulatedDelay::update(void)
// calculate modulation index into circular buffer
cb_mod_index = (_cb_index - (_delay_offset + mod_number));
/* if (cb_mod_index >= _delay_length)
cb_mod_index -= _delay_length; */
if (cb_mod_index < 0) // check for negative offsets and correct them
cb_mod_index += _delay_length;
if (*mp < 0.0)
{
if (cb_mod_index == 0)
cb_mod_index_neighbor = _delay_length;
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));
}
if (cb_mod_index == 0)
cb_mod_index_neighbor = _delay_length;
else
{
if (cb_mod_index == _delay_length)
cb_mod_index_neighbor = 0;
else
cb_mod_index_neighbor = cb_mod_index + 1;
*bp = round(float(_delayline[cb_mod_index_neighbor]) * mod_fraction + float(_delayline[cb_mod_index]) * (1.0 - mod_fraction));
}
// simple Test for DC filter
float bp_f=*bp/SHRT_MAX;
dcBlock->tick(&bp_f, 0.01f);
*bp=bp_f*SHRT_MAX;
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);
// push the pointers forward
bp++; // next audio data
mp++; // next modulation data

7
effect_modulated_delay.h
Unescape View File

@ -27,7 +27,8 @@
#include "Arduino.h"
#include "AudioStream.h"
#include "config.h"
#include "DCBlock.h"
#include "DCfilter.h"
#include "OnePoleLP.h"
#define MODULATION_MAX_FACTOR 0.75
@ -60,7 +61,9 @@ class AudioEffectModulatedDelay :
arm_biquad_casd_df1_inst_f32 modulator_filter_data;
float32_t modulator_filter_state[4];
float32_t modulator_filter_coeffs[5];
DCBlock *dcBlock;
int16_t z1;
OnePoleLP *lp;
DCfilter *dcFilter;
};
#endif

Write Preview
Loading…
Cancel
Save