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hexefx_audiolib_F32/src/effect_phaserStereo_F32.h

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/* Stereo Phaser/Vibrato effect for Teensy Audio library
*
* Author: Piotr Zapart
* www.hexefx.com
*
* Copyright (c) 2021 by Piotr Zapart
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
#ifndef _EFFECT_PHASERSTEREO_F32_H
#define _EFFECT_PHASERSTEREO_F32_H
#include <Arduino.h>
#include "Audio.h"
#include "AudioStream.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
#define PHASER_STEREO_STAGES 12
class AudioEffectPhaserStereo_F32 : public AudioStream_F32
{
public:
AudioEffectPhaserStereo_F32();
~AudioEffectPhaserStereo_F32();
virtual void update();
/**
* @brief Scale and offset the modulation signal. It can be the internal LFO
* or the incomig routed modulation AudioSignal.
* LFO will oscillate between these two max and min values.
*
* @param top top level of the LFO
* @param bottom bottom level of the LFO
*/
void depth(float32_t top, float32_t bottom)
{
float32_t a, b;
lfo_top = constrain(top, 0.0f, 1.0f);
lfo_btm = constrain(bottom, 0.0f, 1.0f);
a = abs(lfo_top - lfo_btm); // scaler
b = min(lfo_top, lfo_btm); // bias
__disable_irq();
lfo_bias = b;
lfo_scaler = a;
__enable_irq();
}
/**
* @brief classic way of setting the depth: LFO centered around 0.5
*
* @param dpth modulation wave amplitude
*/
void depth(float32_t value)
{
value = constrain(value, 0.0f, 1.0f);
value *= 0.5f;
lfo_top = 0.5f + value;
lfo_btm = 0.5f - value;
__disable_irq();
lfo_bias = 0.5f;
lfo_scaler = value;
__enable_irq();
}
void top(float32_t value)
{
lfo_top = constrain(value, 0.0f, 1.0f);
depth(lfo_top, lfo_btm);
}
void btm(float32_t value)
{
lfo_btm = constrain(value, 0.0f, 1.0f);
depth(lfo_top, lfo_btm);
}
/**
* @brief Controls the internal LFO, or if a control signal is used, scales it
* Use this function to update all lfo parameteres at once
*
* @param f_Hz lfo frequency, use 0.0f for manual phaser control
* @param phase phase shift between the LFOs L and R waveforms, 0.0-1.0 range
* @param top lfo top level
* @param btm lfo bottm level
*/
void lfo(float32_t f_Hz, float32_t phase, float32_t top, float32_t btm)
{
float32_t a, b, c;
uint32_t add;
uint8_t bs;
a = constrain(top, 0.0f, 1.0f);
b = constrain(btm, 0.0f, 1.0f);
c = abs(a - b); // scaler
a = min(a, b); // bias
f_Hz = constrain(f_Hz, 0.0f, AUDIO_SAMPLE_RATE_EXACT/2);
phase = constrain(phase, 0.0f, 1.0f);
add = f_Hz * (4294967296.0f / AUDIO_SAMPLE_RATE_EXACT);
bs = (uint8_t)(phase * 128.0f);
__disable_irq();
lfo_scaler = c;
lfo_bias = a;
lfo_add = add;
lfo_lroffset = bs;
__enable_irq();
}
void stereo(float32_t phase)
{
uint8_t bs;
phase = constrain(phase, 0.0f, 1.0f);
bs = (uint8_t)(phase * 128.0f);
__disable_irq();
lfo_lroffset = bs;
__enable_irq();
}
/**
* @brief Set the rate of the internal LFO
*
* @param f_Hz lfo frequency, use 0.0f for manual phaser control
*/
void lfo_rate(float32_t f_Hz)
{
float32_t c;
uint32_t add;
c = constrain(f_Hz, 0.0f, AUDIO_SAMPLE_RATE_EXACT/2);
add = c * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
__disable_irq();
lfo_add = add;
__enable_irq();
}
/**
* @brief Controls the feedback parameter
*
* @param fdb ffedback value in range 0.0f to 1.0f
*/
void feedback(float32_t fdb)
{
feedb = constrain(fdb, -1.0f, 1.0f);
}
/**
* @brief Dry / Wet mixer ratio. Classic Phaser sound uses 0.5f for 50% dry and 50%Wet
* 1.0f will produce 100% wet signal craeting a vibrato effect
*
* @param ratio mixing ratio, range 0.0f (full dry) to 1.0f (full wet)
*/
void mix(float32_t ratio)
{
mix_ratio = constrain(ratio, 0.0f, 1.0f);
}
/**
* @brief Sets the number of stages used in the phaser
* Allowed values are: 2, 4, 6, 8, 10, 12
*
* @param st number of stages, even value <= 12
*/
void stages(uint8_t st)
{
if (st && st == ((st >> 1) << 1) && st <= PHASER_STEREO_STAGES) // only 2, 4, 6, 8, 12 allowed
{
stg = st;
}
}
/**
* @brief Use to bypass the effect (true)
*
* @param state true = bypass on, false = phaser on
*/
void bypass_set(bool state) {bps = state;}
bool bypass_tgl(void) {bps ^= 1; return bps;}
private:
uint8_t stg; // number of stages
bool bps; // bypass
audio_block_f32_t *inputQueueArray_f32[3];
float32_t allpass_x[2][PHASER_STEREO_STAGES]; // allpass inputs
float32_t allpass_y[2][PHASER_STEREO_STAGES]; // allpass outputs
float32_t mix_ratio; // 0 = dry. 1.0 = wet
float32_t feedb; // feedback
float32_t last_sampleL;
float32_t last_sampleR;
uint32_t lfo_phase_acc; // interfnal lfo
uint32_t lfo_add;
float32_t lfo_lrphase;
uint32_t lfo_lroffset;
float32_t lfo_scaler;
float32_t lfo_bias;
float32_t lfo_top;
float32_t lfo_btm;
};
#endif // _EFFECT_PHASER_H