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hexefx_audiolib_F32/src/effect_delaystereo_F32.h

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/* Stereo Ping Pong delay for Teensy 4
*
* Author: Piotr Zapart
* www.hexefx.com
*
* Copyright (c) 2024 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_DELAYSTEREO_H_
#define _EFFECT_DELAYSTEREO_H_
#include <Arduino.h>
#include "Audio.h"
#include "AudioStream.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
#include "basic_components.h"
class AudioEffectDelayStereo_F32 : public AudioStream_F32
{
public:
AudioEffectDelayStereo_F32(uint32_t dly_range_ms=400, bool use_psram=false);
~AudioEffectDelayStereo_F32(){};
virtual void update();
/**
* @brief set the delay time
*
* @param t delay time scaled to range 0.0 to 1.0
* @param force bypass the smoothing, immediate change
*/
void time(float t, bool force = false)
{
t = constrain(t, 0.0f, 1.0f);
t = t * t;
t = map(t, 0.0f, 1.0f, (float32_t)(dly_length-dly_time_min), 0.0f);
__disable_irq();
if (force) dly_time = t;
dly_time_set = t;
__enable_irq();
}
/**
* @brief delay time set in samples
*
* @param samples range is from 0 to delay buffer lengts - minimum delay
*/
void delay(uint32_t samples)
{
samples = constrain(samples, 0u, dly_length-dly_time_min);
samples = dly_length-dly_time_min - samples;
__disable_irq();
dly_time_set = samples;
__enable_irq();
}
/**
* @brief Amount of repeats
*
* @param n 0.0f-1.0f range
*/
void feedback(float n)
{
if (infinite) return;
float32_t fb, attn;
n = constrain(n, 0.0f, 1.0f);
fb = map(n, 0.0f, 1.0f, 0.0f, feedb_max) * hp_feedb_limit;
attn = map(n*n*n, 0.0f, 1.0f, 1.0f, 0.4f);
inputGain_tmp = attn;
__disable_irq();
feedb = fb;
inputGainSet = attn;
__enable_irq();
}
/**
* @brief How fast the delay time is updated
* emulates analog tape machines
*
* @param n 0.0f-1.0f range, 0 - fastest update
*/
void inertia(float n)
{
n = constrain(n, 0.0f, 1.0f);
n = 2.0f * n - (n*n);
n = map (n, 0.0f, 1.0f, 10.0f, 0.3f);
__disable_irq();
dly_time_step = n;
__enable_irq();
}
/**
* @brief Output treble control
*
* @param n 0.0f-1.0f range
*/
void treble(float n)
{
n = constrain(n, 0.0f, 1.0f);
__disable_irq();
treble_k = n;
__enable_irq();
}
/**
* @brief Treble loss control (darkens the repeats)
*
* @param n 0.0f-1.0f range
*/
void treble_cut(float n)
{
if (infinite) return;
n = 1.0f - constrain(n, 0.0f, 1.0f);
trebleCut_k_tmp = n;
__disable_irq();
trebleCut_k = n;
__enable_irq();
}
/**
* @brief Output bass control
*
* @param n 0.0f-1.0f range
*/
void bass(float n)
{
n = constrain(n, 0.0f, 1.0f);
n = 1.0f - 2.0f*n + (n*n);
__disable_irq();
bass_k = -n;
__enable_irq();
}
/**
* @brief Bass loss (repeats will loose low end)
*
* @param n 0.0f-1.0f range
*/
void bass_cut(float n)
{
if (infinite) return;
n = constrain(n, 0.0f, 1.0f);
n = 2.0f * n - (n*n);
bassCut_k_tmp = -n;
__disable_irq();
bassCut_k = -n;
__enable_irq();
}
/**
* @brief dry/wet mixer
* 0 = dry only, 1=wet only
*
* @param m 0.0f-1-0f range
*/
void mix(float m)
{
float32_t dry, wet;
m = constrain(m, 0.0f, 1.0f);
mix_pwr(m, &wet, &dry);
__disable_irq();
wet_gain = wet;
dry_gain = dry;
__enable_irq();
}
/**
* @brief Modulation frequency in Hz
*
* @param f range 0.0f - 16.0f
*/
void mod_rateHz(float32_t f)
{
f = constrain(f, 0.0f, 16.0f);
__disable_irq();
lfo.setRate(f);
__enable_irq();
}
/**
* @brief modulation frequency scaled to 0.0f-1.0f range
*
* @param r rate
*/
void mod_rate(float32_t r)
{
r = constrain(r*r*r, 0.0f, 1.0f);
r = map(r, 0.0f, 1.0f, 0.0f, lfo_fmax);
__disable_irq();
lfo.setRate(r);
__enable_irq();
}
/**
* @brief Modulation depth
*
* @param d 0.0f-1-0f range
*/
void mod_depth(float32_t d)
{
d = constrain(d, 0.0f, 1.0f);
d = map(d, 0.0f, 1.0f, 0.0f, lfo_ampl_max);
__disable_irq();
lfo.setDepth(d);
__enable_irq();
}
typedef enum
{
BYPASS_MODE_PASS, // pass the input signal to the output
BYPASS_MODE_OFF, // mute the output
BYPASS_MODE_TRAILS // mutes the input only
}bypass_mode_t;
void bypass_setMode(bypass_mode_t m)
{
if (m <= BYPASS_MODE_TRAILS) bp_mode = m;
}
bypass_mode_t bypass_geMode() {return bp_mode;}
bool bypass_get(void) {return bp;}
void bypass_set(bool state)
{
if (bp == state) return;
bp = state;
if (bp)
{
__disable_irq();
memCleanupStart = 0;
memCleanupEnd = memCleanupStep;
__enable_irq();
freeze(false);
}
else
{
__disable_irq();
inputGainSet = inputGain_tmp;
__enable_irq();
}
}
bool bypass_tgl(void)
{
bypass_set(bp ^ 1);
return bp;
}
void freeze(bool state);
bool freeze_tgl() {freeze(infinite^1); return infinite;}
bool freeze_get() {return infinite;}
uint32_t tap_tempo(bool avg=true)
{
int32_t delta;
uint32_t tempo_ticks = 0;
if (!tap_active)
{
tap_counter = 0;
tap_active = true;
}
else
{
__disable_irq();
tap_counter_new = tap_counter;
tap_counter = 0;
__enable_irq();
delta = tap_counter_new - tap_counter_last;
if (abs(delta) > tap_counter_deltamax || !avg) // new tempo?
{
tempo_ticks = tap_counter_new;
}
else
{
tempo_ticks = (tap_counter_new>>1) + (tap_counter_last>>1);
}
while (tempo_ticks > dly_length - dly_time_min)
{
tempo_ticks >>= 1;
}
tap_counter_last = tempo_ticks;
delay(tempo_ticks);
}
return tempo_ticks;
}
bool is_initialized() {return initialized;}
private:
audio_block_f32_t *inputQueueArray[2];
uint32_t dly_length;
AudioBasicDelay dly0a;
AudioBasicDelay dly0b;
AudioBasicDelay dly1a;
AudioBasicDelay dly1b;
AudioFilterShelvingLPHP flt0L;
AudioFilterShelvingLPHP flt1L;
AudioFilterShelvingLPHP flt0R;
AudioFilterShelvingLPHP flt1R;
static constexpr float32_t lfo_fmax = 16.0f;
static constexpr float32_t lfo_ampl_max = 127.0f;
float32_t lfo_ampl = 0.0f;
AudioBasicLfo lfo = AudioBasicLfo(0.0f, lfo_ampl);
bool psram_mode;
bool memsetup_done = false;
bool bp = true;
bypass_mode_t bp_mode = BYPASS_MODE_TRAILS;
bool cleanup_done = false;
bool infinite = false;
bool extInputMode = false; // external input via pointers passed to constructor
static constexpr float32_t feedb_max = 0.96f;
float32_t feedb = 0;
float32_t hp_feedb_limit = 1.0f;
float32_t wet_gain;
float32_t dry_gain;
float32_t inputGainSet = 1.0f;
float32_t inputGain = 1.0f;
float32_t trebleCut_k = 1.0f;
float32_t bassCut_k = 0.0f;
float32_t treble_k = 1.0f;
float32_t bass_k = 0.0f;
float32_t dly_time, dly_time_set;
float32_t dly_time_step = 10.0f;
static const uint32_t dly_time_min = 128;
bool initialized = false;
// freeze variables
float32_t freeze_ingain = 0.00f;
float32_t inputGain_tmp = 1.0f;
float32_t bassCut_k_tmp = 0.0f;
float32_t trebleCut_k_tmp = 1.0f;
float32_t feedb_tmp = 0;
bool tap_active = false;
uint32_t tap_counter = 0;
uint32_t tap_counter_last=0, tap_counter_new=0;
static const uint32_t tap_counter_max = 3000*AUDIO_SAMPLE_RATE; // 3 sec
static const int32_t tap_counter_deltamax = 0.3f*AUDIO_SAMPLE_RATE_EXACT;
bool memCleanup(void);
void begin(uint32_t dly_range_ms, bool use_psram);
const uint32_t memCleanupStep = 2048;
uint32_t memCleanupStart = 0;
uint32_t memCleanupEnd = memCleanupStep;
};
#endif // _EFFECT_DELAYSTEREO_H_