add TGX4 components

README.md

@@ -27,6 +27,12 @@ Versatile stereo ping-pong delay with modulation.
 **AudioEffectNoiseGateStereo_F32**  
 Stereo noise gate with external SideChain input.  
 
+**AudioEffectGuitarBooster_F32**  
+Overdrive emulation using oversampled wave shaper, switchable octave up.  
+
+**AudioEffectWahMono_F32**  
+WAH pedal emulation including 8 models and versatile range handling.  
+
 **AudioFilterToneStackStereo_F32**  
 Stereo guitar tone stack (EQ) emulator.  
 
@@ -43,12 +49,20 @@ Simple 3 band (Treble, Mid, Bass) equalizer.
 **AudioEffectGainStereo_F32**  
 Stereo gain control (volume etc.)  
 
+**AudioSwitchSelectorStereo**  
+Stereo/mono signal selector.  
+
 
 ## I/O  
 **AudioInputI2S2_F32**  
 **AudioOutputI2S2_F32**  
 Input and output for the I2S2 interface, Teensy 4.1 only.  
 
+**AudioInputI2S_ext_F32**  
+**AudioOutputI2S_ext_F32**  
+Custom input and output for the I2S interface, including a few extra options (ie. channel swap)  
+
+
 ## Basic  
 Single header basic building blocks for various DSP components:  
 - allpass filter  
@@ -60,8 +74,9 @@ Single header basic building blocks for various DSP components:
 
 ## Example projects  
 https://github.com/hexeguitar/hexefx_audiolib_F32_examples
+https://github.com/hexeguitar/tgx4
 
 ---  
-Copyright 01.2024 by Piotr Zapart  
+Copyright 07.2024 by Piotr Zapart  
 www.hexefx.com
 

library.properties

@@ -1,5 +1,5 @@
 name=hexefx_audiolib_F32
-version=1.0.0
+version=1.1.0
 author=Piotr Zapart
 maintainer=Piotr Zapart <www.hexefx.com>
 sentence=Audio effect extension for the OpenAudio_ArduinoLibrary

src/basic_shelvFilter.h

@@ -29,6 +29,7 @@ public:
 	}
 	inline float process(float input)
 	{
+		if (bp) return input;
 		float tmp1, tmp2;
 		// smoothly update params
 		if (hidamp < (*hidampPtr))
@@ -64,6 +65,7 @@ public:
 		lpreg = 0.0f;
 		hpreg = 0.0f;
 	}
+	void bypass_set(bool state) { bp = state; reset();}
 private:
 	float lpreg;
 	float hpreg;
@@ -74,6 +76,7 @@ private:
 	float hp_f;
 	float lp_f;
 	static constexpr float upd_step = 0.02f;
+	bool bp = false;
 };
 
 

src/control_AK4452_F32.cpp

@@ -0,0 +1,204 @@
+/**
+ * @file control_AK4452_F32.cpp
+ * @author Piotr Zapart
+ * @brief driver for the AK4452 DAC
+ * @version 0.1
+ * @date 2024-06-14
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+#include "control_AK4452_F32.h"
+
+#define AK4452_REG_CTRL1			(0x00)
+#define AK4452_REG_CTRL1_DFLT		(0x0C)
+	#define AK4452_BIT_RSTN			(1<<0)
+	#define AK4452_DIF_MASK			(0x0E)
+	#define AK4452_DIF_SHIFT		(0x01)
+	#define AK4452_DIF(x)			(((x)<<AK4452_DIF_SHIFT)&AK4452_DIF_MASK)
+		#define AK4452_DIF_16B_LSB	(0x00)
+		#define AK4452_DIF_20B_LSB	(0x01)
+		#define AK4452_DIF_24B_MSB	(0x02)
+		#define AK4452_DIF_24B_I2S	(0x03)
+		#define AK4452_DIF_24B_LSB	(0x04)
+		#define AK4452_DIF_32B_LSB	(0x05)
+		#define AK4452_DIF_32B_MSB	(0x06)
+		#define AK4452_DIF_32B_I2S	(0x07)
+	#define AK4452_BIT_ACKS			(1<<7)
+
+#define AK4452_REG_CTRL2			(0x01)
+#define AK4452_REG_CTRL2_DFLT		(0x22)
+	#define AK4452_BIT_SMUTE		(1<<0)
+	#define AK4452_DEM_MASK			(0x06)
+	#define AK4452_DEM_SHIFT		(0x01)
+	#define AK4452_DEM(x)			(((x)<<AK4452_DEM_SHIFT)&AK4452_DEM_MASK)
+		#define AK4452_DEM_44_1K	(0x00)
+		#define AK4452_DEM_OFF		(0x01)
+		#define AK4452_DEM_48K		(0x02)
+		#define AK4452_DEM_32K		(0x03)
+	#define AK4452_DFS10_MASK		(0x18)
+	#define AK4452_DFS10_SHIFT		(0x03)
+	#define AK4452_DFS10(x)			(((x)<<AK4452_DFS10_SHIFT)&AK4452_DFS10_MASK)
+		#define AK4452_DFS_NORMAL	(0x00)
+		#define AK4452_DFS_DOUBLE	(0x01)
+		#define AK4452_DFS_QUAD		(0x02)
+		#define AK4452_DFS_OCT		(0x04)
+		#define AK4452_DFS_HEX		(0x05)
+	#define AK4452_BIT_SD			(1<<5)	
+
+#define AK4452_REG_CTRL3			(0x02)
+#define AK4452_REG_CTRL3_DFLT		(0x00)	
+	#define AK4452_BIT_SLOW			(1<<0)
+	#define AK4452_BIT_SELLR1		(1<<1)
+	#define AK4452_BIT_DZFB			(1<<2)
+	#define AK4452_BIT_MONO1		(1<<3)
+	#define AK4452_BIT_DCKB			(1<<4)
+	#define AK4452_BIT_DCKS			(1<<5)
+	#define AK4452_BIT_DP			(1<<7)
+
+#define AK4452_REG_L1CH_ATT			(0x03)
+#define AK4452_REG_L1CH_ATT_DFLT	(0xFF)
+#define AK4452_REG_R1CH_ATT			(0x04)
+#define AK4452_REG_R1CH_ATT_DFLT	(0xFF)
+
+#define AK4452_REG_CTRL4			(0x05)
+#define AK4452_REG_CTRL4_DFLT		(0x00)
+	#define AK4452_BIT_SSLOW		(1<<0)
+	#define AK4452_DFS2_MASK		(1<<1)
+	#define AK4452_DFS2_SHIFT		(0x01)
+	#define AK4452_DFS2(x)			(((x)<<AK4452_DFS2_SHIFT)&AK4452_DFS2_MASK) // bits10 in CTRL2
+	#define AK4452_BIT_INVL1		(1<<6)
+	#define AK4452_BIT_INVR1		(1<<7)
+
+#define AK4452_REG_DSD1				(0x06)
+#define AK4452_REG_DSD1_DFLT		(0x00)
+	#define AK4452_BIT_DSDSEL0		(1<<0)
+	#define AK4452_BIT_DSDD			(1<<1)
+	#define AK4452_BIT_DMRE			(1<<3)
+	#define AK4452_BIT_DMC			(1<<4)
+	#define AK4452_BIT_DMR1			(1<<5)
+	#define AK4452_BIT_DML1			(1<<6)
+	#define AK4452_BIT_DDM			(1<<7)
+
+#define AK4452_REG_CTRL5			(0x07)
+#define AK4452_REG_CTRL5_DFLT		(0x03)
+	#define AK4452_BIT_SYNCE		(1<<0)
+
+#define AK4452_REG_SND_CTRL			(0x08)
+#define AK4452_REG_SND_CTRL_DFLT	(0x00)
+	#define AK4452_SC_MASK			(0x03)
+	#define AK4452_SC_SHIFT			(0x00)
+	#define AK4452_SC(x)			(((x)<<AK4452_SC_SHIFT)&AK4452_SC_MASK) 
+		#define AK4452_SC_NORMAL	(0x00)
+		#define AK4452_SC_MAX		(0x01)
+		#define AK4452_SC_MIN		(0x02)
+	#define AK4452_BIT_DZF_L1		(1<<6)
+	#define AK4452_BIT_DZF_R1		(1<<7)
+
+#define AK4452_REG_DSD2				(0x09)
+#define AK4452_REG_DSD2_DFLT		(0x00)
+	#define AK4452_BIT_DSDSEL1		(1<<0)
+	#define AK4452_BIT_DSDF			(1<<1)
+
+#define AK4452_REG_CTRL6			(0x0A)
+#define AK4452_REG_CTRL6_DFLT		(0x0D)
+	#define AK4452_BIT_PW1			(1<<2)
+	#define AK4452_SDS21_MASK		(0x30)
+	#define AK4452_SDS21_SHIFT		(0x04)
+	#define AK4452_SDS21(x)			(((x)<<AK4452_SDS21_SHIFT)&AK4452_SDS21_MASK) 
+		#define AK4452_SDS_TDM128_L1R1	(0x00)
+		#define AK4452_SDS_TDM128_L2R2	(0x01)
+		#define AK4452_SDS_TDM256_L1R1	(0x00)
+		#define AK4452_SDS_TDM256_L2R2	(0x01)
+		#define AK4452_SDS_TDM256_L3R3	(0x02)
+		#define AK4452_SDS_TDM256_L4R4	(0x03)
+		#define AK4452_SDS_TDM512_L1R1	(0x00)
+		#define AK4452_SDS_TDM512_L2R2	(0x01)
+		#define AK4452_SDS_TDM512_L3R3	(0x02)
+		#define AK4452_SDS_TDM512_L4R4	(0x03)
+		#define AK4452_SDS_TDM512_L5R5	(0x04)
+		#define AK4452_SDS_TDM512_L6R6	(0x05)
+		#define AK4452_SDS_TDM512_L7R7	(0x06)
+		#define AK4452_SDS_TDM512_L8R8	(0x07)
+	#define AK4452_TDM_MASK			(0xC0)
+	#define AK4452_TDM_SHIFT		(0x06)
+	#define AK4452_TDM(x)			(((x)<<AK4452_TDM_SHIFT)&AK4452_TDM_MASK) 
+		#define AK4452_TDM_NORMAL	(0x00)
+		#define AK4452_TDM_128		(0x01)
+		#define AK4452_TDM_246		(0x02)
+		#define AK4452_TDM_512		(0x03)
+
+#define AK4452_REG_CTRL7			(0x0B)
+#define AK4452_REG_CTRL7_DFLT		(0x0C)
+	#define AK4452_BIT_DCHAIN		(1<<2)
+	#define AK4452_SDS0_MASK		(1<<4)
+	#define AK4452_SDS0_SHIFT		(0x04)
+	#define AK4452_SDS0(x)			(((x)<<AK4452_SDS0_SHIFT)&AK4452_SDS0_MASK) 
+	#define AK4452_ATS_MASK			(0xC0)
+	#define AK4452_ATS_SHIFT		(0x06)
+	#define AK4452_ATS(x)			(((x)<<AK4452_ATS_SHIFT)&AK4452_ATS_MASK)
+		#define AK4452_ATS_4080		(0x00)
+		#define AK4452_ATS_2040		(0x01)
+		#define AK4452_ATS_510		(0x02)
+		#define AK4452_ATS_255		(0x03)
+
+#define AK4452_REG_CTRL8			(0x0C)
+#define AK4452_REG_CTRL8_DFLT		(0x00)
+	#define AK4452_FIR(x)			((x)&0x07)
+
+bool AudioControlAK4452_F32::configured = false;
+
+bool AudioControlAK4452_F32::enable(TwoWire *i2cBus, uint8_t addr)
+{
+	ctrlBus = i2cBus;
+	i2cAddr = addr;
+	ctrlBus->begin();
+	ctrlBus->setClock(400000);
+
+	if (!writeReg(AK4452_REG_CTRL1, 0x00))	// put the registers in reset mode
+	{
+		return false; // codec not found
+	}	
+	// DIF[2:0] = 0b111 - 32bit I2S, Normal mode 
+	// DFS[2:0] = 0b000 (default) Normal speed mode
+	// DSDSEL[1:0] = 0b10 256fs
+	writeReg(AK4452_REG_DSD2, AK4452_BIT_DSDSEL1);
+	writeReg(AK4452_REG_CTRL1, AK4452_DIF(AK4452_DIF_32B_I2S) | AK4452_BIT_RSTN);
+
+	configured = true;
+	return true;
+}
+
+bool AudioControlAK4452_F32::writeReg(uint8_t addr, uint8_t val)
+{
+	ctrlBus->beginTransmission(i2cAddr);
+	ctrlBus->write(addr);
+	ctrlBus->write(val);
+	return ctrlBus->endTransmission() == 0;
+}
+bool AudioControlAK4452_F32::readReg(uint8_t addr, uint8_t *valPtr)
+{
+	ctrlBus->beginTransmission(i2cAddr);
+	ctrlBus->write(addr);
+	if (ctrlBus->endTransmission(false) != 0)
+		return false;	
+	if (ctrlBus->requestFrom((int)i2cAddr, 1) < 1)  return false;
+	*valPtr = ctrlBus->read();
+	return true;
+}
+
+uint8_t AudioControlAK4452_F32::modifyReg(uint8_t reg, uint8_t val, uint8_t iMask)
+{
+	uint8_t val1;
+	val1 = (readReg(reg, &val1) & (~iMask)) | val;
+	if (!writeReg(reg, val1))
+		return 0;
+	return val1;
+}

src/control_AK4452_F32.h

@@ -0,0 +1,56 @@
+/**
+ * @file control_AK4452_F32.h
+ * @author Piotr Zapart
+ * @brief driver for the AK4452 DAC
+ * @version 0.1
+ * @date 2024-06-14
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+#ifndef _CONTROL_AK4452_H_
+#define _CONTROL_AK4452_H_
+#include <Arduino.h>
+#include <Wire.h>
+#include "AudioControl.h"
+
+#define AK4452_ADDR00	(0x10)
+#define AK4452_ADDR01	(0x11)
+#define AK4452_ADDR10	(0x12)
+#define AK4452_ADDR11	(0x13)
+
+class AudioControlAK4452_F32 : public AudioControl
+{
+public:
+	AudioControlAK4452_F32(){};
+	~AudioControlAK4452_F32(){};
+	bool enable()
+	{
+		return enable(&Wire, AK4452_ADDR10);
+	}
+	bool enable(TwoWire *i2cBus, uint8_t addr);
+	
+	// not used but required by AudioControl
+	bool disable() {return true;}
+	bool volume(float volume)  {return true;};
+	bool inputLevel(float volume) {return true;}
+	bool inputSelect(int n) {return true;}
+
+private:
+	static bool configured;
+	TwoWire *ctrlBus;
+	uint8_t i2cAddr;
+
+	bool writeReg(uint8_t addr, uint8_t val);
+	bool readReg(uint8_t addr, uint8_t* valPtr);
+	uint8_t modifyReg(uint8_t reg, uint8_t val, uint8_t iMask);
+};
+
+#endif // _CONTROL_AK4452_H_

src/control_AK5552_F32.cpp

@@ -0,0 +1,114 @@
+/**
+ * @file control_AK5552_F32.cpp
+ * @author Piotr Zapart
+ * @brief driver for the AK5552 ADC
+ * @version 0.1
+ * @date 2024-06-14
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+#include "control_AK5552_F32.h"
+
+#define AK5552_REG_PWR_MAN1			(0x00)
+#define AK5552_REG_PWR_MAN1_DFLT	(0xFF)
+	#define AK5552_BIT_PW1			(1<<0)
+	#define AK5552_BIT_PW2			(1<<1)
+
+#define AK5552_REG_PWR_MAN2			(0x01)
+#define AK5552_REG_PWR_MAN2_DFLT	(0x01)
+	#define AK5552_BIT_RSTN			(1<<0)
+	#define AK5552_BIT_MONO1		(1<<1)
+	#define AK5552_BIT_MONO2		(1<<2)
+
+#define AK5552_REG_PWR_CTRL1		(0x02)
+#define AK5552_REG_PWR_CTRL1_DFLT	(0x01)
+	#define AK5552_BIT_HPFE			(1<<0)
+	#define AK5552_BIT_DIF0			(1<<1)
+	#define AK5552_BIT_DIF1			(1<<2)
+	#define AK5552_BIT_CKS0			(1<<3)
+	#define AK5552_BIT_CKS1			(1<<4)
+	#define AK5552_BIT_CKS2			(1<<5)
+	#define AK5552_BIT_CKS3			(1<<6)
+
+#define AK5552_REG_PWR_CTRL2		(0x03)
+#define AK5552_REG_PWR_CTRL2_DFLT	(0x00)
+	#define AK5552_BIT_TDM0			(1<<5)
+	#define AK5552_BIT_TDM1			(1<<6)
+
+#define AK5552_REG_PWR_CTRL3		(0x04)
+#define AK5552_REG_PWR_CTRL3_DFLT	(0x00)
+	#define AK5552_BIT_SLOW			(1<<0)
+	#define AK5552_BIT_SD			(1<<1)
+
+#define AK5552_REG_PWR_DSD			(0x04)
+#define AK5552_REG_PWR_DSD_DFLT		(0x00)
+	#define AK5552_DSDSEL_64FS		(0x00)
+	#define AK5552_DSDSEL_128FS		(0x01)
+	#define AK5552_DSDSEL_256FS		(0x10)
+	#define AK5552_DSDSEL(x)		((x)&0x03)
+	#define AK5552_BIT_DCKB			(1<<2)
+	#define AK5552_BIT_PMOD			(1<<3)
+	#define AK5552_BIT_DCKS			(1<<5)
+
+bool AudioControlAK5552_F32::configured = false;
+
+
+bool AudioControlAK5552_F32::enable(TwoWire *i2cBus, uint8_t addr)
+{
+	ctrlBus = i2cBus;
+	i2cAddr = addr;
+	ctrlBus->begin();
+	ctrlBus->setClock(400000);
+
+	if (!writeReg(AK5552_REG_PWR_MAN2, 0x00))	// put the registers in reset mode
+	{
+		return false; // codec not found
+	}	
+	// Normal Speed 256fs, table 5, page 34
+	// CKS3=0, CKS2=0, CKS1=1, CKS0=0
+	// 32bit I2S, Slave mode, table 8, page 62
+	// TDM1=0, TDM0=0, MSN=0, DIF1=1, DIF0=1 
+	writeReg(	AK5552_REG_PWR_CTRL1, 	AK5552_BIT_CKS1 | 
+										AK5552_BIT_DIF1 | 
+										AK5552_BIT_DIF0);
+	// enable short delay
+	writeReg(AK5552_REG_PWR_CTRL3, AK5552_BIT_SD);
+	writeReg(AK5552_REG_PWR_MAN2, AK5552_BIT_RSTN); // register in normal operation
+	configured = true;
+	return true;
+}
+
+bool AudioControlAK5552_F32::writeReg(uint8_t addr, uint8_t val)
+{
+	ctrlBus->beginTransmission(i2cAddr);
+	ctrlBus->write(addr);
+	ctrlBus->write(val);
+	return ctrlBus->endTransmission() == 0;
+}
+bool AudioControlAK5552_F32::readReg(uint8_t addr, uint8_t *valPtr)
+{
+	ctrlBus->beginTransmission(i2cAddr);
+	ctrlBus->write(addr);
+	if (ctrlBus->endTransmission(false) != 0)
+		return false;	
+	if (ctrlBus->requestFrom((int)i2cAddr, 1) < 1)  return false;
+	*valPtr = ctrlBus->read();
+	return true;
+}
+
+uint8_t AudioControlAK5552_F32::modifyReg(uint8_t reg, uint8_t val, uint8_t iMask)
+{
+	uint8_t val1;
+	val1 = (readReg(reg, &val1) & (~iMask)) | val;
+	if (!writeReg(reg, val1))
+		return 0;
+	return val1;
+}

src/control_AK5552_F32.h

@@ -0,0 +1,58 @@
+/**
+ * @file control_AK5552_F32.h
+ * @author Piotr Zapart
+ * @brief driver for the AK5552 ADC
+ * @version 0.1
+ * @date 2024-06-14
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+#ifndef _CONTROL_AK5552_F32_H_
+#define _CONTROL_AK5552_F32_H_
+
+#include <Arduino.h>
+#include <Wire.h>
+#include "AudioControl.h"
+
+#define AK5552_ADDR00	(0x10)
+#define AK5552_ADDR01	(0x11)
+#define AK5552_ADDR10	(0x12)
+#define AK5552_ADDR11	(0x13)
+
+class AudioControlAK5552_F32 : public AudioControl
+{
+public:
+	AudioControlAK5552_F32(){};
+	~AudioControlAK5552_F32(){};
+	bool enable()
+	{
+		return enable(&Wire, AK5552_ADDR00);
+	}
+	bool enable(TwoWire *i2cBus, uint8_t addr);
+
+	// not used but required by AudioControl
+	bool disable() {return true;}
+	bool volume(float volume)  {return true;};
+	bool inputLevel(float volume) {return true;}
+	bool inputSelect(int n) {return true;}
+
+private:
+	static bool configured;
+	TwoWire *ctrlBus;
+	uint8_t i2cAddr;
+
+	bool writeReg(uint8_t addr, uint8_t val);
+	bool readReg(uint8_t addr, uint8_t* valPtr);
+	uint8_t modifyReg(uint8_t reg, uint8_t val, uint8_t iMask);
+};
+
+
+#endif // _CONTROL_AK5552_H_

src/control_ES8388_F32.cpp

@@ -1,7 +1,5 @@
 #include "control_ES8388_F32.h"
 
-
-
 #define ES8388_REG_CHIP_CTRL1		(0x00)	// Default 0000 0110
 #define ES8388_REG_CHIP_CTRL1_DFLT	(0x06)
 	#define ES8388_BIT_SCPRESET		(1<<7) 	// 1=reset registers to default
@@ -476,19 +474,6 @@ bool AudioControlES8388_F32::enable(TwoWire *i2cBus, uint8_t addr, config_t cfg)
 	optimizeConversion(0);
 	writeReg(ES8388_REG_CHIP_PWR_MAN, 0x00); // Power up DEM and STM
 
-	// ALC config  - disabled for now, will be tested someday.. 
-	// writeReg(ES8388_REG_ADC_CTRL10, ES8388_ALCSEL(ES8388_ALCSEL_LR) |			// ALC OFF
-	// 									ES8388_MAXGAIN(ES8388_MAXGAIN_M0_5DB) | // max gain -0.5dB
-	// 									ES8388_MINGAIN(ES8388_MINGAIN_M12DB));	// min gain -12dB
-	// writeReg(ES8388_REG_ADC_CTRL11, ES8388_ALCLVL(0x0A));						// target gain -1.5dB, hold time=0
-	// writeReg(ES8388_REG_ADC_CTRL12, ES8388_ALCATK(0x02) |						// ALC limiter attack time 90.8us
-	// 									ES8388_ALCDCY(0x01));					// ALC limiter decay time 182us
-	// writeReg(ES8388_REG_ADC_CTRL13, ES8388_BIT_ALCMODE | ES8388_WINSIZE(0x06));	// Limiter mode, no ZC, 96*16 samples peak window
-	// writeReg(ES8388_REG_ADC_CTRL14, 0x00);										// disable noise gate
-	//writeReg(ES8388_REG_ADC_CTRL14, ES8388_NGTH(0x1F) | ES8388_NGG(ES8388_NGG_ADCMUTE)| ES8388_BIT_NGAT_EN);
-	// ADC PGA gain
-	//writeReg(ES8388_REG_ADC_CTRL1, 0x00);		
-
 	configured = true;
 	return true;
 }
@@ -555,7 +540,7 @@ void AudioControlES8388_F32::set_noiseGate(float thres)
 	writeReg(ES8388_REG_ADC_CTRL14, ES8388_NGTH(thres_val) | ES8388_NGG(ES8388_NGG_ADCMUTE)| ES8388_BIT_NGAT_EN);	
 }
 
-// bypassed the analog input to the output, disconnect the digital i / o 
+
 bool AudioControlES8388_F32::analogBypass(bool bypass)
 {
 	bool res = true;
@@ -572,16 +557,15 @@ bool AudioControlES8388_F32::analogBypass(bool bypass)
 	return res;
 }
 
-// bypassed the analog input to the output, disconnect the digital input, preserve the digital output connection
 bool AudioControlES8388_F32::analogSoftBypass(bool bypass)
 {
 	bool res = true;
 	if (bypass)
 	{
-		res &= writeReg(ES8388_REG_DAC_CTRL17, 	ES8388_BIT_LI2LO | 					// Lin in on
+		res &= writeReg(ES8388_REG_DAC_CTRL17, 	ES8388_BIT_LI2LO | 					// Lin on
 												ES8388_BIT_LD2LO |					// L Dac on
 												ES8388_LI2LOVOL(ES8388_VOL_0DB)); 	// Lin gain 0dB
-		res &= writeReg(ES8388_REG_DAC_CTRL20,  ES8388_BIT_RI2RO | 					// Rin in on
+		res &= writeReg(ES8388_REG_DAC_CTRL20,  ES8388_BIT_RI2RO | 					// Rin on
 												ES8388_BIT_RD2RO |					// R Dac on
 												ES8388_RI2ROVOL(ES8388_VOL_0DB)); 	// Rin gain 0dB
 	}

src/control_ES8388_F32.h

@@ -1,3 +1,20 @@
+/**
+ * @file control_ES8388_F32.h
+ * @author your name (you@domain.com)
+ * @brief driver for the ES8388 codec chip
+ * @version 0.1
+ * @date 2024-06-14
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
 #ifndef _CONTROL_ES8388_F32_H_
 #define _CONTROL_ES8388_F32_H_
 
@@ -8,7 +25,7 @@
 #define ES8388_I2C_ADDR_L	(0x10) 		// CS/ADD pin low
 #define ES8388_I2C_ADDR_H	(0x11) 		// CS/ADD pin high
 
-class AudioControlES8388_F32 //: public AudioControl
+class AudioControlES8388_F32 : public AudioControl
 {
 public:
 	AudioControlES8388_F32(void){};
@@ -27,8 +44,8 @@ public:
 	bool enable(TwoWire *i2cBus, uint8_t addr,  config_t cfg);
 	bool disable(void) { return false; }
 	bool volume(float n);
-	bool inputLevel(float n); // range: 0.0f to 1.0f
+	bool inputLevel(float n) {return true;} // range: 0.0f to 1.0f
-
+	bool inputSelect(int n) {return true;}
 
 	void set_noiseGate(float thres);
 	

src/control_WM8731_F32.cpp

@@ -236,13 +236,12 @@ bool AudioControlWM8731_F32::enable(bit_depth_t bits, TwoWire *i2cBus, uint8_t a
 void AudioControlWM8731_F32::dac_mute(bool m)
 {
 	modify(WM8731_REG_DIGITAL, m ? WM8731_BITS_DACMU(1) : WM8731_BITS_DACMU(0), WM8731_BITS_DACMU_MASK);
-	//write(WM8731_REG_DIGITAL, ); // DAC soft mute
 	DACmute = m;
 }
 // ----------------------------------------------------------------------------------
 void AudioControlWM8731_F32::hp_filter(bool state)
 {
-	modify(WM8731_REG_DIGITAL, WM8731_BITS_ADCHPD(state), WM8731_BITS_ADCHPD_MASK);
+	modify(WM8731_REG_DIGITAL, WM8731_BITS_ADCHPD(state^1), WM8731_BITS_ADCHPD_MASK);
 	//write(WM8731_REG_DIGITAL, WM8731_BITS_DACMU(DACmute) | WM8731_BITS_ADCHPD(state));
 }
 // ----------------------------------------------------------------------------------

src/effect_delaystereo_F32.cpp

@@ -232,9 +232,13 @@ void AudioEffectDelayStereo_F32::freeze(bool state)
 	{
 		feedb_tmp = feedb;      // store the settings
 		inputGain_tmp = inputGainSet;
-		bassCut_k_tmp = bassCut_k;
+		// bassCut_k_tmp = bassCut_k;
-		trebleCut_k_tmp = trebleCut_k;
+		// trebleCut_k_tmp = trebleCut_k;
 		__disable_irq();
+		flt0R.bypass_set(true);
+		flt1R.bypass_set(true);
+		flt0L.bypass_set(true);
+		flt1L.bypass_set(true);
 		feedb = 1.0f; // infinite echo                                    
 		inputGainSet = freeze_ingain;
 		__enable_irq();
@@ -244,8 +248,12 @@ void AudioEffectDelayStereo_F32::freeze(bool state)
 		__disable_irq();
 		feedb = feedb_tmp;
 		inputGainSet = inputGain_tmp;
-		bassCut_k = bassCut_k_tmp;
+		flt0R.bypass_set(false);
-		trebleCut_k = trebleCut_k_tmp;
+		flt1R.bypass_set(false);
+		flt0L.bypass_set(false);
+		flt1L.bypass_set(false);		
+		// bassCut_k = bassCut_k_tmp;
+		// trebleCut_k = trebleCut_k_tmp;
 		__enable_irq();
 	}
 }

src/effect_gainStereo_F32.h

@@ -76,12 +76,9 @@ public:
 
 	void setPan(float32_t p)
 	{
-		float32_t tmp, gL, gR;
+		float32_t gL, gR;
-		pan = constrain(p, -1.0f, 1.0f);
+		pan = constrain(p, 0.0f, 1.0f);
-		tmp = (pan + 1.0f) * 0.5f; // map to 0..1
+		mix_pwr(pan, &panR, &panL);
-		
-		mix_pwr(tmp, &panR, &panL);
-
 
 		gL = panL * gain;
 		gR = panR * gain;	
@@ -89,7 +86,6 @@ public:
 		gainLset = gL;
 		gainRset = gR;
 		__enable_irq();
-
 	}
 	float32_t getPan() { return pan;}
 	void phase_inv(bool inv)

src/effect_guitarBooster_F32.h

@@ -1,6 +1,3 @@
-#ifndef _EFFECT_GUITARBOOSTER_F32_H_
-#define _EFFECT_GUITARBOOSTER_F32_H_
-
 /**
  * @file effect_guitarBooster_F32.h
  * @author Piotr Zapart
@@ -20,6 +17,8 @@
  * You should have received a copy of the GNU General Public License along with this program. 
  * If not, see <https://www.gnu.org/licenses/>."
  */
+#ifndef _EFFECT_GUITARBOOSTER_F32_H_
+#define _EFFECT_GUITARBOOSTER_F32_H_
 
 #include <AudioStream_F32.h>
 #include "basic_DSPutils.h"
@@ -78,7 +77,8 @@ public:
 	{
 		value = fabs(value);
 		value = constrain(value, 0.0f, 1.0f);
-		value = 1.0f + value * upsample_k * gainRange;
+		// start with 0.5 - L+R are summed giving x2 gain
+		value = 0.5f + value * upsample_k * gainRange;
 		__disable_irq()
 		gainSet = value;
 		__enable_irq();
@@ -168,7 +168,7 @@ private:
 	float32_t dryGain = 0.0f;
 	float32_t wetGain = 1.0f;
 	float32_t  DCbias = 0.175f;
-	float32_t gainRange = 4.0f;
+	float32_t gainRange = 4.5f;
     float32_t gainSet = 1.0f; // gain is in range 0.0 to 1.0, scaled to 0.0 to gainRange
     float32_t gain = 0.0f;
 	float32_t gain_hp = 1.0f;

src/effect_noiseGateStereo_F32.h

@@ -86,7 +86,7 @@ public:
 		}		
 		//sum L + R
 		arm_add_f32(p_sideChain_inL, p_sideChain_inR, blockSideCh->data, blockSideCh->length);
-		arm_scale_f32(blockSideCh->data, 0.5f, blockSideCh->data, blockSideCh->length); // divide by 2
+		arm_scale_f32(blockSideCh->data, sideChain_gain * 0.5f, blockSideCh->data, blockSideCh->length); // divide by 2
 		
 		calcGain(blockSideCh, blockGain);
 		calcSmoothedGain(blockGain);
@@ -153,6 +153,10 @@ public:
 		time = map_sat(time, 0.0f, 1.0f, NOISEGATE_HOLDT_MIN, NOISEGATE_HOLDT_MAX);
 		setHoldTime(time);
 	}
+	void setSideChainGain(float g)
+	{
+		sideChain_gain = g;
+	}
 
 	bool infoIsOpen()
 	{
@@ -176,6 +180,7 @@ private:
 	float32_t fs = AUDIO_SAMPLE_RATE_EXACT;
 	float32_t* p_sideChain_inL = NULL;
 	float32_t* p_sideChain_inR = NULL;
+	float32_t sideChain_gain = 1.0f;;
 	float32_t linearThreshold;
 	float32_t prev_gain_dB = 0;
 	float32_t openingTimeConst, closingTimeConst;

src/effect_platereverb_F32.cpp

@@ -55,6 +55,8 @@ bool AudioEffectPlateReverb_F32::begin()
 	loop_allp_k = LOOP_ALLOP_COEFF;
 	rv_time_scaler = 1.0f;
 	rv_time_k = 0.2f;
+	pitch_semit = 0;
+	pitchShim_semit = 0;
 
 	if(!in_allp_1L.init(&in_allp_k)) return false;
 	if(!in_allp_2L.init(&in_allp_k)) return false;

src/effect_platereverb_F32.h

@@ -292,7 +292,9 @@ public:
 	void chorus(float c)
 	{
 		c = map(c, 0.0f, 1.0f, 1.0f, 100.0f);
+		__disable_irq();
 		LFO_AMPLset = (uint32_t)c; 
+		__enable_irq();
 	}
 
 	/**
@@ -305,9 +307,11 @@ public:
 		if (flags.freeze) return; // do not update the shimmer if in freeze mode
 		s = constrain(s, 0.0f, 1.0f);
 		s = 2*s - s*s;
+		__disable_irq();
 		pitchShimL.setMix(s);
 		pitchShimR.setMix(s);
 		shimmerRatio = s;
+		__enable_irq();
 	}
 	/**
 	 * @brief Sets the pitch of the shimmer effect
@@ -316,8 +320,10 @@ public:
 	 */
 	void shimmerPitch(float ratio)
 	{
+		__disable_irq();
 		pitchShimL.setPitch(ratio);
 		pitchShimR.setPitch(ratio);
+		__enable_irq();
 	}
 	/**
 	 * @brief Sets the shimmer effect pitch in semitones
@@ -326,9 +332,28 @@ public:
 	 */
 	void shimmerPitchSemitones(int8_t semitones)
 	{
+		__disable_irq();
 		pitchShimL.setPitchSemintone(semitones);
 		pitchShimR.setPitchSemintone(semitones);
+		__enable_irq();
 	}
+	/**
+	 * @brief shimemr pitch set using built in semitzone table
+	 * 
+	 * 
+	 * @param value float 0.0f to 1.0f
+	 */
+	void shimmerPitchNormalized(float32_t value)
+	{
+		value = constrain(value, 0.0f, 1.0f);
+		float32_t idx = map(value, 0.0f, 1.0f, 0.0f, (float32_t)sizeof(semitoneTable)+0.499f);
+		pitchShim_semit = semitoneTable[(uint8_t)idx];
+		__disable_irq();
+		pitchShimL.setPitchSemintone(pitchShim_semit);
+		pitchShimR.setPitchSemintone(pitchShim_semit);
+		__enable_irq();
+	}
+	int8_t shimmerPitch_get() {return pitchShim_semit;}
 	/**
 	 * @brief set the reverb pitch.  Range -12 to +24 
 	 * 
@@ -336,9 +361,28 @@ public:
 	 */
 	void pitchSemitones(int8_t semitones)
 	{
+		__disable_irq();
 		pitchL.setPitchSemintone(semitones);
 		pitchR.setPitchSemintone(semitones);
+		__enable_irq();
 	}
+	/**
+	 * @brief sets the reverb pitch using the built in table
+	 * 			input range is float 0.0 to 1.0
+	 * 
+	 * @param value  
+	 */
+	void pitchNormalized(float32_t value)
+	{
+		value = constrain(value, 0.0f, 1.0f);
+		float32_t idx = map(value, 0.0f, 1.0f, 0.0f, (float32_t)sizeof(semitoneTable)+0.499f);
+		pitch_semit = semitoneTable[(uint8_t)idx];
+		__disable_irq();
+		pitchL.setPitchSemintone(pitch_semit);
+		pitchR.setPitchSemintone(pitch_semit);
+		__enable_irq();
+	}
+	int8_t pitch_get() {return pitch_semit;}
 	/**
 	 * @brief Reverb pitch shifter dry/wet mixer
 	 * 
@@ -347,9 +391,11 @@ public:
 	void pitchMix(float s)
 	{
 		s = constrain(s, 0.0f, 1.0f);
+		__disable_irq();
 		pitchL.setMix(s);
 		pitchR.setMix(s);
 		pitchRatio = s;
+		__enable_irq();
 	}
 
 private:
@@ -451,6 +497,9 @@ private:
 	AudioBasicPitch	pitchShimL;
 	AudioBasicPitch	pitchShimR;
 
+	const int8_t semitoneTable[9] = {-12, -7, -5, -3, 0, 3, 5, 7, 12};
+	int8_t pitch_semit;
+	int8_t pitchShim_semit;
     const float rv_time_k_max = 0.97f;
     float rv_time_k, rv_time_k_tmp;         // reverb time coeff
     float rv_time_scaler;    // with high lodamp settings lower the max reverb time to avoid clipping

src/effect_wahMono_F32.cpp

@@ -0,0 +1,254 @@
+/**
+ * @file effect_wahMono_F32.cpp
+ * @author Piotr Zapart
+ * @brief Mono WAH effect
+ * @version 0.1
+ * @date 2024-07-09
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * 
+ * Implementation is based on the work of Transmogrifox
+ * https://cackleberrypines.net/transmogrifox/src/bela/inductor_wah_C_src/
+ * 
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+
+
+#include "effect_wahMono_F32.h"
+
+#define k *1e3
+#define nF *1e-9
+const AudioEffectWahMono_F32::wah_componentValues_t AudioEffectWahMono_F32::compValues[WAH_MODEL_LAST] = 
+{//  Lp     Cf        Ci        Rpot      Ri       Rs      Rp,      Rc   	Rbias     Re      beta      name
+	{0.50f, 10.0f nF, 10.0f nF, 100.0f k, 68.0f k, 1.5f k, 33.0f k, 22.0f k, 470.0f k, 390.0f, 250.0f}, // G1
+	{0.50f, 10.0f nF, 10.0f nF, 100.0f k, 68.0f k, 1.5f k, 33.0f k, 22.0f k, 470.0f k, 510.0f, 650.0f}, // G2
+	{0.66f, 10.0f nF, 10.0f nF, 100.0f k, 68.0f k, 1.5f k, 33.0f k, 22.0f k, 470.0f k, 390.0f, 250.0f}, // G3
+	{0.50f, 10.0f nF, 10.0f nF, 100.0f k, 68.0f k, 1.5f k, 100.0f k,22.0f k, 470.0f k, 470.0f, 250.0f}, // G4
+	{0.50f, 15.0f nF,  8.0f nF, 100.0f k, 220.0f k, 0.1f k, 47.0f k, 22.0f k, 470.0f k, 510.0f, 250.0f}, // VOCAL
+	{0.50f, 10.0f nF, 47.0f nF, 100.0f k, 68.0f k, 1.5f k,150.0f k, 22.0f k, 470.0f k, 150.0f, 250.0f},  // EXTREME
+	{0.66f, 10.0f nF, 10.0f nF, 100.0f k, 68.0f k, 1.5f k,33.0f k, 22.0f k, 470.0f k, 470.0f, 250.0f},	 // CUSTOM
+	{0.50f, 100.0f nF, 10.0f nF, 100.0f k, 100.0f k, 1.5f k,100.0f k, 22.0f k, 470.0f k, 470.0f, 200.0f} // BASS
+};
+#undef k
+#undef nF
+
+void AudioEffectWahMono_F32::update()
+{
+	audio_block_f32_t *blockL, *blockR, *blockMod;
+	float32_t a0, a1, a2, ax, drySig;
+	uint16_t i;
+	if (bp) // handle bypass
+	{
+		blockL = AudioStream_F32::receiveReadOnly_f32(0);
+		blockR = AudioStream_F32::receiveReadOnly_f32(1);
+		if (!blockL || !blockR)
+		{
+			if (blockL)
+				AudioStream_F32::release(blockL);
+			if (blockR)
+				AudioStream_F32::release(blockR);
+			return;
+		}
+		AudioStream_F32::transmit(blockL, 0);
+		AudioStream_F32::transmit(blockR, 1);
+		AudioStream_F32::release(blockL);
+		AudioStream_F32::release(blockR);
+		return;
+	}
+	blockL = AudioStream_F32::receiveWritable_f32(0);
+	blockR = AudioStream_F32::receiveWritable_f32(1);
+	if (!blockL || !blockR)
+	{
+		if (blockL)
+			AudioStream_F32::release(blockL);
+		if (blockR)
+			AudioStream_F32::release(blockR);
+		return;
+	}
+	blockMod = AudioStream_F32::receiveReadOnly_f32(2);
+
+	arm_add_f32(blockL->data, blockR->data, blockL->data, blockL->length); // add two channels
+	arm_scale_f32(blockL->data, input_gain, blockL->data, blockL->length);
+	for (i=0; i<blockL->length; i++)
+	{
+		//variable gp is the pot gain, nominal range 0.0 to 1.0
+		//although this can be abused for extended range.
+		//A value less than zero would make the filter go unstable
+		//and put a lot of NaNs in your audio output
+		if (blockMod) 
+		{
+			// TODO: scale the range to 0.0-1.0
+			gp = blockMod->data[i];
+		}
+		if(gp < 0.0f) gp = 0.0f;
+		float32_t gp_scaled = map_sat(gp, 0.0f, 1.0f, gp_top, gp_btm);
+		
+		//The magic numbers below approximate frequency warping characteristic
+		float32_t gw = 4.6f-18.4f/(4.0f + gp_scaled);
+
+		//Update Biquad coefficients
+		a0 = a0b + gw*a0c;
+		a1 = -(a1b + gw*a1c);
+		a2 = -(a2b + gw*a2c);
+		ax = 1.0f/a0;
+		
+		drySig = blockL->data[i];
+
+		//run it through the 1-pole HPF and gain first
+		float32_t hpf = ghpf * (drySig - xh1) - a1p*yh1;
+		xh1 = drySig;
+		yh1 = hpf;
+
+		//Apply modulated biquad
+		float32_t y0 = b0*hpf + b1*x1 + b2*x2 + a1*y1 + a2*y2;
+		y0 *= ax;
+		float32_t  out = clip(y0);
+		y0 = 0.95f*y0 + 0.05f*out; //Let a little harmonic distortion feed back into the filter loop
+		x2 = x1;
+		x1 = hpf;
+		y2 = y1;
+		y1 = y0;
+		
+		out = out * wet_gain + drySig * dry_gain;
+
+		blockL->data[i] = out;
+		blockR->data[i] = out;
+	}
+	AudioStream_F32::transmit(blockL, 0); // send blockL on both output channels
+	AudioStream_F32::transmit(blockL, 1);
+	AudioStream_F32::release(blockL);
+	AudioStream_F32::release(blockR);
+	if (blockMod) AudioStream_F32::release(blockMod);
+}
+
+FLASHMEM void AudioEffectWahMono_F32::setModel(wahModel_t model)
+{
+	if (model > WAH_MODEL_LAST) model = WAH_MODEL_G1;
+	
+	float32_t _b0, _b1, _b2;
+	float32_t _a0c, _a1c, _a2c;
+	float32_t _b0h, _b1h, _b2h;
+	float32_t _b0b, _b2b;
+	float32_t _a0b, _a1b, _a2b;
+	float32_t _a1p, _ghpf;
+
+	//helper variables 
+	float32_t ro = 0.0f;
+	float32_t re = 0.0f;
+	float32_t Req = 0.0f;
+	float32_t ic = 0.0f;
+
+    float32_t RpRi, f0, w0, Q, c, s, alpha;
+
+	//Equivalent output resistance seen from BJT collector
+	ro = compValues[model].Rc * compValues[model].Rpot / (compValues[model].Rc + compValues[model].Rpot) ;
+	ro = ro * compValues[model].Rbias / (ro + compValues[model].Rbias);
+ 	ic = 3.7f/compValues[model].Rc;  //Typical bias current
+	re = 0.025f/ic; 		//BJT gain stage equivalent internal emitter resistance
+							// gm = Ic/Vt, re = 1/gm
+	Req = compValues[model].Re + re;
+	gf = -ro/Req;  //forward gain of transistor stage
+	re = compValues[model].beta*Req; //Resistance looking into BJT emitter
+	Rp = compValues[model].Rp*re/(re + compValues[model].Rp);
+
+	RpRi = Rp*compValues[model].Ri/(Rp + compValues[model].Ri);
+	f0 = 1.0f/(2.0f*M_PI*sqrtf(compValues[model].Lp*compValues[model].Cf));
+	w0 = 2.0f*M_PI*f0/fs;
+	Q = RpRi*sqrtf(compValues[model].Cf/compValues[model].Lp);
+	c = cosf(w0);
+	s = sinf(w0);
+	alpha = s/(2.0f*Q);
+
+	//High Pass Biquad Coefficients
+	_b0h = (1.0f + c)/2.0f;
+	_b1h = -(1.0f + c);
+	_b2h = (1.0f + c)/2.0f;
+
+    //Band-pass biquad coefficients
+	_b0b = Q*alpha;
+	_b2b = -Q*alpha;
+	_a0b = 1.0f + alpha;
+	_a1b = -2.0f*c;
+	_a2b = 1.0f - alpha;
+
+    //1-pole high pass filter coefficients
+    // H(z) = g * (1 - z^-1)/(1 - a1*z^-1)
+    // Direct Form 1:
+    //      y[n] = ghpf * ( x[n] - x[n-1] ) - a1p*y[n-1]
+
+    _a1p = -expf(-1.0f/(compValues[model].Ri*compValues[model].Ci*fs));
+    _ghpf = gf*(1.0f - a1p)*0.5f;   //BJT forward gain worked in here to
+                                        // save extra multiplications in
+                                        // updating biquad coefficients
+
+    //Distill all down to final biquad coefficients
+    float32_t Gi = compValues[model].Rs/(compValues[model].Ri + compValues[model].Rs);
+    float32_t gbpf = 1.0f/(2.0f*M_PI*f0*compValues[model].Ri*compValues[model].Cf);  //band-pass component equivalent gain
+
+    //Final Biquad numerator coefficients
+    _b0 = gbpf*b0b + Gi*a0b;
+    _b1 = Gi*a1b;
+    _b2 = gbpf*b2b + Gi*a2b;
+
+    //Constants to make denominator coefficients computation more efficient
+    //in real-time
+	_a0c = -gf*b0h;
+	_a1c = -gf*b1h;
+	_a2c = -gf*b2h;
+
+	__disable_irq();
+	b0h = _b0h;
+	b1h = _b1h;
+	b2h = _b2h;
+	b0b = _b0b;
+	b2b = _b2b;
+	a0b = _a0b;
+	a1b = _a1b;
+	a2b = _a2b;
+	a1p = _a1p;
+	ghpf = _ghpf;
+    b0 = _b0;
+    b1 = _b1;
+    b2 = _b2;
+	a0c = _a0c;
+	a1c = _a1c;
+	a2c = _a2c;
+	y1 = 0.0f;	//biquad state variables
+	y2 = 0.0f;
+	x1 = 0.0f;
+	x2 = 0.0f;
+	//First order high-pass filter state variables
+	yh1 = 0.0f;
+	xh1 = 0.0f;	
+	__enable_irq();
+}
+
+
+float32_t AudioEffectWahMono_F32::clip(float32_t x)
+{
+
+    float32_t thrs = 0.8f;
+    float32_t nthrs = -0.72f;
+    float32_t f=1.25f;
+
+    //Hard limiting
+    if(x >= 1.2f) x = 1.2f;
+    if(x <= -1.12f) x = -1.12f;
+    
+    //Soft clipping
+    if(x > thrs)
+	{
+        x -= f*sqr(x - thrs);
+    }
+    if(x < nthrs){
+        x += f*sqr(x - nthrs);
+    }   
+    return x;
+}

src/effect_wahMono_F32.h

@@ -0,0 +1,167 @@
+/**
+ * @file effect_wahMono_F32.h
+ * @author Piotr Zapart
+ * @brief Mono WAH effect
+ * @version 0.1
+ * @date 2024-07-09
+ * 
+ * @copyright Copyright (c) 2024 www.hexefx.com
+ * 
+ * Implementation is based on the work of Transmogrifox
+ * https://cackleberrypines.net/transmogrifox/src/bela/inductor_wah_C_src/
+ * 
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+#ifndef _EFFECT_WAHMONO_F32_H_
+#define _EFFECT_WAHMONO_F32_H_
+
+#include <Arduino.h>
+#include "AudioStream_F32.h"
+#include "basic_DSPutils.h"
+
+typedef enum
+{
+	WAH_MODEL_G1 = 0,
+	WAH_MODEL_G2,
+	WAH_MODEL_G3,
+	WAH_MODEL_G4,
+	WAH_MODEL_VOCAL,
+	WAH_MODEL_EXTREME,
+	WAH_MODEL_CUSTOM,
+	WAH_MODEL_BASS,
+	WAH_MODEL_LAST
+}wahModel_t;
+
+class AudioEffectWahMono_F32 : public AudioStream_F32
+{
+public:
+	AudioEffectWahMono_F32(void) : AudioStream_F32(3, inputQueueArray_f32) 
+	{
+		setModel(WAH_MODEL_G3);
+	}
+	// Alternate specification of block size.  Sample rate does not apply for analyze_rms
+	AudioEffectWahMono_F32(const AudioSettings_F32 &settings) : AudioStream_F32(3, inputQueueArray_f32)
+	{
+		block_size = settings.audio_block_samples;
+		fs = settings.sample_rate_Hz;
+		setModel(WAH_MODEL_G3);
+	}
+	virtual void update();
+
+	void setModel(wahModel_t model);
+	void setFreq(float32_t val)
+	{
+		val = constrain(val, 0.0f, 1.0f);
+		val = 1.0f - val;
+		__disable_irq();
+		gp = val;
+		__enable_irq();
+	}
+	void setRange(float32_t heel, float32_t toe)
+	{
+		gp_top = 1.0f - constrain(heel, 0.0f, 1.0f);
+		gp_btm = 1.0f - constrain(toe, 0.0f, 1.0f);
+	}
+
+	void setMix(float32_t mix)
+	{
+		mix = constrain(mix, 0.0f, 1.0f);
+		float32_t dry, wet;
+		mix_pwr(mix, &wet, &dry);
+		__disable_irq();
+		dry_gain = dry;
+		wet_gain = wet;
+		__enable_irq();
+	}
+    bool bypass_get(void) {return bp;}
+    void bypass_set(bool state) {bp = state;}
+    bool bypass_tgl(void) 
+    {
+        bp ^= 1; 
+        return bp;
+    }	
+private:
+	bool bp = true; // bypass flag
+	audio_block_f32_t *inputQueueArray_f32[3];
+	uint16_t block_size = AUDIO_BLOCK_SAMPLES;
+	float32_t fs = AUDIO_SAMPLE_RATE_EXACT;
+	typedef struct
+	{
+		//Circuit parameters
+		//Using these makes it straight-forward to model other 
+		//variants of the circuit
+		float32_t Lp; //RLC tank inductor
+		float32_t Cf; //feedback capacitor
+		float32_t Ci; //input capacitor
+		
+		float32_t Rpot; //Pot resistance value
+		float32_t Ri; //input feed resistor
+		float32_t Rs; //RLC tank to BJT base resistor (dry mix)
+		float32_t Rp; //resistor placed parallel with the inductor
+		
+		//Gain-setting components
+		float32_t Rc; //BJT gain stage collector resistor
+		
+		float32_t Rbias; //Typically 470k bias resistor shows up in parallel with output
+		float32_t Re; //BJT gain stage emitter resistor
+		
+		float32_t beta;	//BJT forward gain
+	}wah_componentValues_t;
+
+	static const wah_componentValues_t compValues[WAH_MODEL_LAST];
+
+	float32_t gp = 0.0f;	// pot gain
+	float32_t input_gain = 0.5f;
+	float32_t dry_gain = 0.0f;
+	float32_t wet_gain = 1.0f;
+	float32_t gp_top = 0.0f;
+	float32_t gp_btm = 1.0f;
+
+	float32_t re; //equivalent resistance looking into input BJT base
+	float32_t Rp; //resistor placed parallel with the inductor
+	float32_t gf; 	//forward gain of BJT amplifier
+	//High-Pass biquad coefficients
+	float32_t b0h, b1h, b2h;
+
+	//Band-Pass biquad coefficients
+	float32_t b0b, b2b;
+	float32_t a0b, a1b, a2b;
+
+	//Final combined biquad coefficients used by run_filter()
+	float32_t b0;
+	float32_t b1;
+	float32_t b2;
+
+	float32_t a0c;
+	float32_t a1c;
+	float32_t a2c;
+	//First order high-pass filter coefficients
+	//y[n] = ghpf * ( x[n] - x[n-1] ) - a1p*y[n-1]
+	float32_t a1p;
+	float32_t ghpf;
+
+	//biquad state variables
+	float32_t y1;
+	float32_t y2;
+	float32_t x1;
+	float32_t x2;
+
+	//First order high-pass filter state variables
+	float32_t yh1;
+	float32_t xh1;
+	float32_t clip(float32_t x);
+	float32_t sqr(float32_t x)
+	{
+		return x*x;
+	}
+};
+
+
+#endif // _EFFECT_WAHMONO_F32_H_

src/filter_DCblockerStereo_F32.h

@@ -0,0 +1,153 @@
+/**
+ * @file filter_DCblockerStereo_F32.h
+ * @author Piotr Zapart
+ * @brief simple IIR based stereo DB blocking filter
+ * @version 0.1
+ * @date 2024-06-01
+ * 
+ * @copyright Copyright (c) 2024
+ * This program is free software: you can redistribute it and/or modify it under 
+ * the terms of the GNU General Public License as published by the Free Software Foundation, 
+ * either version 3 of the License, or (at your option) any later version.
+ * This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; 
+ * without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. 
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License along with this program. 
+ * If not, see <https://www.gnu.org/licenses/>."
+ */
+
+#ifndef _FILTER_DCBLOCKERSTEREO_F32_H_
+#define _FILTER_DCBLOCKERSTEREO_F32_H_
+
+#include <AudioStream_F32.h>
+#include "basic_DSPutils.h"
+#include <arm_math.h>
+
+//   y = x - xm1 + 0.995 * ym1;
+//   xm1 = x;
+//   ym1 = y;
+
+class AudioFilterDCblockerStereo_F32 : public AudioStream_F32
+{
+public:
+	AudioFilterDCblockerStereo_F32(void) : AudioStream_F32(2, inputQueueArray)
+	{
+		fs_Hz = AUDIO_SAMPLE_RATE_EXACT;
+		blockSize = AUDIO_BLOCK_SAMPLES;
+	}
+
+	AudioFilterDCblockerStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray)
+	{
+		fs_Hz = settings.sample_rate_Hz;
+		blockSize = settings.audio_block_samples;	
+	}
+
+	void update()
+	{
+		audio_block_f32_t *blockL, *blockR;
+		uint16_t i;
+		float32_t tmpf32;
+
+
+		if (bp) // handle bypass
+		{
+			blockL = AudioStream_F32::receiveReadOnly_f32(0);
+			blockR = AudioStream_F32::receiveReadOnly_f32(1);
+			if (!blockL || !blockR)
+			{
+				if (blockL)
+					AudioStream_F32::release(blockL);
+				if (blockR)
+					AudioStream_F32::release(blockR);
+				return;
+			}
+			AudioStream_F32::transmit(blockL, 0);
+			AudioStream_F32::transmit(blockR, 1);
+			AudioStream_F32::release(blockL);
+			AudioStream_F32::release(blockR);
+			return;
+		}
+		blockL = AudioStream_F32::receiveWritable_f32(0);
+		blockR = AudioStream_F32::receiveWritable_f32(1);
+		if (!blockL || !blockR)
+		{
+			if (blockL)
+				AudioStream_F32::release(blockL);
+			if (blockR)
+				AudioStream_F32::release(blockR);
+			return;
+		}
+
+		float32_t _xRegL = xRegL;
+		float32_t _xRegR = xRegR;
+		float32_t _yRegL = yRegL;
+		float32_t _yRegR = yRegR;
+
+		for (i=0; i<blockSize; i=i+4)
+		{
+			tmpf32 = blockL->data[i] - _xRegL + k * _yRegL;
+			_yRegL = tmpf32;
+			_xRegL = blockL->data[i];
+			blockL->data[i] = _yRegL;
+
+			tmpf32 = blockL->data[i+1] - _xRegL + k * _yRegL;
+			_yRegL = tmpf32;
+			_xRegL = blockL->data[i+1];
+			blockL->data[i+1] = _yRegL;
+
+			tmpf32 = blockL->data[i+2] - _xRegL + k * _yRegL;
+			_yRegL = tmpf32;
+			_xRegL = blockL->data[i+2];
+			blockL->data[i+2] = _yRegL;
+
+			tmpf32 = blockL->data[i+3] - _xRegL + k * _yRegL;
+			_yRegL = tmpf32;
+			_xRegL = blockL->data[i+3];
+			blockL->data[i+3] = _yRegL;
+
+			tmpf32 = blockR->data[i] - _xRegR + k * _yRegR;
+			_yRegR = tmpf32;
+			_xRegR = blockR->data[i];
+			blockR->data[i] = _yRegR;
+
+			tmpf32 = blockR->data[i+1] - _xRegR + k * _yRegR;
+			_yRegR = tmpf32;
+			_xRegR = blockR->data[i+1];
+			blockR->data[i+1] = _yRegR;
+
+			tmpf32 = blockR->data[i+2] - _xRegR + k * _yRegR;
+			_yRegR = tmpf32;
+			_xRegR = blockR->data[i+2];
+			blockR->data[i+2] = _yRegR;
+
+			tmpf32 = blockR->data[i+3] - _xRegR + k * _yRegR;
+			_yRegR = tmpf32;
+			_xRegR = blockR->data[i+3];
+			blockR->data[i+3] = _yRegR;
+		}
+		xRegL = _xRegL;
+		yRegL = _yRegL;
+		xRegR = _xRegR;
+		yRegR = _yRegR;
+
+		AudioStream_F32::transmit(blockL, 0); // send blockL on both output channels
+		AudioStream_F32::transmit(blockL, 1);
+		AudioStream_F32::release(blockL);
+		AudioStream_F32::release(blockR);
+
+	}
+
+private:
+	audio_block_f32_t *inputQueueArray[2];
+	float32_t fs_Hz;
+	uint16_t blockSize;
+	float k = 0.995f;
+	bool bp = false; // bypass flag
+	float32_t xRegL = 0.0f;
+	float32_t xRegR = 0.0f;
+	float32_t yRegL = 0.0f;
+	float32_t yRegR = 0.0f;
+};
+
+
+#endif // _FILTER_DCBLOCKERSTEREO_F32_H_

src/hexefx_audio_F32.h

@@ -4,6 +4,8 @@
 #include "control_WM8731_F32.h"
 #include "control_SGTL5000_F32.h"
 #include "control_ES8388_F32.h"
+#include "control_AK4452_F32.h"
+#include "control_AK5552_F32.h"
 
 #include "input_i2s_ext_F32.h"
 #include "output_i2s_ext_F32.h" // extended version
@@ -17,6 +19,7 @@
 #include "filter_equalizer_F32.h"
 #include "filter_3bandeq.h"
 #include "filter_biquadStereo_F32.h"
+#include "filter_DCblockerStereo_F32.h"
 
 #include "effect_gainStereo_F32.h"
 #include "effect_platereverb_F32.h"
@@ -30,5 +33,6 @@
 #include "effect_compressorStereo_F32.h"
 #include "effect_guitarBooster_F32.h"
 #include "effect_xfaderStereo_F32.h"
+#include "effect_wahMono_F32.h"
 
 #endif // _HEXEFX_AUDIO_H

src/input_i2s2_F32.cpp

@@ -178,8 +178,8 @@ void AudioInputI2S2_F32::update(void)
 		block_right_f32 = new_right;
 		block_offset = 0;
 		__enable_irq();
-		update_1chan(0, out_left);	// uses audio_block_samples and update_counter
+		update_1chan(0^channel_swap, out_left);	// uses audio_block_samples and update_counter
-		update_1chan(1, out_right); // uses audio_block_samples and update_counter
+		update_1chan(1^channel_swap, out_right); // uses audio_block_samples and update_counter
 	}
 	else if (new_left != NULL)
 	{

src/input_i2s2_F32.h

@@ -60,6 +60,8 @@ public:
 	int get_isOutOfMemory(void) { return flag_out_of_memory; }
 	void clear_isOutOfMemory(void) { flag_out_of_memory = 0; }
 	bool get_update_responsibility() { return update_responsibility;}
+	void set_channel_swap(bool sw) { channel_swap = sw ? 1 : 0;}
+	bool get_channel_swap() {return (bool)channel_swap;}
 protected:	
 	AudioInputI2S2_F32(int dummy): AudioStream_F32(0, NULL) {} // to be used only inside AudioInputI2Sslave !!
 	static bool update_responsibility;
@@ -74,6 +76,7 @@ private:
 	static uint16_t block_offset;
 	static int flag_out_of_memory;
 	static unsigned long update_counter;
+	uint8_t channel_swap = 0;
 };
 
 class AudioInputI2S2slave_F32 : public AudioInputI2S2_F32

src/input_i2s_ext_F32.cpp

@@ -199,8 +199,8 @@ void AudioInputI2S_ext_F32::update(void)
 		__enable_irq();
 
 		// update_counter++; //I chose to update it in the ISR instead.
-		update_1chan(0, out_left);	// uses audio_block_samples and update_counter
+		update_1chan(0^channel_swap, out_left);	// uses audio_block_samples and update_counter
-		update_1chan(1, out_right); // uses audio_block_samples and update_counter
+		update_1chan(1^channel_swap, out_right); // uses audio_block_samples and update_counter
 	}
 	else if (new_left != NULL)
 	{

src/input_i2s_ext_F32.h

@@ -58,7 +58,8 @@ public:
 	void begin(void);
 	int get_isOutOfMemory(void) { return flag_out_of_memory; }
 	void clear_isOutOfMemory(void) { flag_out_of_memory = 0; }
-
+	void set_channel_swap(bool sw) { channel_swap = sw ? 1 : 0;}
+	bool get_channel_swap() {return (bool)channel_swap;}
 protected:
 	AudioInputI2S_ext_F32(int dummy) : AudioStream_F32(0, NULL) {} // to be used only inside AudioInputI2Sslave !!
 	static bool update_responsibility;
@@ -74,7 +75,7 @@ private:
 	static uint16_t block_offset;
 	static int flag_out_of_memory;
 	static unsigned long update_counter;
-	static bool msbFirstMode; // some codecs like the new AKM series (AK4558) use MSB exclusively
+	uint8_t channel_swap = 0;
 };
 
 class AudioInputI2Sslave_ext_F32 : public AudioInputI2S_ext_F32

src/output_i2s2_F32.cpp

@@ -202,7 +202,7 @@ void AudioOutputI2S2_F32::update(void)
 		return;
 	}
 	// now that we have our working memory, proceed with getting the audio data and processing
-	block_f32 = receiveReadOnly_f32(0); // input 0 = left channel
+	block_f32 = receiveReadOnly_f32(0^channel_swap); // input 0 = left channel
 	if (block_f32)
 	{
 		if (block_f32->length != audio_block_samples)
@@ -247,7 +247,7 @@ void AudioOutputI2S2_F32::update(void)
 	}
 
 	block_f32_scaled = block2_f32_scaled; // this is simply renaming the pre-allocated buffer
-	block_f32 = receiveReadOnly_f32(1);	  // input 1 = right channel
+	block_f32 = receiveReadOnly_f32(1^channel_swap);	  // input 1 = right channel
 	if (block_f32)
 	{
 		scale_float_to_int32range(block_f32->data, block_f32_scaled->data, audio_block_samples);

src/output_i2s2_F32.h

@@ -63,6 +63,8 @@ public:
 	friend class AudioOutputI2SQuad_F32;
 	friend class AudioInputI2SQuad_F32;
 	bool get_update_responsibility() { return update_responsibility;}
+	void set_channel_swap(bool sw) { channel_swap = sw ? 1 : 0;}
+	bool get_channel_swap() {return (bool)channel_swap;}
 protected:
 	AudioOutputI2S2_F32(int dummy): AudioStream_F32(2, inputQueueArray) {} // to be used only inside AudioOutputI2Sslave !!
 	static void config_i2s(void);
@@ -84,6 +86,7 @@ private:
 	static float sample_rate_Hz;
 	static int audio_block_samples;
 	volatile uint8_t enabled = 1;
+	uint8_t channel_swap = 0;
 };
  
 class AudioOutputI2S2slave_F32 : public AudioOutputI2S2_F32

src/output_i2s_ext_F32.cpp

@@ -203,7 +203,7 @@ void AudioOutputI2S_ext_F32::update(void)
 		return;
 	}
 	// now that we have our working memory, proceed with getting the audio data and processing
-	block_f32 = receiveReadOnly_f32(0); // input 0 = left channel
+	block_f32 = receiveReadOnly_f32(0^channel_swap); // input 0 = left channel
 	if (block_f32)
 	{
 		if (block_f32->length != audio_block_samples)
@@ -251,7 +251,7 @@ void AudioOutputI2S_ext_F32::update(void)
 
 	block_f32_scaled = block2_f32_scaled; // this is simply renaming the pre-allocated buffer
 
-	block_f32 = receiveReadOnly_f32(1); // input 1 = right channel
+	block_f32 = receiveReadOnly_f32(1^channel_swap); // input 1 = right channel
 	if (block_f32)
 	{
 		// Optional scaling for easy volume control.  Leave outputScale==1.0f for default

src/output_i2s_ext_F32.h

@@ -60,6 +60,8 @@ public:
     void setGain(float _oscale) {outputScale = _oscale; }
 	virtual void update(void);
 	void begin(void);
+	void set_channel_swap(bool sw) { channel_swap = sw ? 1 : 0;}
+	bool get_channel_swap() {return (bool)channel_swap;}
 	friend class AudioInputI2S_ext_F32;
 	#if defined(__IMXRT1062__)
 		friend class AudioOutputI2SQuad_F32;
@@ -84,6 +86,7 @@ private:
 	static int audio_block_samples;
 	volatile uint8_t enabled = 1;
     float outputScale = 1.0f;  // Quick volume control
+	uint8_t channel_swap = 0;
 };
 
 class AudioOutputI2Sslave_ext_F32 : public AudioOutputI2S_ext_F32