major update and fixes

keywords.txt

@@ -27,6 +27,8 @@ setMix	KEYWORD2
 AudioFilterShelvingLPHP	KEYWORD1
 AudioFilterLP	KEYWORD1
 
+AudioBasicTempBuffer_F32	KEYWORD1
+
 AudioEffectInfinitePhaser_F32	KEYWORD1
 depth	KEYWORD2
 depth_top	KEYWORD2
@@ -43,7 +45,6 @@ AudioEffectMonoToStereo_F32	KEYWORD1
 stereo_set	KEYWORD2
 pan_set	KEYWORD2
 
-
 AudioEffectPhaserStereo_F32	KEYWORD1
 top	KEYWORD2
 btm	KEYWORD2
@@ -99,7 +100,7 @@ bass_cut	KEYWORD2
 AudioEffectDelayStereo_F32	KEYWORD1
 delay	KEYWORD2
 feedback	KEYWORD2
-inertia	
+inertia	KEYWORD2
 mod_rateHz	KEYWORD2
 mod_rate	KEYWORD2
 mod_depth	KEYWORD2
@@ -108,9 +109,66 @@ tap_tempo	KEYWORD2
 AudioEffectReverbSc_F32	KEYWORD1
 lowpass	KEYWORD2
 
+AudioEffectCompressorStereo_F32	KEYWORD1
+setDefaultValues	KEYWORD2
+calcAudioLevel_dB	KEYWORD2
+calcGain	KEYWORD2
+calcInstantaneousTargetGain	KEYWORD2
+calcSmoothedGain_dB	KEYWORD2
+resetStates	KEYWORD2
+setPreGain	KEYWORD2
+setPreGain_dB	KEYWORD2
+setPostGain	KEYWORD2
+setPostGain_dB	KEYWORD2
+setCompressionRatio	KEYWORD2
+setAttack_sec	KEYWORD2
+setRelease_sec	KEYWORD2
+setLevelTimeConst_sec	KEYWORD2
+setThresh_dBFS	KEYWORD2
+enableHPFilter	KEYWORD2
+getPreGain_dB	KEYWORD2
+getAttack_sec	KEYWORD2
+getRelease_sec	KEYWORD2
+getLevelTimeConst_sec	KEYWORD2
+getThresh_dBFS	KEYWORD2
+getCompressionRatio	KEYWORD2
+getCurrentLevel_dBFS	KEYWORD2
+getCurrentGain_dB	KEYWORD2
+setHPFilterCoeff_N2IIR_Matlab	
+bypass_get	KEYWORD2
+bypass_set	KEYWORD2
+bypass_tgl	KEYWORD2
+setSideChainMode	KEYWORD2
+
+AudioEffectGainStereo_F32	KEYWORD1
+setGain	KEYWORD2
+setGain_dB	KEYWORD2
+getGain	KEYWORD2
+getGain_dB	KEYWORD2
+setPan	KEYWORD2
+getPan	KEYWORD2
+
+AudioEffectGuitarBooster_F32	KEYWORD1
+drive	KEYWORD2
+bottom	KEYWORD2
+tone	KEYWORD2
+bias	KEYWORD2
+mix	KEYWORD2
+volume	KEYWORD2
+octave_get	KEYWORD2
+octave_set	KEYWORD2
+octave_tgl	KEYWORD2
+
 
 memcpyInterleave_f32	KEYWORD2
 memcpyDeinterleave_f32	KEYWORD2
 
 AudioInputI2S2_F32	KEYWORD1
 AudioOutputI2S2_F32	KEYWORD1
+
+AudioControlSGTL5000_Ext	KEYWORD1
+
+AudioControlWM8731_Ext	KEYWORD1
+dac_mute	KEYWORD2
+HPfilter	KEYWORD2
+

src/basic_DSPutils.cpp

@@ -0,0 +1,42 @@
+#include "basic_DSPutils.h"
+
+
+/**
+ * @brief scale a float vector (range -1.0 - 1.0) to a new float vector
+ *  in range of int32_t + saturation.
+ *  based on arm_float_to_31 
+ * 
+ * @param pSrc pointer to the source vector
+ * @param pDst 	pointer to the destination verctor
+ * @param blockSize 
+ */
+void scale_float_to_int32range(const float32_t *pSrc, float32_t *pDst, uint32_t blockSize)
+{
+	uint32_t blkCnt;			 /* Loop counter */
+	const float32_t *pIn = pSrc; /* Source pointer */
+	/* Loop unrolling: Compute 4 outputs at a time */
+	blkCnt = blockSize >> 2U;
+
+	while (blkCnt > 0U)
+	{
+		/* C = A * 2147483648 */
+		/* Convert from float to Q31 and then store the results in the destination buffer */
+		*pDst++ = (float32_t)clip_q63_to_q31((q63_t)(*pIn++ * 2147483648.0f));
+		*pDst++ = (float32_t)clip_q63_to_q31((q63_t)(*pIn++ * 2147483648.0f));
+		*pDst++ = (float32_t)clip_q63_to_q31((q63_t)(*pIn++ * 2147483648.0f));
+		*pDst++ = (float32_t)clip_q63_to_q31((q63_t)(*pIn++ * 2147483648.0f));
+		blkCnt--;
+	}
+
+	/* Loop unrolling: Compute remaining outputs */
+	blkCnt = blockSize % 0x4U;
+
+	while (blkCnt > 0U)
+	{
+		/* C = A * 2147483648 */
+		*pDst++ = (float32_t)clip_q63_to_q31((q63_t)(*pIn++ * 2147483648.0f));
+		/* Decrement loop counter */
+		blkCnt--;
+	}
+}
+

src/basic_DSPutils.h

@@ -1,14 +1,19 @@
 #ifndef _BASIC_DSPUTILS_H_
 #define _BASIC_DSPUTILS_H_
 
+#include <Arduino.h>
 #include <arm_math.h>
 
+#define F32_TO_I32_NORM_FACTOR (2147483647) // which is 2^31-1
+#define I32_TO_F32_NORM_FACTOR (4.656612875245797e-10)   //which is 1/(2^31 - 1)
+
 static inline void mix_pwr(float32_t mix, float32_t *wetMix, float32_t *dryMix);
 static inline void mix_pwr(float32_t mix, float32_t *wetMix, float32_t *dryMix)
 {
     // Calculate mix parameters
     //    A cheap mostly energy constant crossfade from SignalSmith Blog
     //    https://signalsmith-audio.co.uk/writing/2021/cheap-energy-crossfade/
+	mix = constrain(mix, 0.0f, 1.0f);
     float32_t x2 = 1.0f - mix;
     float32_t A = mix*x2;
     float32_t B = A * (1.0f + 1.4186f * A);
@@ -19,4 +24,41 @@ static inline void mix_pwr(float32_t mix, float32_t *wetMix, float32_t *dryMix)
     *dryMix = D * D;
 }
 
+void scale_float_to_int32range(const float32_t *pSrc, float32_t *pDst, uint32_t blockSize);
+
+/**
+  * @brief  combine two separate buffers into interleaved one
+  * @param  sz -  samples per output buffer (divisible by 2)
+  * @param  dst - pointer to source buffer
+  * @param  srcA - pointer to A source buffer (even samples)
+  * @param  srcB - pointer to B source buffer (odd samples)
+  * @retval none
+  */
+inline void memcpyInterleave_f32(float32_t *srcA, float32_t *srcB, float32_t *dst, int16_t sz)
+{
+	while(sz)
+	{
+		    *dst++ = *srcA++;
+		    *dst++ = *srcB++;
+			sz--;
+ 		    *dst++ = *srcA++;
+		    *dst++ = *srcB++;
+			sz--;           
+	}
+}
+inline void memcpyInterleave_f32(float32_t *srcA, float32_t *srcB, float32_t *dst, int16_t sz);
+
+inline void memcpyDeinterleave_f32(float32_t *src, float32_t *dstA, float32_t *dstB, int16_t sz)
+{
+	while(sz)
+	{
+		*dstA++ = *src++;
+		*dstB++ = *src++;
+		sz--;
+		*dstA++ = *src++;
+		*dstB++ = *src++;
+		sz--;		
+	}
+}
+inline void memcpyDeinterleave_f32(float32_t *src, float32_t *dstA, float32_t *dstB, int16_t sz);
 #endif // _BASIC_DSPUTILS_H_

src/basic_components.h

@@ -7,5 +7,6 @@
 #include "basic_shelvFilter.h"
 #include "basic_pitch.h"
 #include "basic_DSPutils.h"
+#include "basic_tempBuffer.h"
 
 #endif // _BASIC_COMPONENTS_H_

src/basic_delay.h

@@ -30,9 +30,10 @@ public:
 	bool init(uint32_t size_samples,  bool psram=false)
 	{
 		if(bf) free(bf);
+		use_psram = psram;
 		size = size_samples;
-		if (psram) bf = (float *)extmem_malloc(size * sizeof(float)); // allocate buffer
+		if (use_psram) 	bf = (float *)extmem_malloc(size * sizeof(float)); 	// allocate buffer in PSRAM
-		else bf = (float *)malloc(size * sizeof(float)); // allocate buffer
+		else 			bf = (float *)malloc(size * sizeof(float)); 		// allocate buffer in DMARAM
 		if (!bf) return false;
 		idx = 0;
 		reset();
@@ -40,7 +41,17 @@ public:
 	}
 	void reset()
 	{
-		memset(bf, 0, size * sizeof(float));
+		memset(bf, 0, size * sizeof(float32_t));
+		if (use_psram) arm_dcache_flush_delete(&bf[0], size * sizeof(float32_t));
+	}
+	void reset(uint32_t startAddr, uint32_t endAddr)
+	{
+		if (startAddr > endAddr) return;
+		if (endAddr > (uint32_t)size) endAddr = size;
+		float32_t* memPtr = &bf[0]+startAddr;
+		uint32_t l = (endAddr - startAddr) * sizeof(float32_t);
+		memset(memPtr, 0, l);
+		if (use_psram) arm_dcache_flush_delete(memPtr, l);
 	}
 	/**
 	 * @brief get the tap from the delay buffer
@@ -106,6 +117,7 @@ private:
 	int32_t size; 
 	float *bf;
 	int32_t idx;
+	bool use_psram = false;
 };
 
 #endif // _BASIC_DELAY_H_

src/basic_shelvFilter.h

@@ -59,6 +59,11 @@ public:
         hpreg += tmp1 * hp_f;
 		return (lpreg + hidamp*tmp2 + lodamp * hpreg);
 	}
+	void reset()
+	{
+		lpreg = 0.0f;
+		hpreg = 0.0f;
+	}
 private:
 	float lpreg;
 	float hpreg;

src/basic_tempBuffer.h

@@ -0,0 +1,37 @@
+#ifndef _BASIC_TEMPBUFFER_H_
+#define _BASIC_TEMPBUFFER_H_
+
+#include <Arduino.h>
+#include "AudioStream_F32.h"
+
+class AudioBasicTempBuffer_F32 : public AudioStream_F32
+{
+public:
+	AudioBasicTempBuffer_F32() : AudioStream_F32(1, inputQueueArray_f32)
+	{
+		blockSize = AUDIO_BLOCK_SAMPLES;
+		memset(&data[0], 0, AUDIO_BLOCK_SAMPLES * sizeof(float32_t));
+		dataPtr = &data[0];
+	};
+	AudioBasicTempBuffer_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32)
+	{
+		blockSize = settings.audio_block_samples;
+		memset(&data[0], 0, AUDIO_BLOCK_SAMPLES * sizeof(float32_t));
+		dataPtr = &data[0];
+	};
+	~AudioBasicTempBuffer_F32(){};
+	void update(void)
+	{
+		audio_block_f32_t* block = AudioStream_F32::receiveReadOnly_f32();
+		if (!block) return;
+		memcpy(&data[0], block->data, blockSize * sizeof(float32_t));
+		AudioStream_F32::release(block);
+	}
+	float32_t* dataPtr;
+private:
+	audio_block_f32_t *inputQueueArray_f32[1];
+	uint16_t blockSize = AUDIO_BLOCK_SAMPLES;
+	float32_t data[AUDIO_BLOCK_SAMPLES];
+};
+
+#endif // _BASIC_TEMPBUFFER_H_

src/control_SGTL5000_F32.cpp

@@ -0,0 +1,16 @@
+#include "control_SGTL5000_F32.h"
+#include <Wire.h>
+
+#define CHIP_I2S_CTRL			0x0006
+#define CHIP_ADCDAC_CTRL		0x000E
+
+void AudioControlSGTL5000_F32::set_bitDepth(bit_depth_t bits)
+{
+	uint16_t regTmp = read(CHIP_I2S_CTRL);
+	regTmp &= ~(0x30);						// clear bit 5:4 (DLEN)
+	regTmp |= ((uint8_t)bits << 4) & 0x30; 	// update DLEN
+
+	write(CHIP_ADCDAC_CTRL, 0x000C); 		// mute DAC
+	write(CHIP_I2S_CTRL, regTmp); 			// write new config
+	write(CHIP_ADCDAC_CTRL, 0x0000); 		// unmute DAC
+}

src/control_SGTL5000_F32.h

@@ -0,0 +1,40 @@
+#ifndef _CONTROL_SGTL5000_F32_H_
+#define _CONTROL_SGTL5000_F32_H_
+/**
+ * @file control_SGTL5000_ext.h
+ * @author Piotr Zapart 
+ * @brief enables the bit depth setting for the SGTL5000 codec chip
+ * @version 0.1
+ * @date 2024-03-20
+ * 
+ * @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 <Arduino.h>
+#include <control_sgtl5000.h>
+
+class AudioControlSGTL5000_F32 : public AudioControlSGTL5000
+{
+  //GUI: inputs:0, outputs:0  //this line used for automatic generation of GUI node
+  public:
+    AudioControlSGTL5000_F32(void) {};
+	typedef enum
+	{ 
+		I2S_BITS_32 = 0,
+		I2S_BITS_24,
+		I2S_BITS_20,
+		I2S_BITS_16
+	}bit_depth_t;
+
+	void set_bitDepth(bit_depth_t bits);
+};
+
+#endif // _CONTROL_SGTL5000_EXT_H_

src/control_WM8731_F32.cpp

@@ -0,0 +1,333 @@
+#include "control_WM8731_F32.h"
+#include <Wire.h>
+
+#define WM8731_I2C_ADDR_A 0x1A
+#define WM8731_I2C_ADDR_B 0x1B
+
+#define WM8731_MUTE_ON						(1)
+#define WM8731_MUTE_OFF						(0)
+
+#define WM8731_REG_LLINEIN					(0)
+	#define WM8731_BITS_LINVOL_SHIFT		(0)
+	#define WM8731_BITS_LINVOL_MASK			(0x1F)
+	#define WM8731_BITS_LINVOL(x)			(((x)<<WM8731_BITS_LINVOL_SHIFT)&WM8731_BITS_LINVOL_MASK)
+
+	#define WM8731_BITS_LINMUTE_SHIFT		(7)
+	#define WM8731_BITS_LINMUTE_MASK		(1<<WM8731_BITS_LINMUTE_SHIFT)
+	#define WM8731_BITS_LINMUTE(x)			(((x)<<WM8731_BITS_LINMUTE_SHIFT)&WM8731_BITS_LINMUTE_MASK)
+
+	#define WM8731_BITS_LRINBOTH_SHIFT		(8)
+	#define WM8731_BITS_LRINBOTH_MASK		(1<<WM8731_BITS_LRINBOTH_SHIFT)
+	#define WM8731_BITS_LRINBOTH(x)			(((x)<<WM8731_BITS_LRINBOTH_SHIFT)&WM8731_BITS_LRINBOTH_MASK)
+
+#define WM8731_REG_RLINEIN	(1)
+	#define WM8731_BITS_RINVOL_SHIFT		(0)
+	#define WM8731_BITS_RINVOL_MASK			(0x1F)
+	#define WM8731_BITS_RINVOL(x)			(((x)<<WM8731_BITS_RINVOL_SHIFT)&WM8731_BITS_RINVOL_MASK)
+
+	#define WM8731_BITS_RINMUTE_SHIFT		(7)
+	#define WM8731_BITS_RINMUTE_MASK		(1<<WM8731_BITS_RINMUTE_SHIFT)
+	#define WM8731_BITS_RINMUTE(x)			(((x)<<WM8731_BITS_RINMUTE_SHIFT)&WM8731_BITS_RINMUTE_MASK)
+
+	#define WM8731_BITS_RLINBOTH_SHIFT		(8)
+	#define WM8731_BITS_RLINBOTH_MASK		(1<<WM8731_BITS_RLINBOTH_SHIFT)
+	#define WM8731_BITS_RLINBOTH(x)			(((x)<<WM8731_BITS_RLINBOTH_SHIFT)&WM8731_BITS_RLINBOTH_MASK)
+
+#define WM8731_REG_LHEADOUT					(2)
+	#define WM8731_BITS_LHPVOL_SHIFT		(0)
+	#define WM8731_BITS_LHPVOL_MASK			(0x7F)
+	#define WM8731_BITS_LHPVOL(x)			(((x))<<WM8731_BITS_LHPVOL_SHIFT)&WM8731_BITS_LHPVOL_MASK)
+
+	#define WM8731_BITS_LZCEN_SHIFT			(7)
+	#define WM8731_BITS_LZCEN_MASK			(1<<WM8731_BITS_LZCEN_SHIFT)
+	#define WM8731_BITS_LZCEN(x)			(((x)<<WM8731_BITS_LZCEN_SHIFT)&WM8731_BITS_LZCEN_MASK)
+
+	#define WM8731_BITS_LRHPBOTH_SHIFT		(8)
+	#define WM8731_BITS_LRHPBOTH_MASK		(1<<WM8731_BITS_LRHPBOTH_SHIFT)
+	#define WM8731_BITS_LRHPBOTH(x)			(((x)<<WM8731_BITS_LRHPBOTH_SHIFT)&WM8731_BITS_LRHPBOTH_MASK)
+
+#define WM8731_REG_RHEADOUT					(3)
+	#define WM8731_BITS_RHPVOL_SHIFT		(0)
+	#define WM8731_BITS_RHPVOL_MASK			(0x7F)
+	#define WM8731_BITS_RHPVOL(x)			(((x))<<WM8731_BITS_RHPVOL_SHIFT)&WM8731_BITS_RHPVOL_MASK)
+
+	#define WM8731_BITS_RZCEN_SHIFT			(7)
+	#define WM8731_BITS_RZCEN_MASK			(1<<WM8731_BITS_RZCEN_SHIFT)
+	#define WM8731_BITS_RZCEN(x)			(((x)<<WM8731_BITS_RZCEN_SHIFT)&WM8731_BITS_RZCEN_MASK)
+
+	#define WM8731_BITS_RLHPBOTH_SHIFT		(8)
+	#define WM8731_BITS_RLHPBOTH_MASK		(1<<WM8731_BITS_RLHPBOTH_SHIFT)
+	#define WM8731_BITS_RLHPBOTH(x)			(((x)<<WM8731_BITS_RLHPBOTH_SHIFT)&WM8731_BITS_RLHPBOTH_MASK)
+
+#define WM8731_REG_ANALOG					(4)
+	#define WM8731_BITS_MICBOOST_SHIFT		(0)
+	#define WM8731_BITS_MICBOOST_MASK		(1<<WM8731_BITS_MICBOOST_SHIFT)
+	#define WM8731_BITS_MICBOOST(x)			(((x)<<WM8731_BITS_MICBOOST_SHIFT)&WM8731_BITS_MICBOOST_MASK)
+
+	#define WM8731_BITS_MUTEMIC_SHIFT		(1)
+	#define WM8731_BITS_MUTEMIC_MASK		(1<<WM8731_BITS_MUTEMIC_SHIFT)
+	#define WM8731_BITS_MUTEMIC(x)			(((x)<<WM8731_BITS_MUTEMIC_SHIFT)&WM8731_BITS_MUTEMIC_MASK)
+
+	#define WM8731_BITS_INSEL_SHIFT			(2)
+	#define WM8731_BITS_INSEL_MASK			(1<<WM8731_BITS_INSEL_SHIFT)
+	#define WM8731_BITS_INSEL(x)			(((x)<<WM8731_BITS_INSEL_SHIFT)&WM8731_BITS_INSEL_MASK)
+
+	#define WM8731_BITS_BYPASS_SHIFT		(3)
+	#define WM8731_BITS_BYPASS_MASK			(1<<WM8731_BITS_BYPASS_SHIFT)
+	#define WM8731_BITS_BYPASS(x)			(((x)<<WM8731_BITS_BYPASS_SHIFT)&WM8731_BITS_BYPASS_MASK)
+
+	#define WM8731_BITS_DACSEL_SHIFT		(4)
+	#define WM8731_BITS_DACSEL_MASK			(1<<WM8731_BITS_DACSEL_SHIFT)
+	#define WM8731_BITS_DACSEL(x)			(((x)<<WM8731_BITS_DACSEL_SHIFT)&WM8731_BITS_DACSEL_MASK)
+
+	#define WM8731_BITS_SIDETONE_SHIFT		(5)
+	#define WM8731_BITS_SIDETONE_MASK		(1<<WM8731_BITS_SIDETONE_SHIFT)
+	#define WM8731_BITS_SIDETONE(x)			(((x)<<WM8731_BITS_SIDETONE_SHIFT)&WM8731_BITS_SIDETONE_MASK)
+
+	#define WM8731_BITS_SIDEATT_SHIFT		(6)
+	#define WM8731_BITS_SIDEATT_MASK		(0xC0)
+	#define WM8731_BITS_SIDEATT(x)			(((x)<<WM8731_BITS_SIDEATT_SHIFT)&WM8731_BITS_SIDEATT_MASK)	
+	#define WM8731_SIDEATT_M6_DB			(0x00)
+	#define WM8731_SIDEATT_M9_DB			(0x01)
+	#define WM8731_SIDEATT_M12_DB			(0x02)
+	#define WM8731_SIDEATT_M15_DB			(0x03)
+
+#define WM8731_REG_DIGITAL					(5)
+	#define WM8731_BITS_ADCHPD_SHIFT		(0)
+	#define WM8731_BITS_ADCHPD_MASK			(1<<WM8731_BITS_ADCHPD_SHIFT)
+	#define WM8731_BITS_ADCHPD(x)			(((x)<<WM8731_BITS_ADCHPD_SHIFT)&WM8731_BITS_ADCHPD_MASK)
+
+	#define WM8731_BITS_DEEMP_SHIFT			(6)
+	#define WM8731_BITS_DEEMP_MASK			(0x06)
+	#define WM8731_BITS_DEEMP(x)			(((x)<<WM8731_BITS_DEEMP_SHIFT)&WM8731_BITS_DEEMP_MASK)	
+	#define WM8731_DEEMP_OFF				(0x00)
+	#define WM8731_DEEMP_32KHZ				(0x01)
+	#define WM8731_DEEMP_44_1KHZ			(0x02)
+	#define WM8731_DEEMP_48KHZ				(0x03)
+
+	#define WM8731_BITS_DACMU_SHIFT			(3)
+	#define WM8731_BITS_DACMU_MASK			(1<<WM8731_BITS_DACMU_SHIFT)
+	#define WM8731_BITS_DACMU(x)			(((x)<<WM8731_BITS_DACMU_SHIFT)&WM8731_BITS_DACMU_MASK)
+
+	#define WM8731_BITS_HPOR_SHIFT			(4)
+	#define WM8731_BITS_HPOR_MASK			(1<<WM8731_BITS_HPOR_SHIFT)
+	#define WM8731_BITS_HPOR(x)				(((x)<<WM8731_BITS_HPOR_SHIFT)&WM8731_BITS_HPOR_MASK)
+
+#define WM8731_REG_POWERDOWN				(6)
+	#define WM8731_BITS_LINEINPD_SHIFT		(0)
+	#define WM8731_BITS_LINEINPD_MASK		(1<<WM8731_BITS_LINEINPD_SHIFT)
+	#define WM8731_BITS_LINEINPD(x)			(((x)<<WM8731_BITS_LINEINPD_SHIFT)&WM8731_BITS_LINEINPD_MASK)
+
+	#define WM8731_BITS_MICPD_SHIFT			(1)
+	#define WM8731_BITS_MICPD_MASK			(1<<WM8731_BITS_MICPD_SHIFT)
+	#define WM8731_BITS_MICPD(x)			(((x)<<WM8731_BITS_MICPD_SHIFT)&WM8731_BITS_MICPD_MASK)
+
+	#define WM8731_BITS_ADCPD_SHIFT			(2)
+	#define WM8731_BITS_ADCPD_MASK			(1<<WM8731_BITS_ADCPD_SHIFT)
+	#define WM8731_BITS_ADCPD(x)			(((x)<<WM8731_BITS_ADCPD_SHIFT)&WM8731_BITS_ADCPD_MASK)
+
+	#define WM8731_BITS_DACPD_SHIFT			(3)
+	#define WM8731_BITS_DACPD_MASK			(1<<WM8731_BITS_DACPD_SHIFT)
+	#define WM8731_BITS_DACPD(x)			(((x)<<WM8731_BITS_DACPD_SHIFT)&WM8731_BITS_DACPD_MASK)
+
+	#define WM8731_BITS_OUTPD_SHIFT			(4)
+	#define WM8731_BITS_OUTPD_MASK			(1<<WM8731_BITS_OUTPD_SHIFT)
+	#define WM8731_BITS_OUTPD(x)			(((x)<<WM8731_BITS_OUTPD_SHIFT)&WM8731_BITS_OUTPD_MASK)
+
+	#define WM8731_BITS_OSCPD_SHIFT			(5)
+	#define WM8731_BITS_OSCPD_MASK			(1<<WM8731_BITS_OSCPD_SHIFT)
+	#define WM8731_BITS_OSCPD(x)			(((x)<<WM8731_BITS_OSCPD_SHIFT)&WM8731_BITS_OSCPD_MASK)
+
+	#define WM8731_BITS_CLKOUTPD_SHIFT		(6)
+	#define WM8731_BITS_CLKOUTPD_MASK		(1<<WM8731_BITS_CLKOUTPD_SHIFT)
+	#define WM8731_BITS_CLKOUTPD(x)			(((x)<<WM8731_BITS_CLKOUTPD_SHIFT)&WM8731_BITS_CLKOUTPD_MASK)
+
+	#define WM8731_BITS_POWEROFF_SHIFT		(7)
+	#define WM8731_BITS_POWEROFF_MASK		(1<<WM8731_BITS_POWEROFF_SHIFT)
+	#define WM8731_BITS_POWEROFF(x)			(((x)<<WM8731_BITS_POWEROFF_SHIFT)&WM8731_BITS_POWEROFF_MASK)
+
+#define WM8731_REG_INTERFACE				(7)
+	#define WM8731_BITS_FORMAT_SHIFT		(0)
+	#define WM8731_BITS_FORMAT_MASK			(0x03)
+	#define WM8731_BITS_FORMAT(x)			(((x)<<WM8731_BITS_FORMAT_SHIFT)&WM8731_BITS_FORMAT_MASK)	
+	#define WM8731_FORMAT_DSP_MODE			(0x03)
+	#define WM8731_FORMAT_I2S_MSB_LEFT		(0x02)
+	#define WM8731_FORMAT_MSB_LEFT			(0x01)
+	#define WM8731_FORMAT_MSB_RIGHT			(0x00)
+
+	#define WM8731_BITS_IWL_SHIFT			(2)
+	#define WM8731_BITS_IWL_MASK			(0x0C)
+	#define WM8731_BITS_IWL(x)				((x<<WM8731_BITS_IWL_SHIFT)&WM8731_BITS_IWL_MASK)
+
+	#define WM8731_BITS_LRP_SHIFT			(4)
+	#define WM8731_BITS_LRP_MASK			(1<<WM8731_BITS_LRP_SHIFT)
+	#define WM8731_BITS_LRP(x)				(((x)<<WM8731_BITS_LRP_SHIFT)&WM8731_BITS_LRP_MASK)
+
+	#define WM8731_BITS_LRSWAP_SHIFT		(5)
+	#define WM8731_BITS_LRSWAP_MASK			(1<<WM8731_BITS_LRSWAP_SHIFT)
+	#define WM8731_BITS_LRSWAP(x)			(((x)<<WM8731_BITS_LRSWAP_SHIFT)&WM8731_BITS_LRSWAP_MASK)
+
+	#define WM8731_BITS_MS_SHIFT			(6)
+	#define WM8731_BITS_MS_MASK				(1<<WM8731_BITS_MS_SHIFT)
+	#define WM8731_BITS_MS(x)				(((x)<<WM8731_BITS_MS_SHIFT)&WM8731_BITS_MS_MASK)	
+
+	#define WM8731_BITS_BCLKINV_SHIFT		(7)
+	#define WM8731_BITS_BCLKINV_MASK		(1<<WM8731_BITS_BCLKINV_SHIFT)
+	#define WM8731_BITS_BCLKINV(x)			(((x)<<WM8731_BITS_BCLKINV_SHIFT)&WM8731_BITS_BCLKINV_MASK)	
+
+#define WM8731_REG_SAMPLING					(8)
+	#define WM8731_BITS_USB_NORMAL_SHIFT	(0)
+	#define WM8731_BITS_USB_NORMAL_MASK		(1<<WM8731_BITS_USB_NORMAL_SHIFT)
+	#define WM8731_BITS_USB_NORMAL(x)		(((x)<<WM8731_BITS_USB_NORMAL_SHIFT)&WM8731_BITS_USB_NORMAL_MASK)
+
+	#define WM8731_BITS_BOSR_SHIFT			(1)
+	#define WM8731_BITS_BOSR_MASK			(1<<WM8731_BITS_BOSR_SHIFT)
+	#define WM8731_BITS_BOSR(x)				(((x)<<WM8731_BITS_BOSR_SHIFT)&WM8731_BITS_BOSR_MASK)
+
+	#define WM8731_BITS_SR_SHIFT			(2)
+	#define WM8731_BITS_SR_MASK				(0x3C)
+	#define WM8731_BITS_SR(x)				(((x)<<WM8731_BITS_SR_SHIFT)&WM8731_BITS_SR_MASK)
+
+	#define WM8731_BITS_CLKIDIV2_SHIFT		(6)
+	#define WM8731_BITS_CLKIDIV2_MASK		(1<<WM8731_BITS_CLKIDIV2_SHIFT)
+	#define WM8731_BITS_CLKIDIV2(x)			(((x)<<WM8731_BITS_CLKIDIV2_SHIFT)&WM8731_BITS_CLKIDIV2_MASK)	
+
+	#define WM8731_BITS_CLKODIV2_SHIFT		(7)
+	#define WM8731_BITS_CLKODIV2_MASK		(1<<WM8731_BITS_CLKODIV2_SHIFT)
+	#define WM8731_BITS_CLKODIV2(x)			(((x)<<WM8731_BITS_CLKODIV2_SHIFT)&WM8731_BITS_CLKODIV2_MASK)		
+
+#define WM8731_REG_ACTIVE					(9)
+#define WM8731_REG_RESET					(15)
+
+bool AudioControlWM8731_F32::enable(bit_depth_t bits, uint8_t addr)
+{
+	i2c_addr = addr;
+	Wire.begin();
+	delay(5);
+	if (!write(WM8731_REG_RESET, 0))
+	{
+		return false; // no WM8731 chip responding
+	}
+	write(WM8731_REG_INTERFACE, WM8731_BITS_FORMAT(WM8731_FORMAT_I2S_MSB_LEFT) | 
+							WM8731_BITS_IWL(bits)); // I2S, x bit, MCLK slave
+	write(WM8731_REG_SAMPLING, 0x20);						   // 256*Fs, 44.1 kHz, MCLK/1
+
+	// In order to prevent pops, the DAC should first be soft-muted (DACMU),
+	// the output should then be de-selected from the line and headphone output
+	// (DACSEL), then the DAC powered down (DACPD).
+
+	write(WM8731_REG_DIGITAL, 0x08); // DAC soft mute
+	write(WM8731_REG_ANALOG, 0x00);	 // disable all
+
+	write(WM8731_REG_POWERDOWN, 0x00); // codec powerdown
+
+	write(WM8731_REG_LHEADOUT, 0x80); // volume off
+	write(WM8731_REG_RHEADOUT, 0x80);
+
+	delay(100); // how long to power up?
+
+	write(WM8731_REG_ACTIVE, 1);
+	delay(5);
+	write(WM8731_REG_DIGITAL, 0x00); // DAC unmuted
+	write(WM8731_REG_ANALOG, 0x10);	 // DAC selected
+
+	return true;
+}
+
+void AudioControlWM8731_F32::dac_mute(bool m)
+{
+	write(WM8731_REG_DIGITAL, m ? WM8731_BITS_DACMU(1) : WM8731_BITS_DACMU(0)); // DAC soft mute
+	DACmute = m;
+}
+
+void AudioControlWM8731_F32::HPfilter(bool state)
+{
+	write(WM8731_REG_DIGITAL, WM8731_BITS_DACMU(DACmute) | WM8731_BITS_ADCHPD(state));
+}
+
+bool AudioControlWM8731_F32::write(unsigned int reg, unsigned int val)
+{
+	int attempt = 0;
+	while (1)
+	{
+		attempt++;
+		Wire.beginTransmission(i2c_addr);
+		Wire.write((reg << 1) | ((val >> 8) & 1));
+		Wire.write(val & 0xFF);
+		int status = Wire.endTransmission();
+		if (status == 0) return true;
+		if (attempt >= 12)  return false;
+		delayMicroseconds(80);
+	}
+}
+
+
+bool AudioControlWM8731_F32::volumeInteger(unsigned int n)
+{
+	// n = 127 for max volume (+6 dB)
+	// n = 48 for min volume (-73 dB)
+	// n = 0 to 47 for mute
+	if (n > 127)
+		n = 127;
+	// Serial.print("volumeInteger, n = ");
+	// Serial.println(n);
+	write(WM8731_REG_LHEADOUT, n | 0x180);
+	write(WM8731_REG_RHEADOUT, n | 0x80);
+	return true;
+}
+
+bool AudioControlWM8731_F32::inputLevel(float n)
+{
+	// range is 0x00 (min) - 0x1F (max)
+
+	int _level = int(n * 31.f);
+
+	_level = _level > 0x1F ? 0x1F : _level;
+	write(WM8731_REG_LLINEIN, _level);
+	write(WM8731_REG_RLINEIN, _level);
+	return true;
+}
+
+bool AudioControlWM8731_F32::inputSelect(input_select_t n)
+{
+	if (n == INPUT_SELECT_LINEIN) 	write(WM8731_REG_ANALOG, 0x12);
+	else if (n == INPUT_SELECT_MIC)	write(WM8731_REG_ANALOG, 0x15);
+	else return false;
+	return true;
+}
+
+/******************************************************************/
+
+bool AudioControlWM8731_F32_master::enable(bit_depth_t bits, uint8_t addr)
+{
+	i2c_addr = addr;
+	Wire.begin();
+	delay(5);
+	// write(WM8731_REG_RESET, 0);
+	write(WM8731_REG_INTERFACE, 
+			WM8731_BITS_FORMAT(WM8731_FORMAT_I2S_MSB_LEFT) | 
+			WM8731_BITS_IWL(bits)|
+			WM8731_BITS_MS(1)); 		// I2S, x bit, MCLK slave
+	write(WM8731_REG_SAMPLING, 0x20);  // 256*Fs, 44.1 kHz, MCLK/1
+
+	// In order to prevent pops, the DAC should first be soft-muted (DACMU),
+	// the output should then be de-selected from the line and headphone output
+	// (DACSEL), then the DAC powered down (DACPD).
+
+	write(WM8731_REG_DIGITAL, 0x08); // DAC soft mute
+	write(WM8731_REG_ANALOG, 0x00);	 // disable all
+
+	write(WM8731_REG_POWERDOWN, 0x00); // codec powerdown
+
+	write(WM8731_REG_LHEADOUT, 0x80); // volume off
+	write(WM8731_REG_RHEADOUT, 0x80);
+
+	delay(100); // how long to power up?
+
+	write(WM8731_REG_ACTIVE, 1);
+	delay(5);
+	write(WM8731_REG_DIGITAL, 0x00); // DAC unmuted
+	write(WM8731_REG_ANALOG, 0x10);	 // DAC selected
+
+	return true;
+}

src/control_WM8731_F32.h

@@ -0,0 +1,69 @@
+/**
+ * @file control_WM8731_ext.h
+ * @author Piotr Zapart
+ * @brief Alternative WM8731 driver with configurable bit depth
+ * @version 1.0
+ * @date 2024-03-21
+ * 
+ * @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_WM8731_F32_H_
+#define _CONTROL_WM8731_F32_H_
+
+#include <Arduino.h>
+
+#define WM8731_I2C_ADDR_CSB0 0x1A
+#define WM8731_I2C_ADDR_CSB1 0x1B
+
+class AudioControlWM8731_F32 
+{
+public:	
+	AudioControlWM8731_F32(){};
+	
+	typedef enum
+	{ 
+		I2S_BITS_16 = 0,
+		I2S_BITS_20,
+		I2S_BITS_24,
+		I2S_BITS_32
+	}bit_depth_t;
+
+	typedef enum
+	{
+		INPUT_SELECT_LINEIN = 0,
+		INPUT_SELECT_MIC
+	}input_select_t;
+
+	bool enable(bit_depth_t bits = I2S_BITS_16, uint8_t addr=WM8731_I2C_ADDR_CSB0);
+	bool disable(void) { return false; }
+	bool volume(float n) { return volumeInteger(n * 80.0f + 47.499f); }
+	bool inputLevel(float n); // range: 0.0f to 1.0f
+	bool inputSelect(input_select_t n=INPUT_SELECT_LINEIN);
+	void dac_mute(bool m);
+	void HPfilter(bool state);
+
+protected:
+	bool write(unsigned int reg, unsigned int val);
+	bool volumeInteger(unsigned int n); // range: 0x2F to 0x7F
+private:
+	uint8_t bit_depth = I2S_BITS_16;
+	uint8_t i2c_addr;
+	bool DACmute = false;
+};
+
+class AudioControlWM8731_F32_master : public AudioControlWM8731_F32
+{
+public:
+	bool enable(bit_depth_t bits = I2S_BITS_16, uint8_t addr=WM8731_I2C_ADDR_CSB0);
+private:
+	uint8_t i2c_addr;
+};
+#endif // _CONTROL_WM8731_EXTENDED_H_

src/effect_compressorStereo_F32.h

@@ -0,0 +1,447 @@
+/*
+   AudioEffectCompressor
+
+   Created: Chip Audette, Dec 2016 - Jan 2017
+   Purpose; Apply dynamic range compression to the audio stream.
+			Assumes floating-point data.
+
+   This processes a single stream fo audio data (ie, it is mono)
+
+   MIT License.  use at your own risk.
+
+	Stereo version - Piotr Zapart www.hexefx.com 03.2024
+
+*/
+
+#ifndef _EFFECT_COMPRESSORSTEREO_F32
+#define _EFFECT_COMPRESSORSTEREO_F32
+
+#include <arm_math.h> //ARM DSP extensions.  https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html
+#include <AudioStream_F32.h>
+
+class AudioEffectCompressorStereo_F32 : public AudioStream_F32
+{
+	// GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node
+public:
+	// constructor
+	AudioEffectCompressorStereo_F32(void) : AudioStream_F32(2, inputQueueArray_f32)
+	{
+		setDefaultValues(AUDIO_SAMPLE_RATE);
+		resetStates();
+	};
+
+	AudioEffectCompressorStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32)
+	{
+		setDefaultValues(settings.sample_rate_Hz);
+		resetStates();
+	};
+
+	typedef enum
+	{
+		COMP_SIDECHAIN_SRC_LR,			// l + r separate
+		COMP_SIDECHAIN_SRC_LRSUM		// l + r sum / 2
+	}sideChainMode_t;
+
+	void setDefaultValues(const float sample_rate_Hz)
+	{
+		fs_Hz = sample_rate_Hz;
+		setThresh_dBFS(-20.0f);					// set the default value for the threshold for compression
+		setCompressionRatio(5.0f);				// set the default copression ratio
+		setAttack_sec(0.005f);					// default to this value
+		setRelease_sec(0.200f); 				// default to this value
+		setHPFilterCoeff();
+		enableHPFilter(true); // enable the HP filter to remove any DC offset from the audio
+		sidechainMode = COMP_SIDECHAIN_SRC_LRSUM;
+	}
+
+	// here's the method that does all the work
+	void update(void)
+	{
+		audio_block_f32_t *blockL, *blockR;
+		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;
+		}
+		// allocate blocks required for gain calculations
+		audio_block_f32_t* audio_level_dB_blockL = AudioStream_F32::allocate_f32();
+		audio_block_f32_t* audio_level_dB_blockR = AudioStream_F32::allocate_f32();
+		audio_block_f32_t *gain_blockL = AudioStream_F32::allocate_f32();
+		audio_block_f32_t *gain_blockR = AudioStream_F32::allocate_f32();
+		// no memory for the audio gain blocks
+		if ( !audio_level_dB_blockL || !audio_level_dB_blockR || !gain_blockL || !gain_blockL)
+		{
+			if (audio_level_dB_blockL) AudioStream_F32::release(audio_level_dB_blockL);
+			if (audio_level_dB_blockR) AudioStream_F32::release(audio_level_dB_blockR);
+			if (gain_blockL) AudioStream_F32::release(gain_blockL);
+			if (gain_blockR) AudioStream_F32::release(gain_blockR);
+			AudioStream_F32::release(blockL);
+			AudioStream_F32::release(blockR);
+			return;
+		}
+		// apply a high-pass filter to get rid of the DC offset
+		if (use_HP_prefilter)
+		{
+			arm_biquad_cascade_df1_f32(&hp_filt_structL, blockL->data, blockL->data, blockL->length);
+			arm_biquad_cascade_df1_f32(&hp_filt_structR, blockR->data, blockR->data, blockR->length);
+		}
+		// apply the pre-gain...a negative gain value will disable
+		if (pre_gain > 0.0f) 
+		{
+			arm_scale_f32(blockL->data, pre_gain, blockL->data, blockL->length); // use ARM DSP for speed!
+			arm_scale_f32(blockR->data, pre_gain, blockR->data, blockR->length); 
+		}			
+		// Side chain processing
+		switch (sidechainMode)
+		{
+			case COMP_SIDECHAIN_SRC_LR:			// l + r separate
+				calcAudioLevel_dB(blockL, audio_level_dB_blockL);
+				calcAudioLevel_dB(blockR, audio_level_dB_blockR);
+				calcGain(audio_level_dB_blockL, gain_blockL);
+				calcGain(audio_level_dB_blockR, gain_blockR);
+				arm_mult_f32(blockL->data, gain_blockL->data, blockL->data, blockL->length);
+				arm_mult_f32(blockR->data, gain_blockR->data, blockR->data, blockR->length);
+				break;
+			case COMP_SIDECHAIN_SRC_LRSUM:		// l + r sum / 2
+				arm_add_f32(blockL->data, blockR->data, audio_level_dB_blockL->data, audio_level_dB_blockL->length); // L+R -> db_L
+				arm_scale_f32(audio_level_dB_blockL->data, 0.5f, audio_level_dB_blockL->data, audio_level_dB_blockL->length); // L+R / 2
+				calcAudioLevel_dB(audio_level_dB_blockL, audio_level_dB_blockL); // chn L used for L&R
+				calcGain(audio_level_dB_blockL, gain_blockL);
+				arm_mult_f32(blockL->data, gain_blockL->data, blockL->data, blockL->length);
+				arm_mult_f32(blockR->data, gain_blockL->data, blockR->data, blockR->length);				
+				break;
+			default:
+				break;
+		}
+		if (post_gain > 0.0f) 
+		{
+			arm_scale_f32(blockL->data, post_gain, blockL->data, blockL->length); // use ARM DSP for speed!
+			arm_scale_f32(blockR->data, post_gain, blockR->data, blockR->length); 
+		}	
+		// transmit the block and release memory
+		AudioStream_F32::transmit(blockL, 0);
+		AudioStream_F32::transmit(blockR, 1);
+		AudioStream_F32::release(blockL);
+		AudioStream_F32::release(blockR);		
+		AudioStream_F32::release(gain_blockL);
+		AudioStream_F32::release(gain_blockR);
+		AudioStream_F32::release(audio_level_dB_blockL);
+		AudioStream_F32::release(audio_level_dB_blockR);
+	}
+
+	// Here's the method that estimates the level of the audio (in dB)
+	// It squares the signal and low-pass filters to get a time-averaged
+	// signal power.  It then
+	void calcAudioLevel_dB(audio_block_f32_t *wav_block, audio_block_f32_t *level_dB_block)
+	{
+		// calculate the instantaneous signal power (square the signal)
+		audio_block_f32_t *wav_pow_block = AudioStream_F32::allocate_f32();
+		arm_mult_f32(wav_block->data, wav_block->data, wav_pow_block->data, wav_block->length);
+
+		// low-pass filter and convert to dB
+		float c1 = level_lp_const, c2 = 1.0f - c1; // prepare constants
+		for (int i = 0; i < wav_pow_block->length; i++)
+		{
+			// first-order low-pass filter to get a running estimate of the average power
+			wav_pow_block->data[i] = c1 * prev_level_lp_pow + c2 * wav_pow_block->data[i];
+
+			// save the state of the first-order low-pass filter
+			prev_level_lp_pow = wav_pow_block->data[i];
+
+			// now convert the signal power to dB (but not yet multiplied by 10.0)
+			level_dB_block->data[i] = log10f_approx(wav_pow_block->data[i]);
+		}
+
+		// limit the amount that the state of the smoothing filter can go toward negative infinity
+		if (prev_level_lp_pow < (1.0E-13))
+			prev_level_lp_pow = 1.0E-13; // never go less than -130 dBFS
+
+		// scale the wav_pow_block by 10.0 to complete the conversion to dB
+		arm_scale_f32(level_dB_block->data, 10.0f, level_dB_block->data, level_dB_block->length); // use ARM DSP for speed!
+
+		// release memory and return
+		AudioStream_F32::release(wav_pow_block);
+		return; // output is passed through level_dB_block
+	}
+
+	// This method computes the desired gain from the compressor, given an estimate
+	// of the signal level (in dB)
+	void calcGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *gain_block)
+	{
+		// first, calculate the instantaneous target gain based on the compression ratio
+		audio_block_f32_t *inst_targ_gain_dB_block = AudioStream_F32::allocate_f32();
+		calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block);
+
+		// second, smooth in time (attack and release) by stepping through each sample
+		audio_block_f32_t *gain_dB_block = AudioStream_F32::allocate_f32();
+		calcSmoothedGain_dB(inst_targ_gain_dB_block, gain_dB_block);
+
+		// finally, convert from dB to linear gain: gain = 10^(gain_dB/20);  (ie this takes care of the sqrt, too!)
+		arm_scale_f32(gain_dB_block->data, 1.0f / 20.0f, gain_dB_block->data, gain_dB_block->length); // divide by 20
+		for (int i = 0; i < gain_dB_block->length; i++)
+			gain_block->data[i] = pow10f(gain_dB_block->data[i]); // do the 10^(x)
+
+		// release memory and return
+		AudioStream_F32::release(gain_dB_block);
+		AudioStream_F32::release(inst_targ_gain_dB_block);
+		return; // output is passed through gain_block
+	}
+
+	// Compute the instantaneous desired gain, including the compression ratio and
+	// threshold for where the comrpession kicks in
+	void calcInstantaneousTargetGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *inst_targ_gain_dB_block)
+	{
+		// how much are we above the compression threshold?
+		audio_block_f32_t *above_thresh_dB_block = AudioStream_F32::allocate_f32();
+		arm_offset_f32(audio_level_dB_block->data,	// CMSIS DSP for "add a constant value to all elements"
+					   -thresh_dBFS,				// this is the value to be added
+					   above_thresh_dB_block->data, // this is the output
+					   audio_level_dB_block->length);
+
+		// scale by the compression ratio...this is what the output level should be (this is our target level)
+		arm_scale_f32(above_thresh_dB_block->data,	 // CMSIS DSP for "multiply all elements by a constant value"
+					  1.0f / comp_ratio,			 // this is the value to be multiplied
+					  inst_targ_gain_dB_block->data, // this is the output
+					  above_thresh_dB_block->length);
+
+		// compute the instantaneous gain...which is the difference between the target level and the original level
+		arm_sub_f32(inst_targ_gain_dB_block->data, // CMSIS DSP for "subtract two vectors element-by-element"
+					above_thresh_dB_block->data,   // this is the vector to be subtracted
+					inst_targ_gain_dB_block->data, // this is the output
+					inst_targ_gain_dB_block->length);
+
+		// limit the target gain to attenuation only (this part of the compressor should not make things louder!)
+		for (int i = 0; i < inst_targ_gain_dB_block->length; i++)
+		{
+			if (inst_targ_gain_dB_block->data[i] > 0.0f)
+				inst_targ_gain_dB_block->data[i] = 0.0f;
+		}
+
+		// release memory before returning
+		AudioStream_F32::release(above_thresh_dB_block);
+		return; // output is passed through inst_targ_gain_dB_block
+	}
+
+	// this method applies the "attack" and "release" constants to smooth the
+	// target gain level through time.
+	void calcSmoothedGain_dB(audio_block_f32_t *inst_targ_gain_dB_block, audio_block_f32_t *gain_dB_block)
+	{
+		float32_t gain_dB;
+		float32_t one_minus_attack_const = 1.0f - attack_const;
+		float32_t one_minus_release_const = 1.0f - release_const;
+		for (int i = 0; i < inst_targ_gain_dB_block->length; i++)
+		{
+			gain_dB = inst_targ_gain_dB_block->data[i];
+
+			// smooth the gain using the attack or release constants
+			if (gain_dB < prev_gain_dB)
+			{ // are we in the attack phase?
+				gain_dB_block->data[i] = attack_const * prev_gain_dB + one_minus_attack_const * gain_dB;
+			}
+			else
+			{ // or, we're in the release phase
+				gain_dB_block->data[i] = release_const * prev_gain_dB + one_minus_release_const * gain_dB;
+			}
+
+			// save value for the next time through this loop
+			prev_gain_dB = gain_dB_block->data[i];
+		}
+
+		// return
+		return; // the output here is gain_block
+	}
+
+	// methods to set parameters of this module
+	void resetStates(void)
+	{
+		prev_level_lp_pow = 1.0f;
+		prev_gain_dB = 0.0f;
+
+		// initialize the HP filter.  (This also resets the filter states,)
+		arm_biquad_cascade_df1_init_f32(&hp_filt_structL, hp_nstages, hp_coeff, hp_stateL);
+		arm_biquad_cascade_df1_init_f32(&hp_filt_structR, hp_nstages, hp_coeff, hp_stateR);
+	}
+	void setPreGain(float g) { pre_gain = g; }
+	void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0f, gain_dB / 20.0f)); }
+	void setPostGain(float g) { post_gain = g; }
+	void setPostGain_dB(float gain_dB) { setPostGain(pow(10.0f, gain_dB / 20.0f)); }	
+	void setCompressionRatio(float cr)
+	{
+		comp_ratio = max(0.001f, cr); // limit to positive values
+		updateThresholdAndCompRatioConstants();
+	}
+	void setAttack_sec(float a)
+	{
+		attack_sec = a;
+		attack_const = expf(-1.0f / (attack_sec * fs_Hz)); // expf() is much faster than exp()
+
+		// also update the time constant for the envelope extraction
+		setLevelTimeConst_sec(min(attack_sec, release_sec) / 5.0f); // make the level time-constant one-fifth the gain time constants
+	}
+	void setRelease_sec(float r)
+	{
+		release_sec = r;
+		release_const = expf(-1.0f / (release_sec * fs_Hz)); // expf() is much faster than exp()
+
+		// also update the time constant for the envelope extraction
+		setLevelTimeConst_sec(min(attack_sec, release_sec) / 5.0f); // make the level time-constant one-fifth the gain time constants
+	}
+	void setLevelTimeConst_sec(float t_sec)
+	{
+		const float min_t_sec = 0.002f; // this is the minimum allowed value
+		level_lp_sec = max(min_t_sec, t_sec);
+		level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); // expf() is much faster than exp()
+	}
+	void setThresh_dBFS(float val)
+	{
+		thresh_dBFS = val;
+		setThreshPow(pow(10.0f, thresh_dBFS / 10.0f));
+	}
+	void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
+
+	// methods to return information about this module
+	float32_t getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain); }
+	float32_t getAttack_sec(void) { return attack_sec; }
+	float32_t getRelease_sec(void) { return release_sec; }
+	float32_t getLevelTimeConst_sec(void) { return level_lp_sec; }
+	float32_t getThresh_dBFS(void) { return thresh_dBFS; }
+	float32_t getCompressionRatio(void) { return comp_ratio; }
+	float32_t getCurrentLevel_dBFS(void) { return 10.0 * log10f_approx(prev_level_lp_pow); }
+	float32_t getCurrentGain_dB(void) { return prev_gain_dB; }
+
+	void setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[])
+	{
+		// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+		// Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
+		hp_coeff[0] = b[0];
+		hp_coeff[1] = b[1];
+		hp_coeff[2] = b[2]; // here are the matlab "b" coefficients
+		hp_coeff[3] = -a[1];
+		hp_coeff[4] = -a[2]; // the DSP needs the "a" terms to have opposite sign vs Matlab
+	}
+    bool bypass_get(void) {return bp;}
+    void bypass_set(bool state) {bp = state;}
+    bool bypass_tgl(void) 
+    {
+        bp ^= 1; 
+        return bp;
+    }
+	void setSideChainMode(sideChainMode_t newMode) {sidechainMode = newMode;}
+private:
+	// state-related variables
+	audio_block_f32_t *inputQueueArray_f32[2]; // memory pointer for the input to this module
+	float32_t prev_level_lp_pow = 1.0f;
+	float32_t prev_gain_dB = 0.0f; // last gain^2 used
+	float32_t fs_Hz = AUDIO_SAMPLE_RATE_EXACT;
+	bool bp = true; // bypass flag
+	sideChainMode_t sidechainMode = COMP_SIDECHAIN_SRC_LRSUM;
+	// HP filter state-related variables
+	arm_biquad_casd_df1_inst_f32 hp_filt_structL;
+	arm_biquad_casd_df1_inst_f32 hp_filt_structR;
+	static const uint8_t hp_nstages = 1;
+	float32_t hp_coeff[5 * hp_nstages] = {1.0f, 0.0f, 0.0f, 0.0f, 0.0f}; // no filtering. actual filter coeff set later
+	float32_t hp_stateL[4 * hp_nstages];
+	float32_t hp_stateR[4 * hp_nstages];
+	void setHPFilterCoeff(void)
+	{
+		// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+		// Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
+		const float32_t b[] = {9.979871156751189e-01,    -1.995974231350238e+00, 9.979871156751189e-01}; // from Matlab
+		const float32_t a[] = {1.000000000000000e+00,    -1.995970179642828e+00,    9.959782830576472e-01}; // from Matlab
+		setHPFilterCoeff_N2IIR_Matlab((float32_t *)b, (float32_t *)a);
+	}
+
+	// private parameters related to gain calculation
+	float32_t attack_const, release_const, level_lp_const; // used in calcGain().  set by setAttack_sec() and setRelease_sec();
+	float32_t comp_ratio_const, thresh_pow_FS_wCR;		   // used in calcGain();  set in updateThresholdAndCompRatioConstants()
+	void updateThresholdAndCompRatioConstants(void)
+	{
+		comp_ratio_const = 1.0f - (1.0f / comp_ratio);
+		thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);
+	}
+
+	// settings
+	float32_t attack_sec = 0.002f, release_sec = 0.2f, level_lp_sec;
+	float32_t thresh_dBFS = 0.0f;	// threshold for compression, relative to digital full scale
+	float32_t thresh_pow_FS = 1.0f; // same as above, but not in dB
+	void setThreshPow(float t_pow)
+	{
+		thresh_pow_FS = t_pow;
+		updateThresholdAndCompRatioConstants();
+	}
+	float32_t comp_ratio = 1.0f; // compression ratio
+	float32_t pre_gain = -1.0f;	// gain to apply before the compression.  negative value disables
+	float32_t post_gain = -1.0f;
+	boolean use_HP_prefilter;
+
+	// Accelerate the powf(10.0,x) function
+	static float32_t pow10f(float x)
+	{
+		// return powf(10.0f,x)   //standard, but slower
+		return expf(2.302585092994f * x); // faster:  exp(log(10.0f)*x)
+	}
+
+	// Accelerate the log10f(x)  function?
+	static float32_t log10f_approx(float x)
+	{
+		// return log10f(x);   //standard, but slower
+		return log2f_approx(x) * 0.3010299956639812f; // faster:  log2(x)/log2(10)
+	}
+
+	/* ----------------------------------------------------------------------
+	** Fast approximation to the log2() function.  It uses a two step
+	** process.  First, it decomposes the floating-point number into
+	** a fractional component F and an exponent E.  The fraction component
+	** is used in a polynomial approximation and then the exponent added
+	** to the result.  A 3rd order polynomial is used and the result
+	** when computing db20() is accurate to 7.984884e-003 dB.
+	** ------------------------------------------------------------------- */
+	// https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
+	// float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
+	static float log2f_approx(float X)
+	{
+		// float *C = &log2f_approx_coeff[0];
+		float Y;
+		float F;
+		int E;
+
+		// This is the approximation to log2()
+		F = frexpf(fabsf(X), &E);
+		//  Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
+		// Y = *C++;
+		Y = 1.23149591368684f;
+		Y *= F;
+		// Y += (*C++);
+		Y += -4.11852516267426f;
+		Y *= F;
+		// Y += (*C++);
+		Y += 6.02197014179219f;
+		Y *= F;
+		// Y += (*C++);
+		Y += -3.13396450166353f;
+		Y += E;
+
+		return (Y);
+	}
+};
+
+#endif

src/effect_delaystereo_F32.cpp

@@ -23,17 +23,28 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
  * SOFTWARE.
  */
-#include "effect_delaystereo.h"
+#include "effect_delaystereo_F32.h"
 
 #define TREBLE_LOSS_FREQ    (0.20f)
 #define BASS_LOSS_FREQ      (0.05f)
 #define BASS_FREQ      		(0.15f)
 
-
+extern uint8_t external_psram_size;
 
 AudioEffectDelayStereo_F32::AudioEffectDelayStereo_F32(uint32_t dly_range_ms, bool use_psram) : AudioStream_F32(2, inputQueueArray)
 {
-	psram_mode = use_psram;
+	begin(dly_range_ms, use_psram);
+}
+
+void AudioEffectDelayStereo_F32::begin(uint32_t dly_range_ms, bool use_psram)
+{
+	// failsafe if psram is required but not found
+	// limit the delay time to 500ms (88200 bytes at 44.1kHz)
+	if (psram_mode && external_psram_size == 0)
+	{
+		psram_mode = false;
+		if (dly_range_ms > 500) dly_range_ms = 500;
+	}	
 	bool memOk = true;
 	dly_length = ((float32_t)(dly_range_ms+500)/1000.0f) * AUDIO_SAMPLE_RATE_EXACT;	
 	if (!dly0a.init(dly_length, use_psram)) memOk = false;
@@ -55,11 +66,7 @@ void AudioEffectDelayStereo_F32::update()
 	if (!initialized) return;
 	if (!memsetup_done)
 	{
-		dly0a.reset();
+		memsetup_done = memCleanup();
-		dly0b.reset();
-		dly1a.reset();
-		dly1b.reset();
-		memsetup_done = true;
 		return;		
 	}
 
@@ -71,18 +78,17 @@ void AudioEffectDelayStereo_F32::update()
 
     if (bp)
     {
-		if (!cleanup_done)
+		// mem cleanup not required in TRAILS mode
+		if (!cleanup_done && bp_mode != BYPASS_MODE_TRAILS)
 		{
-			dly0a.reset();
+			cleanup_done = memCleanup();
-			dly0b.reset();
-			dly1a.reset();
-			dly1b.reset();
-			cleanup_done = true;
 			tap_active = false;	// reset tap tempo
 			tap_counter = 0;
 		}
-        if (dry_gain > 0.0f) 		// if dry/wet mixer is used
+		if (infinite) freeze(false);
+		switch(bp_mode)
 		{
+			case BYPASS_MODE_PASS:
 				blockL = AudioStream_F32::receiveReadOnly_f32(0);
 				blockR = AudioStream_F32::receiveReadOnly_f32(1);
 				if (!blockL || !blockR) 
@@ -96,7 +102,8 @@ void AudioEffectDelayStereo_F32::update()
 				AudioStream_F32::release(blockL);
 				AudioStream_F32::release(blockR);
 				return;
-		}
+				break;
+			case BYPASS_MODE_OFF:
 				blockL = AudioStream_F32::allocate_f32();
 				if (!blockL) return;
 				memset(&blockL->data[0], 0, blockL->length*sizeof(float32_t));
@@ -104,8 +111,16 @@ void AudioEffectDelayStereo_F32::update()
 				AudioStream_F32::transmit(blockL, 1);
 				AudioStream_F32::release(blockL);	
 				return;
+				break;
+			case BYPASS_MODE_TRAILS:
+				inputGainSet = 0.0f;
+				tap_active = false;	// reset tap tempo
+				tap_counter = 0;
+				break;
+			default:
+				break;
+		}
 	}
-	cleanup_done = false;
 	blockL = AudioStream_F32::receiveWritable_f32(0);
 	blockR = AudioStream_F32::receiveWritable_f32(1);
 	if (!blockL || !blockR)
@@ -116,8 +131,11 @@ void AudioEffectDelayStereo_F32::update()
 			AudioStream_F32::release(blockR);
 		return;
 	}
+	cleanup_done = false;
+
 	for (i=0; i < blockL->length; i++) 
     {  
+		inputGain += (inputGainSet - inputGain) * 0.25f;
 		// tap tempo
 		if (tap_active)
 		{
@@ -138,7 +156,7 @@ void AudioEffectDelayStereo_F32::update()
 			dly_time -= dly_time_step;
 			if (dly_time < dly_time_set) dly_time = dly_time_set;
 		}
-		// lowpass the dely time
+		// lowpass the delay time
 		acc1 = dly_time - dly_time_flt;
 		dly_time_flt += acc1 * 0.1f;
 		dly_time = dly_time_flt;
@@ -165,12 +183,10 @@ void AudioEffectDelayStereo_F32::update()
 		acc2 = (float32_t)dly_length - 1.0f - (dly_time + mod_fr[3]);
 		if (acc2 < 0.0f) mod_fr[3] += acc2;		
 
-		float32_t idx = dly_time + mod_fr[0];
-
 		acc1 = dly0b.getTapHermite(dly_time+mod_fr[0]);
 		outR = acc1 * 0.6f;
 		acc1 = flt0R.process(acc1) * feedb;
-		acc1 += blockR->data[i] * input_attn;
+		acc1 += blockR->data[i] * inputGain;
 		acc1 = flt1R.process(acc1);
 		acc2 = dly0a.getTapHermite(dly_time+mod_fr[1]);
 		dly0b.write_toOffset(acc2, 0);
@@ -180,7 +196,7 @@ void AudioEffectDelayStereo_F32::update()
 		acc1 = dly1b.getTapHermite(dly_time+mod_fr[2]);
 		outR += acc1 * 0.6f;
 		acc1 = flt0L.process(acc1) * feedb;
-		acc1 += blockL->data[i] * input_attn;
+		acc1 += blockL->data[i] * inputGain;
 		acc1 = flt1L.process(acc1);
 		acc2 = dly1a.getTapHermite(dly_time+mod_fr[3]);
 		dly1b.write_toOffset(acc2, 0);
@@ -200,3 +216,115 @@ void AudioEffectDelayStereo_F32::update()
 	AudioStream_F32::release(blockL);
 	AudioStream_F32::release(blockR);
 }
+void AudioEffectDelayStereo_F32::freeze(bool state)
+{
+	if (infinite == state) return;
+	infinite = state;
+	if (state)
+	{
+		feedb_tmp = feedb;      // store the settings
+		inputGain_tmp = inputGainSet;
+		bassCut_k_tmp = bassCut_k;
+		trebleCut_k_tmp = trebleCut_k;
+		__disable_irq();
+		feedb = 1.0f; // infinite echo                                    
+		inputGainSet = freeze_ingain;
+		__enable_irq();
+	}
+	else
+	{
+		__disable_irq();
+		feedb = feedb_tmp;
+		inputGainSet = inputGain_tmp;
+		bassCut_k = bassCut_k_tmp;
+		trebleCut_k = trebleCut_k_tmp;
+		__enable_irq();
+	}
+}
+
+/**
+ * @brief Partial memory clear
+ * 	Clearing all the delay buffers at once, esp. if 
+ * 	the PSRAM is used takes too long for the audio ISR.
+ *  Hence the buffer clear is done in configurable portions
+ *  spread over a few audio update routines.
+ * 
+ * @return true 	Memory clean is complete
+ * @return false 	Memory clean still in progress
+ */
+bool AudioEffectDelayStereo_F32::memCleanup()
+{
+	static uint8_t dlyIdx = 0;
+	bool result = false;
+	if (dlyIdx == 0) // value 0 is used to reset the addr
+	{
+		memCleanupStart = 0;
+		memCleanupEnd = memCleanupStep;
+		flt0L.reset();
+		flt0R.reset();
+		flt1L.reset();
+		flt1R.reset();
+		dlyIdx = 1;
+	}
+	if (memCleanupEnd > dly_length) 		// last segment
+	{
+		memCleanupEnd = dly_length;
+		result = true;
+	}
+	switch(dlyIdx)
+	{
+		case 1:
+			dly0a.reset(memCleanupStart, memCleanupEnd);
+			memCleanupStart = memCleanupEnd;
+			memCleanupEnd += memCleanupStep;
+			if (result) // if done, reset the mem addr
+			{
+				memCleanupStart = 0;
+				memCleanupEnd = memCleanupStep;
+				dlyIdx = 2;
+				result = false;
+			}
+			break;
+		case 2:
+			dly0b.reset(memCleanupStart, memCleanupEnd);
+			memCleanupStart = memCleanupEnd;
+			memCleanupEnd += memCleanupStep;			
+			if (result) // if done, reset the mem addr
+			{
+				memCleanupStart = 0;
+				memCleanupEnd = memCleanupStep;
+				dlyIdx = 3;
+				result = false;
+			}		
+			break;
+		case 3:
+			dly1a.reset(memCleanupStart, memCleanupEnd);
+			memCleanupStart = memCleanupEnd;
+			memCleanupEnd += memCleanupStep;			
+			if (result) // if done, reset the mem addr
+			{
+				memCleanupStart = 0;
+				memCleanupEnd = memCleanupStep;
+				dlyIdx = 4;
+				result = false;
+			}	
+			break;
+		case 4:
+			dly1b.reset(memCleanupStart, memCleanupEnd);
+			memCleanupStart = memCleanupEnd;
+			memCleanupEnd += memCleanupStep;			
+			if (result) // if done, reset the mem addr
+			{
+				dlyIdx = 0;
+				result = true;
+			}				
+			break;
+		default:
+			dlyIdx = 0; // cleanup done, reset the dly line idx
+			result = false;
+			break;
+	}
+
+	
+	return result;
+}

src/effect_delaystereo_F32.h

@@ -40,16 +40,18 @@ public:
 	~AudioEffectDelayStereo_F32(){};
 	virtual void update();
 	/**
-	 * @brief delay time
+	 * @brief set the delay time
 	 * 
-	 * @param t scaled to 0.0f-1.0f range
+	 * @param t delay time scaled to range 0.0 to 1.0
+	 * @param force bypass the smoothing, immediate change
 	 */
-	void time(float t)
+	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();
 	}
@@ -73,13 +75,15 @@ public:
 	 */
 	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;
-        input_attn = attn;
+        inputGainSet = attn;
         __enable_irq();
     }
 	/**
@@ -116,7 +120,9 @@ public:
 	 */
     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();
@@ -141,8 +147,10 @@ public:
 	 */
     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();
@@ -201,15 +209,45 @@ public:
 		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) {bp = state;}
+    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) 
     {
-        bp ^= 1; 
+		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;
@@ -243,7 +281,6 @@ public:
 		}
 		return tempo_ticks;
 	} 
-
 private:
 	audio_block_f32_t *inputQueueArray[2];
 
@@ -264,15 +301,19 @@ private:
 	AudioBasicLfo lfo = AudioBasicLfo(0.0f, lfo_ampl);
 	bool psram_mode;
 	bool memsetup_done = false;
-	bool bp = 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 input_attn = 1.0f;
+	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;
@@ -282,11 +323,24 @@ private:
 	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_

src/effect_gainStereo_F32.h

@@ -1,12 +1,20 @@
-/*
+/**
- * AudioEffectGain_F32
+ * @file effect_gainStereo_F32.h
+ * @author Piotr Zapart
+ * @brief Stereo volume + pan control
+ * @version 0.1
+ * @date 2024-03-20
  * 
- * Created: Chip Audette, November 2016
+ * @copyright Copyright (c) 2024 www.hexefx.com
- * Purpose; Apply digital gain to the audio data.  Assumes floating-point data.
  * 
- * This processes a single stream fo audio data (ie, it is mono)
+ * 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, 
- * MIT License.  use at your own risk.
+ * 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 _AudioEffectGainStereo_F32_h
@@ -14,14 +22,13 @@
 
 #include <arm_math.h> //ARM DSP extensions.  for speed!
 #include <AudioStream_F32.h>
+#include <basic_components.h>
 
 class AudioEffectGainStereo_F32 : public AudioStream_F32
 {
 public:
-	// constructor
+	AudioEffectGainStereo_F32(void) : AudioStream_F32(2, inputQueueArray_f32) { setPan(0.0f);};
-	AudioEffectGainStereo_F32(void) : AudioStream_F32(2, inputQueueArray_f32){};
+	AudioEffectGainStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32){setPan(0.0f);};
-	AudioEffectGainStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32){};
-
 	void update(void)
 	{
 		audio_block_f32_t *blockL, *blockR;
@@ -35,29 +42,59 @@ public:
 				AudioStream_F32::release(blockR);
 			return;
 		}
-		arm_scale_f32(blockL->data, gain, blockL->data, blockL->length); // use ARM DSP for speed!
+		gainL += (gainLset - gainL) * 0.25f;
-		arm_scale_f32(blockR->data, gain, blockR->data, blockR->length);
+		gainR += (gainRset - gainR) * 0.25f;
+
+		arm_scale_f32(blockL->data, gainL, blockL->data, blockL->length); // use ARM DSP for speed!
+		arm_scale_f32(blockR->data, gainR, blockR->data, blockR->length);
 		AudioStream_F32::transmit(blockL, 0);
 		AudioStream_F32::transmit(blockR, 1);
 		AudioStream_F32::release(blockL);
 		AudioStream_F32::release(blockR);
 	}
+	void setGain(float g) 
+	{ 
+		float32_t gL, gR;
+		gain = g; 
+		gL = panL * gain;
+		gR = panR * gain;
+		__disable_irq();
+		gainLset = gL;
+		gainRset = gR;
+		__enable_irq();
 
-	// methods to set parameters of this module
+	}
-	void setGain(float g) { gain = g; }
 	void setGain_dB(float gain_dB)
 	{
-		float gain = pow(10.0, gain_dB / 20.0);
+		float gain = powf(10.0f, gain_dB / 20.0f);
 		setGain(gain);
 	}
-
-	// methods to return information about this module
 	float getGain(void) { return gain; }
 	float getGain_dB(void) { return 20.0 * log10(gain); }
 
+	void setPan(float32_t p)
+	{
+		float32_t tmp, gL, gR;
+		pan = constrain(p, -1.0f, 1.0f);
+		tmp = (pan + 1.0f) * 0.5f; // map to 0..1
+		
+		mix_pwr(tmp, &panR, &panL);
+
+
+		gL = panL * gain;
+		gR = panR * gain;	
+		__disable_irq();
+		gainLset = gL;
+		gainRset = gR;
+		__enable_irq();
+
+	}
+	float32_t getPan() { return pan;}
 private:
 	audio_block_f32_t *inputQueueArray_f32[2]; // memory pointer for the input to this module
-	float gain = 1.0f;						   // default value
+	float32_t gain = 1.0f;						   // default value
+	float32_t gainL, gainR, gainLset, gainRset;
+	float32_t pan, panL, panR;
 };
 
 #endif

src/effect_guitarBooster_F32.cpp

@@ -0,0 +1,339 @@
+/**
+ * @file effect_guitarBooster_F32.cpp
+ * @author Piotr Zapart
+ * @brief Oversampled Waveshaper based overdrive effect
+ * 			Stereo IO and bypass, the processing is mono
+ * @version 0.1
+ * @date 2024-03-20
+ * 
+ * @copyright Copyright (c) 2024
+ * 
+ */
+#include "effect_guitarBooster_F32.h"
+
+void AudioEffectGuitarBooster_F32::update()
+{
+	audio_block_f32_t *blockL, *blockR;
+	uint16_t i;
+	float32_t sampleWet, sampleDry;
+	float32_t *samplePtr;
+	float32_t _hpPre1_reg;// = hpPre1_reg;
+	float32_t _hpPre2_reg;// = hpPre2_reg;
+	float32_t _lp1_reg;
+	float32_t _lp2_reg;
+	float32_t _hpPost_reg;
+	float32_t _hpPre1_k = hpPre1_k;
+	float32_t _hpPre2_k = hpPre2_k;
+	float32_t _lp1_k = lp1_k;
+	float32_t _lp2_k = lp2_k;
+	float32_t _hpPost_k = hpPost_k;
+	float32_t _gainSet = gainSet;
+	float32_t _gain_hp = gain_hp;
+	float32_t _gain = gain;
+	float32_t _levelSet = levelSet;
+	float32_t _level = level;
+
+	const uint32_t blockLenInterpolated = upsample_k * AUDIO_BLOCK_SAMPLES;
+
+	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;
+	}
+	_hpPre1_reg = hpPre1_reg;
+	_hpPre2_reg = hpPre2_reg;
+	_lp1_reg = lp1_reg;
+	_lp2_reg = lp2_reg;
+	_hpPost_reg = hpPost_reg;
+
+	arm_add_f32(blockL->data, blockR->data, blockL->data, blockL->length); // add two channels
+	arm_fir_interpolate_f32(&interpolator, blockL->data, blockInterpolated, blockL->length);
+	samplePtr = blockInterpolated;
+	for (i = 0; i < blockLenInterpolated; i++)
+	{
+		sampleWet = *samplePtr;
+		sampleDry  = sampleWet;
+		_gain += (_gainSet - _gain) * 0.25f;
+		// octave up
+		if (octave) sampleWet = 2.0f * fabsf(sampleWet) - 1.0f;	
+		// input high pass
+		sampleWet -= (_hpPre1_reg += (sampleWet - _hpPre1_reg) * _hpPre1_k);
+		sampleWet -= (_hpPre2_reg += (sampleWet - _hpPre2_reg) * _hpPre2_k);
+		sampleWet *= _gain * _gain_hp;
+		// waveshaper
+		sampleWet = arm_linear_interp_f32(&waveshaper, sampleWet + DCbias) * -1.0f;
+		// lowpass 
+		sampleWet = (_lp1_reg += (sampleWet - _lp1_reg) * _lp1_k);
+		sampleWet = (_lp2_reg += (sampleWet - _lp2_reg) * _lp2_k);    
+		// output highpass
+		sampleWet -= (_hpPost_reg += (sampleWet - _hpPost_reg) * _hpPost_k);
+		_level += (_levelSet - _level) * 0.25f;
+		*samplePtr++ = (sampleWet * wetGain + sampleDry * dryGain) * level;
+	}
+	arm_fir_decimate_f32(&decimator, blockInterpolated, blockL->data, blockLenInterpolated);
+
+	hpPre1_reg = _hpPre1_reg;
+	hpPre2_reg = _hpPre2_reg;
+	lp1_reg = _lp1_reg;
+	lp2_reg = _lp2_reg;
+	hpPost_reg = _hpPost_reg;
+	gain = _gain;
+	level = _level;
+
+	AudioStream_F32::transmit(blockL, 0); // send blockL on both output channels
+	AudioStream_F32::transmit(blockL, 1);
+	AudioStream_F32::release(blockL);
+	AudioStream_F32::release(blockR);
+}
+
+void AudioEffectGuitarBooster_F32::bottom(float32_t b)
+{
+	b = constrain(b, 0.0f, 1.0f);
+	gain_hp = 1.0f + b * 2.0f;
+	float32_t hp = map(b, 0.0f, 1.0f, GBOOST_BOTTOM_MAXF, GBOOST_BOTTOM_MINF);
+	hp = omega(hp);
+	__disable_irq();
+	hpPre1_k = hp;
+	hpPre2_k = hpPre1_k;
+	__enable_irq();
+}
+
+void AudioEffectGuitarBooster_F32::tone(float32_t t)
+{
+	t = constrain(t, 0.0f, 1.0f);
+	t = t * t;
+	float32_t lp = map(t, 0.0f, 1.0f, GBOOST_TONE_MINF, GBOOST_TONE_MAXF);
+
+	lp = omega(lp);
+	__disable_irq();
+	lp1_k = lp;
+	__enable_irq();
+}
+
+float32_t AudioEffectGuitarBooster_F32::driveWaveform[2001]=
+{
+	-0.34169694, -0.34162512, -0.34155323, -0.34148127, -0.34140924, -0.34133714, -0.34126496, -0.34119272, -0.34112041, -0.34104802, 
+	-0.34097557, -0.34090304, -0.34083044, -0.34075777, -0.34068503, -0.34061222, -0.34053933, -0.34046637, -0.34039334, -0.34032024, 
+	-0.34024707, -0.34017382, -0.34010050, -0.34002711, -0.33995364, -0.33988010, -0.33980649, -0.33973281, -0.33965905, -0.33958521, 
+	-0.33951131, -0.33943733, -0.33936327, -0.33928914, -0.33921494, -0.33914066, -0.33906631, -0.33899188, -0.33891738, -0.33884280, 
+	-0.33876814, -0.33869341, -0.33861861, -0.33854373, -0.33846877, -0.33839374, -0.33831863, -0.33824344, -0.33816818, -0.33809284, 
+	-0.33801743, -0.33794193, -0.33786636, -0.33779072, -0.33771499, -0.33763919, -0.33756331, -0.33748735, -0.33741131, -0.33733520, 
+	-0.33725901, -0.33718273, -0.33710638, -0.33702995, -0.33695345, -0.33687686, -0.33680019, -0.33672344, -0.33664662, -0.33656971, 
+	-0.33649273, -0.33641566, -0.33633851, -0.33626128, -0.33618398, -0.33610659, -0.33602912, -0.33595157, -0.33587393, -0.33579622, 
+	-0.33571842, -0.33564054, -0.33556258, -0.33548454, -0.33540642, -0.33532821, -0.33524992, -0.33517155, -0.33509309, -0.33501455, 
+	-0.33493593, -0.33485722, -0.33477843, -0.33469956, -0.33462060, -0.33454156, -0.33446243, -0.33438322, -0.33430393, -0.33422454, 
+	-0.33414508, -0.33406553, -0.33398589, -0.33390617, -0.33382636, -0.33374646, -0.33366648, -0.33358641, -0.33350626, -0.33342602, 
+	-0.33334569, -0.33326527, -0.33318477, -0.33310418, -0.33302350, -0.33294274, -0.33286188, -0.33278094, -0.33269991, -0.33261879, 
+	-0.33253758, -0.33245628, -0.33237490, -0.33229342, -0.33221185, -0.33213020, -0.33204845, -0.33196662, -0.33188469, -0.33180267, 
+	-0.33172056, -0.33163836, -0.33155607, -0.33147369, -0.33139122, -0.33130865, -0.33122600, -0.33114325, -0.33106041, -0.33097747, 
+	-0.33089444, -0.33081132, -0.33072811, -0.33064480, -0.33056140, -0.33047791, -0.33039432, -0.33031064, -0.33022686, -0.33014299, 
+	-0.33005902, -0.32997496, -0.32989080, -0.32980655, -0.32972220, -0.32963776, -0.32955322, -0.32946858, -0.32938385, -0.32929901, 
+	-0.32921409, -0.32912906, -0.32904394, -0.32895872, -0.32887340, -0.32878799, -0.32870247, -0.32861686, -0.32853115, -0.32844534, 
+	-0.32835943, -0.32827342, -0.32818731, -0.32810110, -0.32801479, -0.32792838, -0.32784187, -0.32775526, -0.32766855, -0.32758174, 
+	-0.32749482, -0.32740781, -0.32732069, -0.32723347, -0.32714614, -0.32705872, -0.32697119, -0.32688356, -0.32679582, -0.32670798, 
+	-0.32662004, -0.32653199, -0.32644384, -0.32635558, -0.32626722, -0.32617876, -0.32609019, -0.32600151, -0.32591273, -0.32582384, 
+	-0.32573484, -0.32564574, -0.32555653, -0.32546722, -0.32537779, -0.32528826, -0.32519862, -0.32510888, -0.32501902, -0.32492906, 
+	-0.32483899, -0.32474880, -0.32465851, -0.32456811, -0.32447760, -0.32438698, -0.32429625, -0.32420541, -0.32411446, -0.32402339, 
+	-0.32393222, -0.32384093, -0.32374953, -0.32365802, -0.32356639, -0.32347466, -0.32338281, -0.32329084, -0.32319877, -0.32310657, 
+	-0.32301427, -0.32292185, -0.32282932, -0.32273667, -0.32264390, -0.32255102, -0.32245802, -0.32236491, -0.32227168, -0.32217834, 
+	-0.32208487, -0.32199129, -0.32189760, -0.32180378, -0.32170985, -0.32161579, -0.32152162, -0.32142733, -0.32133292, -0.32123839, 
+	-0.32114374, -0.32104898, -0.32095409, -0.32085907, -0.32076394, -0.32066869, -0.32057331, -0.32047782, -0.32038220, -0.32028646, 
+	-0.32019059, -0.32009460, -0.31999849, -0.31990225, -0.31980589, -0.31970941, -0.31961280, -0.31951606, -0.31941920, -0.31932222, 
+	-0.31922511, -0.31912787, -0.31903050, -0.31893301, -0.31883539, -0.31873764, -0.31863977, -0.31854176, -0.31844363, -0.31834537, 
+	-0.31824698, -0.31814845, -0.31804980, -0.31795102, -0.31785211, -0.31775307, -0.31765389, -0.31755458, -0.31745515, -0.31735558, 
+	-0.31725587, -0.31715603, -0.31705606, -0.31695596, -0.31685572, -0.31675535, -0.31665484, -0.31655420, -0.31645342, -0.31635250, 
+	-0.31625145, -0.31615026, -0.31604894, -0.31594748, -0.31584588, -0.31574414, -0.31564226, -0.31554025, -0.31543809, -0.31533580, 
+	-0.31523337, -0.31513079, -0.31502808, -0.31492522, -0.31482222, -0.31471908, -0.31461580, -0.31451237, -0.31440881, -0.31430510, 
+	-0.31420124, -0.31409724, -0.31399310, -0.31388881, -0.31378438, -0.31367980, -0.31357507, -0.31347020, -0.31336518, -0.31326001, 
+	-0.31315470, -0.31304923, -0.31294362, -0.31283786, -0.31273195, -0.31262589, -0.31251968, -0.31241332, -0.31230681, -0.31220015, 
+	-0.31209333, -0.31198637, -0.31187925, -0.31177197, -0.31166455, -0.31155697, -0.31144923, -0.31134134, -0.31123330, -0.31112509, 
+	-0.31101674, -0.31090822, -0.31079955, -0.31069072, -0.31058174, -0.31047259, -0.31036329, -0.31025382, -0.31014420, -0.31003442, 
+	-0.30992447, -0.30981437, -0.30970410, -0.30959367, -0.30948308, -0.30937233, -0.30926141, -0.30915033, -0.30903908, -0.30892767, 
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+	0.79815939, 0.79964936, 0.80113359, 0.80261204, 0.80408472, 0.80555161, 0.80701269, 0.80846796, 0.80991739, 0.81136098, 
+	0.81279872, 0.81423059, 0.81565658, 0.81707668, 0.81849088, 0.81989915, 0.82130150, 0.82269791, 0.82408837, 0.82547286, 
+	0.82685138, 0.82822391, 0.82959044, 0.83095096, 0.83230547, 0.83365394, 0.83499637, 0.83633274, 0.83766305, 0.83898729, 
+	0.84030544, 0.84161750, 0.84292346, 0.84422329, 0.84551701, 0.84680459, 0.84808602, 0.84936131, 0.85063043, 0.85189338, 
+	0.85315014, 0.85440072, 0.85564511, 0.85688329, 0.85811525, 0.85934099, 0.86056050, 0.86177378, 0.86298081, 0.86418159, 
+	0.86537611, 0.86656437, 0.86774635, 0.86892205, 0.87009147, 0.87125460, 0.87241143, 0.87356196, 0.87470618, 0.87584409, 
+	0.87697569, 0.87810096, 0.87921990, 0.88033251, 0.88143879, 0.88253873, 0.88363233, 0.88471958, 0.88580049, 0.88687504, 
+	0.88794324, 0.88900509, 0.89006058, 0.89110971, 0.89215248, 0.89318889, 0.89421893, 0.89524262, 0.89625994, 0.89727090, 
+	0.89827550, 0.89927374, 0.90026561, 0.90125113, 0.90223028, 0.90320309, 0.90416953, 0.90512963, 0.90608338, 0.90703078, 
+	0.90797184, 0.90890656, 0.90983495, 0.91075701, 0.91167274, 0.91258216, 0.91348525, 0.91438204, 0.91527253, 0.91615672, 
+	0.91703461, 0.91790623, 0.91877157, 0.91963064, 0.92048345, 0.92133001, 0.92217033, 0.92300441, 0.92383227, 0.92465392, 
+	0.92546936, 0.92627861, 0.92708167, 0.92787857, 0.92866930, 0.92945388, 0.93023233, 0.93100466, 0.93177087, 0.93253099, 
+	0.93328503, 0.93403299, 0.93477491, 0.93551078, 0.93624063, 0.93696446, 0.93768231, 0.93839417, 0.93910008, 0.93980004, 
+	0.94049407, 0.94118220, 0.94186443, 0.94254079, 0.94321130, 0.94387597, 0.94453483, 0.94518788, 0.94583516, 0.94647669, 
+	0.94711248, 0.94774255, 0.94836693, 0.94898564, 0.94959870, 0.95020613, 0.95080795, 0.95140420, 0.95199488, 0.95258003, 
+	0.95315966, 0.95373381, 0.95430249, 0.95486573, 0.95542356, 0.95597600, 0.95652308, 0.95706482, 0.95760125, 0.95813239, 
+	0.95865827, 0.95917892, 0.95969437, 0.96020464, 0.96070976, 0.96120975, 0.96170465, 0.96219449, 0.96267929, 0.96315908, 
+	0.96363389, 0.96410374, 0.96456868, 0.96502872, 0.96548391, 0.96593425, 0.96637980, 0.96682058, 0.96725661, 0.96768793, 
+	0.96811457, 0.96853656, 0.96895393, 0.96936672, 0.96977495, 0.97017865, 0.97057786, 0.97097261, 0.97136293, 0.97174885, 
+	0.97213040, 0.97250762, 0.97288054, 0.97324919, 0.97361360, 0.97397381, 0.97432985, 0.97468174, 0.97502953, 0.97537324, 
+	0.97571291, 0.97604857, 0.97638025, 0.97670799, 0.97703182, 0.97735177, 0.97766787, 0.97798016, 0.97828867, 0.97859343, 
+	0.97889448, 0.97919184, 0.97948556, 0.97977565, 0.98006216, 0.98034512, 0.98062456, 0.98090052, 0.98117301, 0.98144209, 
+	0.98170778, 0.98197010, 0.98222910, 0.98248481, 0.98273726, 0.98298647, 0.98323249, 0.98347534, 0.98371505, 0.98395166, 
+	0.98418520, 0.98441569, 0.98464318, 0.98486768, 0.98508924, 0.98530788, 0.98552363, 0.98573652, 0.98594659, 0.98615386, 
+	0.98635836, 0.98656013, 0.98675919, 0.98695557, 0.98714930, 0.98734042, 0.98752894, 0.98771490, 0.98789833, 0.98807925, 
+	0.98825770, 0.98843370, 0.98860728, 0.98877847, 0.98894729, 0.98911378, 0.98927795, 0.98943985, 0.98959948, 0.98975689, 
+	0.98991209, 0.99006512, 0.99021599, 0.99036474, 0.99051139, 0.99065597, 0.99079850, 0.99093900, 0.99107750, 0.99121403, 
+	0.99134861, 0.99148127, 0.99161202, 0.99174089, 0.99186791, 0.99199310, 0.99211648, 0.99223808, 0.99235791, 0.99247600, 
+	0.99259238, 0.99270706, 0.99282006, 0.99293142, 0.99304114, 0.99314926, 0.99325579, 0.99336075, 0.99346416, 0.99356605, 
+	0.99366643, 0.99376533, 0.99386276, 0.99395874, 0.99405330, 0.99414644, 0.99423820, 0.99432859, 0.99441763, 0.99450533, 
+	0.99459172, 0.99467682, 0.99476063, 0.99484318, 0.99492449, 0.99500457, 0.99508344, 0.99516112, 0.99523762, 0.99531296, 
+	0.99538716, 0.99546022, 0.99553218, 0.99560304, 0.99567282, 0.99574153, 0.99580919, 0.99587582, 0.99594143, 0.99600603, 
+	0.99606963, 0.99613226, 0.99619393, 0.99625465, 0.99631443, 0.99637328, 0.99643123, 0.99648828, 0.99654445, 0.99659975, 
+	0.99665419, 0.99670779, 0.99676055, 0.99681249, 0.99686362, 0.99691396, 0.99696351, 0.99701229, 0.99706031, 0.99710758, 
+	0.99715410, 0.99719990, 0.99724498, 0.99728936, 0.99733304, 0.99737603, 0.99741834, 0.99745999, 0.99750099, 0.99754134, 
+	0.99758105, 0.99762014, 0.99765861, 0.99769647, 0.99773373, 0.99777041, 0.99780650, 0.99784202, 0.99787698, 0.99791138, 
+	0.99794524, 0.99797856, 0.99801135, 0.99804362, 0.99807538, 0.99810663, 0.99813738, 0.99816764, 0.99819742, 0.99822673, 
+	0.99825556, 0.99828394, 0.99831186, 0.99833934, 0.99836638, 0.99839298, 0.99841916, 0.99844492, 0.99847026, 0.99849520, 
+	0.99851974, 0.99854388, 0.99856764, 0.99859102, 0.99861402, 0.99863665, 0.99865891, 0.99868082, 0.99870237, 0.99872358, 
+	0.99874445, 0.99876497, 0.99878517, 0.99880504, 0.99882459, 0.99884383, 0.99886275, 0.99888137, 0.99889969, 0.99891771, 
+	0.99893544, 0.99895288, 0.99897004, 0.99898692, 0.99900353, 0.99901987, 0.99903594, 0.99905176, 0.99906731, 0.99908262, 
+	0.99909767, 0.99911249, 0.99912706, 0.99914139, 0.99915549, 0.99916936, 0.99918301, 0.99919643, 0.99920964, 0.99922263, 
+	0.99923540, 0.99924797, 0.99926034, 0.99927250, 0.99928447, 0.99929624, 0.99930782, 0.99931921, 0.99933041, 0.99934143, 
+	0.99935227, 0.99936294, 0.99937343, 0.99938375, 0.99939389, 0.99940388, 0.99941370, 0.99942336, 0.99943286, 0.99944221, 
+	0.99945140, 0.99946044, 0.99946934, 0.99947809, 0.99948669, 0.99949516, 0.99950348, 0.99951167, 0.99951973, 0.99952765, 
+	0.99953544, 0.99954311, 0.99955065, 0.99955807, 0.99956536, 0.99957253, 0.99957959, 0.99958653, 0.99959336, 0.99960007, 
+	0.99960668, 0.99961317, 0.99961956, 0.99962585, 0.99963203, 0.99963811, 0.99964409, 0.99964997, 0.99965575, 0.99966144, 
+	0.99966704, 0.99967254, 0.99967796, 0.99968328, 0.99968852, 0.99969367, 0.99969873, 0.99970371, 0.99970861, 0.99971343, 
+	0.99971817, 0.99972283, 0.99972742, 0.99973193, 0.99973636, 0.99974072, 0.99974501, 0.99974923, 0.99975338, 0.99975746, 
+	0.99976148, 0.99976543, 0.99976931, 0.99977313, 0.99977688, 0.99978058, 0.99978421, 0.99978778, 0.99979130, 0.99979475, 
+	0.99979815, 0.99980149, 0.99980478, 0.99980801, 0.99981119, 0.99981432, 0.99981740, 0.99982042, 0.99982339, 0.99982632, 
+	0.99982920, 0.99983203, 0.99983481, 0.99983755, 0.99984024, 0.99984289, 0.99984549, 0.99984805, 0.99985057, 0.99985304, 
+	0.99985548, 0.99985788, 0.99986023, 0.99986255, 0.99986483, 0.99986707, 0.99986927, 0.99987144, 0.99987357, 0.99987567, 
+	0.99987773, 0.99987975, 0.99988175, 0.99988371, 0.99988564, 0.99988753, 0.99988940, 0.99989123, 0.99989304, 0.99989481, 
+	0.99989655
+};

src/effect_guitarBooster_F32.h

@@ -0,0 +1,175 @@
+#ifndef _EFFECT_GUITARBOOSTER_F32_H_
+#define _EFFECT_GUITARBOOSTER_F32_H_
+
+/**
+ * @file effect_guitarBooster_F32.h
+ * @author Piotr Zapart
+ * @brief Oversampled Waveshaper based overdrive effect
+ * 			Stereo IO and bypass, but the processing is mono
+ * @version 0.1
+ * @date 2024-03-20
+ * 
+ * @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/>."
+ */
+
+#include <AudioStream_F32.h>
+#include "basic_DSPutils.h"
+#include <arm_math.h>
+
+
+#define GBOOST_TONE_MINF	(800.0f)
+#define GBOOST_TONE_MAXF	(8000.0f)
+#define GBOOST_LP2_F		(10000.0f)
+#define GBOOST_BOTTOM_MINF	(50.0f)
+#define GBOOST_BOTTOM_MAXF	(350.0f)
+
+class AudioEffectGuitarBooster_F32 : public AudioStream_F32
+{
+public:
+	AudioEffectGuitarBooster_F32(void) : AudioStream_F32(2, inputQueueArray)
+	{
+		fs_Hz = AUDIO_SAMPLE_RATE_EXACT;
+		blockSize = AUDIO_BLOCK_SAMPLES;
+		begin();
+	}
+
+	AudioEffectGuitarBooster_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray)
+	{
+		fs_Hz = settings.sample_rate_Hz;
+		blockSize = settings.audio_block_samples;
+		begin();		
+	}
+
+	virtual void update();
+	
+	void begin()
+	{
+		arm_fir_interpolate_init_f32(&interpolator, upsample_k, FIR_taps, (float32_t*)FIR_coeffs, interpState, AUDIO_BLOCK_SAMPLES);
+		arm_fir_decimate_init_f32(&decimator, FIR_taps, upsample_k, (float32_t*)FIR_coeffs, decimState, upsample_k * AUDIO_BLOCK_SAMPLES);
+		bottom(1.0f);
+		tone(1.0f);
+		hpPost_k = omega(GBOOST_BOTTOM_MINF);
+		lp2_k = omega(GBOOST_LP2_F);
+	}
+	void drive(float32_t value)
+	{
+		value = fabs(value);
+		value = 1.0f + value * upsample_k;
+		__disable_irq()
+		gainSet = value;
+		__enable_irq();
+	}
+	void bottom(float32_t bottom);
+	void tone(float32_t t);
+	void bias(float32_t b)
+	{
+		b = constrain(b, -1.0f, 1.0f);
+		__disable_irq();
+		DCbias = b;
+		__enable_irq();
+	}
+
+	void mix(float32_t m)
+	{
+		float32_t d, w;
+		m = constrain(m, 0.0f, 1.0f);
+		mix_pwr(m, &w, &d);
+		__disable_irq();
+		wetGain = w;
+		dryGain = d;
+		__enable_irq();
+	}
+	void volume(float32_t l)
+	{
+		l = constrain(l, 0.0f, 1.0f);
+		__disable_irq();
+		levelSet = l;
+		__enable_irq();
+	}
+	// Bypass 
+    bool bypass_get(void) {return bp;}
+    void bypass_set(bool state) {bp = state;}
+    bool bypass_tgl(void) 
+    {
+        bp ^= 1; 
+        return bp;
+    }
+
+	bool octave_get(void) {return octave;}
+    void octave_set(bool state) {octave = state;}
+    bool octave_tgl(void) 
+    {
+        octave ^= 1; 
+        return octave;
+    }
+private:
+	audio_block_f32_t *inputQueueArray[2];
+	float fs_Hz;
+	uint16_t blockSize;
+	static const uint8_t upsample_k = 5;
+	static const uint8_t FIR_taps = 75;
+	static constexpr float32_t FIR_coeffs[FIR_taps] =
+	{
+		-0.000033, 0.000112,-0.000100,-0.000103, 0.000361,-0.000331,-0.000181, 0.000822,-0.000824,-0.000205, 
+		 0.001556,-0.001737,-0.000054, 0.002607,-0.003263, 0.000461, 0.003979,-0.005641, 0.001621, 0.005626, 
+		-0.009170, 0.003841, 0.007448,-0.014287, 0.007776, 0.009301,-0.021812, 0.014667, 0.011009,-0.033775, 
+		 0.027639, 0.012392,-0.057262, 0.058949, 0.013293,-0.142816, 0.268001, 0.680272, 0.268001,-0.142816, 
+		 0.013293, 0.058949,-0.057262, 0.012392, 0.027639,-0.033775, 0.011009, 0.014667,-0.021812, 0.009301, 
+		 0.007776,-0.014287, 0.007448, 0.003841,-0.009170, 0.005626, 0.001621,-0.005641, 0.003979, 0.000461, 
+		-0.003263, 0.002607,-0.000054,-0.001737, 0.001556,-0.000205,-0.000824, 0.000822,-0.000181,-0.000331, 
+		 0.000361,-0.000103,-0.000100, 0.000112,-0.000033
+	};
+	float32_t blockInterpolated[upsample_k * AUDIO_BLOCK_SAMPLES];
+
+	float32_t interpState[(FIR_taps / upsample_k) + AUDIO_BLOCK_SAMPLES - 1];
+	float32_t decimState[FIR_taps + (upsample_k * AUDIO_BLOCK_SAMPLES) - 1];
+
+	arm_fir_interpolate_instance_f32 interpolator;
+	arm_fir_decimate_instance_f32 decimator;
+	arm_linear_interp_instance_f32 waveshaper = 
+	{
+		2000, -1.0f, 2.0f/2000.0f, &driveWaveform[0]
+	};
+	bool bp = true; // bypass flag
+
+	bool octave = true;
+
+	float32_t dryGain = 0.0f;
+	float32_t wetGain = 1.0f;
+	float32_t  DCbias = 0.175f;
+
+    float32_t gainSet = 1.0f;
+    float32_t gain = 0.0f;
+	float32_t gain_hp = 1.0f;
+    float32_t levelSet = 1.0f;
+    float32_t level = 1.0f;
+    float32_t lp1_k = 0.0f;
+    float32_t lp1_reg = 0.0f;
+    float32_t lp2_k = 0.0f;
+    float32_t lp2_reg = 0.0f;
+    float32_t hpPre1_k = 0.0f;
+    float32_t hpPre1_reg = 0.0f;
+    float32_t hpPre2_k = 0.0f;
+    float32_t hpPre2_reg = 0.0f;	
+    float32_t hpPost_k = 0.0f;
+    float32_t hpPost_reg = 0.0f;	
+
+	static float32_t driveWaveform[2001];
+
+	inline float32_t omega(float f)
+	{
+		float32_t fs = fs_Hz * upsample_k;
+		return 1.0f - expf(-TWO_PI * f / fs);
+	}
+};
+
+#endif // _EFFECT_GUITARBOOSTER_F32_H_

src/effect_noiseGateStereo_F32.h

@@ -4,9 +4,10 @@
  * Created: Max Huster, Feb 2021
  * Purpose: This module mutes the Audio completly, when it's below a given threshold.
  *
- * This processes a single stream fo audio data (ie, it is mono)
- *
  * MIT License.  use at your own risk.
+ * 
+ * 03.2024 - stereo version with optional side chain input via pointers
+ * 	by Piotr Zapart (www.hexefx.com)
  */
 
 #ifndef _AudioEffectNoiseGateStereo_F32_h
@@ -18,94 +19,150 @@
 class AudioEffectNoiseGateStereo_F32 : public AudioStream_F32
 {
 public:
-	// constructor
+	AudioEffectNoiseGateStereo_F32(float32_t* sideChainSrcL, float32_t* sideChainSrcR) : AudioStream_F32(2, inputQueueArray_f32)
-	AudioEffectNoiseGateStereo_F32(void) : AudioStream_F32(4, inputQueueArray_f32){};
+	{
-	AudioEffectNoiseGateStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(4, inputQueueArray_f32){};
+		p_sideChain_inL = sideChainSrcL;
+		p_sideChain_inR = sideChainSrcR;
+		setDefaults();
+	};
+	AudioEffectNoiseGateStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32) 
+	{ 
+		fs = settings.sample_rate_Hz;
+		setDefaults(); 
+	}
 
 	void update(void)
 	{
-		audio_block_f32_t *blockL, *blockR, *blockSideChL, *blockSideChR, *blockSideCh, *blockGain;
+		audio_block_f32_t *blockL, *blockR, *blockSideCh, *blockGain;
-		blockL = AudioStream_F32::receiveWritable_f32(0);
+		audio_block_f32_t *blockOutL, *blockOutR;
-		blockR = AudioStream_F32::receiveWritable_f32(1);
-		blockSideChL = AudioStream_F32::receiveReadOnly_f32(2); // side chain inputL
-		blockSideChR = AudioStream_F32::receiveReadOnly_f32(3); // side chain inputR
-		blockSideCh = AudioStream_F32::allocate_f32();			// allocate new block for summed L+R
-		blockGain = AudioStream_F32::allocate_f32();			// create a new audio block for the gain
 
+		blockL = AudioStream_F32::receiveReadOnly_f32(0);
+		blockR = AudioStream_F32::receiveReadOnly_f32(1);
+		// no input signal
+		if (!blockL || !blockR)
+		{
+			if (blockL) AudioStream_F32::release(blockL);
+			if (blockR) AudioStream_F32::release(blockR);
+			return;
+		}
+		if (bp)
+		{
+			AudioStream_F32::transmit(blockL, 0);	
+			AudioStream_F32::transmit(blockR, 1);
+			AudioStream_F32::release(blockL);
+			AudioStream_F32::release(blockR);
+			return;
+		}
 
-		if (!blockL || !blockR || !blockSideChL || !blockSideChR || !blockSideCh || !blockGain) 
+		blockSideCh = AudioStream_F32::allocate_f32();			// allocate new block for summed L+R
+		blockGain = AudioStream_F32::allocate_f32();			// create a new audio block for the gain	
+		if (!p_sideChain_inL || !p_sideChain_inR || !blockSideCh || !blockGain) 
 		{
-			if (blockSideChL) AudioStream_F32::release(blockSideChL);
-			if (blockSideChR) AudioStream_F32::release(blockSideChR);
 			if (blockSideCh) AudioStream_F32::release(blockSideCh);
 			if (blockGain) AudioStream_F32::release(blockGain);		
-			if (blockL) AudioStream_F32::release(blockL);
+			AudioStream_F32::transmit(blockL, 0);
-			if (blockR) AudioStream_F32::release(blockR);
+			AudioStream_F32::transmit(blockR, 1);
+			AudioStream_F32::release(blockL);
+			AudioStream_F32::release(blockR);
+			return;
+		}
+		blockOutL = AudioStream_F32::allocate_f32();
+		blockOutR = AudioStream_F32::allocate_f32();
+		if (!blockOutL || !blockOutR)
+		{
+			if (blockOutL) AudioStream_F32::release(blockOutL);
+			if (blockOutR)  AudioStream_F32::release(blockOutR);
 			return;
 		}		
 		//sum L + R
-		arm_add_f32(blockSideChL->data, blockSideChR->data, blockSideCh->data, blockSideCh->length);
+		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
 		
-		// calculate the desired gain
 		calcGain(blockSideCh, blockGain);
-		// smooth the "blocky" gain block
 		calcSmoothedGain(blockGain);
 
-		// multiply it to the input singal
+		arm_mult_f32(blockGain->data, blockL->data, blockOutL->data, blockOutL->length);
-		arm_mult_f32(blockGain->data, blockL->data, blockL->data, blockL->length);
+		arm_mult_f32(blockGain->data, blockR->data, blockOutR->data, blockOutR->length);
-		arm_mult_f32(blockGain->data, blockR->data, blockR->data, blockR->length);
+
-		// release gainBlock
 		AudioStream_F32::release(blockGain);
 		AudioStream_F32::release(blockSideCh);
-		AudioStream_F32::release(blockSideChL);
-		AudioStream_F32::release(blockSideChR);
 
-		// transmit the block and be done
+		AudioStream_F32::transmit(blockOutL, 0);
-		AudioStream_F32::transmit(blockL, 0);
+		AudioStream_F32::transmit(blockOutR, 1);
-		AudioStream_F32::transmit(blockR, 1);
 		AudioStream_F32::release(blockL);
 		AudioStream_F32::release(blockR);
+		AudioStream_F32::release(blockOutL);
+		AudioStream_F32::release(blockOutR);
 	}
 
+	/**
+	 * @brief Set gfate threshold in decibels
+	 * 		value less than -99dB turns the bypass on
+	 * 
+	 * @param dbfs gate threshold in dB
+	 */
 	void setThreshold(float dbfs)
 	{
+		if (dbfs < -99.0f) 	bp = true;
+		else 				bp = false;
 		// convert dbFS to linear value to comapre against later
 		linearThreshold = pow10f(dbfs / 20.0f);
 	}
 
 	void setOpeningTime(float timeInSeconds)
 	{
-		openingTimeConst = expf(-1.0f / (timeInSeconds * AUDIO_SAMPLE_RATE));
+		openingTimeConst = expf(-1.0f / (timeInSeconds * fs));
 	}
 
 	void setClosingTime(float timeInSeconds)
 	{
-		closingTimeConst = expf(-1.0f / (timeInSeconds * AUDIO_SAMPLE_RATE));
+		closingTimeConst = expf(-1.0f / (timeInSeconds * fs));
 	}
 
 	void setHoldTime(float timeInSeconds)
 	{
-		holdTimeNumSamples = timeInSeconds * AUDIO_SAMPLE_RATE;
+		holdTimeNumSamples = timeInSeconds * fs;
 	}
 
 	bool infoIsOpen()
 	{
 		return _isOpenDisplay;
 	}
-
+ void bypass_set(bool state) 
+	{
+		__disable_irq();
+		bp = state;
+		__enable_irq();
+	}
+    bool bypass_tgl(void) 
+    {
+		bool bp_new = bp ^ 1;
+		__disable_irq();
+        bp  = bp_new; 
+		__enable_irq();
+        return bp;
+    }
 private:
+	float32_t fs = AUDIO_SAMPLE_RATE_EXACT;
+	float32_t* p_sideChain_inL = NULL;
+	float32_t* p_sideChain_inR = NULL;
 	float32_t linearThreshold;
 	float32_t prev_gain_dB = 0;
 	float32_t openingTimeConst, closingTimeConst;
-	float lastGainBlockValue = 0;
+	float32_t lastGainBlockValue = 0;
 	int32_t counter, holdTimeNumSamples = 0;
 	audio_block_f32_t *inputQueueArray_f32[4];
 	bool falling = false;
-
+	bool bp = false;
 	bool _isOpen = false;
 	bool _isOpenDisplay = false;
-	bool _extSideChain = true;
+
+	void setDefaults()
+	{
+		setOpeningTime(0.01f);
+		setClosingTime(0.05f);
+		setHoldTime(0.01f);
+	}
 
 	void calcGain(audio_block_f32_t *input, audio_block_f32_t *gainBlock)
 	{
@@ -153,7 +210,6 @@ private:
 		for (int i = 0; i < gain_block->length; i++)
 		{
 			gain = gain_block->data[i];
-
 			// smooth the gain using the opening or closing constants
 			if (gain > prev_gain_dB)
 			{ // are we in the opening phase?
@@ -163,7 +219,6 @@ private:
 			{ // or, we're in the closing phase
 				gain_block->data[i] = closingTimeConst * prev_gain_dB + one_minus_closing_const * gain;
 			}
-
 			// save value for the next time through this loop
 			prev_gain_dB = gain_block->data[i];
 		}

src/effect_platereverb_F32.cpp

@@ -46,7 +46,9 @@ AudioEffectPlateReverb_F32::AudioEffectPlateReverb_F32() : AudioStream_F32(2, in
 
 bool AudioEffectPlateReverb_F32::begin()
 {
-    input_attn = 0.5f;
+	inputGainSet = 0.5f;
+    inputGain = 0.5f;
+	inputGain_tmp = 0.5f;
     wet_gain = 1.0f;        // default mode: wet signal only
     dry_gain = 0.0f;
     in_allp_k = INP_ALLP_COEFF;
@@ -129,7 +131,7 @@ void AudioEffectPlateReverb_F32::update()
     // handle bypass, 1st call will clean the buffers to avoid continuing the previous reverb tail
     if (flags.bypass)
     {
-        if (!flags.cleanup_done)
+        if (!flags.cleanup_done && bp_mode != BYPASS_MODE_TRAILS)
         {
 			in_allp_1L.reset();
 			in_allp_2L.reset();
@@ -149,9 +151,9 @@ void AudioEffectPlateReverb_F32::update()
 			lp_dly4.reset();
             flags.cleanup_done = 1;
         }
-
+		switch(bp_mode)
-        if (dry_gain > 0.0f) 		// if dry/wet mixer is used
 		{
+			case BYPASS_MODE_PASS:
 				blockL = AudioStream_F32::receiveReadOnly_f32(0);
 				blockR = AudioStream_F32::receiveReadOnly_f32(1);
 				if (!blockL || !blockR) 
@@ -165,7 +167,8 @@ void AudioEffectPlateReverb_F32::update()
 				AudioStream_F32::release(blockL);
 				AudioStream_F32::release(blockR);
 				return;
-		}
+				break;
+			case BYPASS_MODE_OFF:
 				blockL = AudioStream_F32::allocate_f32();
 				if (!blockL) return;
 				memset(&blockL->data[0], 0, blockL->length*sizeof(float32_t));
@@ -173,9 +176,12 @@ void AudioEffectPlateReverb_F32::update()
 				AudioStream_F32::transmit(blockL, 1);
 				AudioStream_F32::release(blockL);	
 				return;
+				break;
+			case BYPASS_MODE_TRAILS:
+			default:
+				break;
+		}
     }
-    flags.cleanup_done = 0;
-
     blockL = AudioStream_F32::receiveWritable_f32(0);
     blockR = AudioStream_F32::receiveWritable_f32(1);
 
@@ -185,6 +191,7 @@ void AudioEffectPlateReverb_F32::update()
 		if (blockR) AudioStream_F32::release(blockR);
 		return;
 	}
+	flags.cleanup_done = 0;
     rv_time = rv_time_k;
 
 	for (i=0; i < blockL->length; i++) 
@@ -193,7 +200,9 @@ void AudioEffectPlateReverb_F32::update()
 		lfo1.update();
 		lfo2.update();
 
-		acc = blockL->data[i] * input_attn;
+		inputGain += (inputGainSet - inputGain) * 0.25f;
+
+		acc = blockL->data[i] * inputGain;
 
         // chained input allpasses, channel L
 		acc = in_allp_1L.process(acc);
@@ -203,7 +212,7 @@ void AudioEffectPlateReverb_F32::update()
 		in_allp_out_L = pitchL.process(in_allp_out_L); 
 
 		// chained input allpasses, channel R
-        acc = blockR->data[i] * input_attn;
+        acc = blockR->data[i] * inputGain;
 
 		acc = in_allp_1R.process(acc);
 		acc = in_allp_2R.process(acc);

src/effect_platereverb_F32.h

@@ -28,16 +28,6 @@
  * Algorithm based on plate reverbs developed for SpinSemi FV-1 DSP chip
  * 
  * Allpass + modulated delay line based lush plate reverb
- * 
- * Input parameters are float in range 0.0 to 1.0:
- * 
- * size - reverb time
- * hidamp - hi frequency loss in the reverb tail
- * lodamp - low frequency loss in the reverb tail
- * lowpass - output/master lowpass filter, useful for darkening the reverb sound 
- * diffusion - lower settings will make the reverb tail more "echoey".
- * freeze - infinite reverb tail effect
- * 
  */
 
 #ifndef _EFFECT_PLATEREVERB_F32_H_
@@ -60,38 +50,63 @@ public:
 
     bool begin(void);
 
+	/**
+	 * @brief sets the reverb time
+	 * 
+	 * @param n range 0.0f - 1.0f
+	 */
     void size(float n)
     {
         n = constrain(n, 0.0f, 1.0f);
 		n = 2*n - n*n;
         n = map(n, 0.0f, 1.0f, 0.2f, rv_time_k_max);
-        //float attn = map(n, 0.2f, rv_time_k_max, 0.5f, 0.25f);
+		rv_time_k_tmp = n;
+		inputGain_tmp = 0.5f;
 		__disable_irq();
         rv_time_k = n;
-        input_attn = 0.5f;
+        inputGainSet = 0.5f;
 		__enable_irq();
     }
-
+	/**
+	 * @brief returns the set reverb time
+	 * 
+	 * @return float reverb time value
+	 */
 	float size_get(void) {return rv_time_k;}
 
+	/**
+	 * @brief Treble loss in reverb tail
+	 * 
+	 * @param n 0.0f to 1.0f
+	 */
     void hidamp(float n)
     {
         n = 1.0f - constrain(n, 0.0f, 1.0f);
+		lp_hidamp_k_tmp = n;
 		__disable_irq();
         lp_hidamp_k = n;
 		__enable_irq();
     }
-    
+    /**
+     * @brief Bass loss in reverb tails
+     * 
+     * @param n 0.0f to 1.0f
+     */
     void lodamp(float n)
     {
         n = -constrain(n, 0.0f, 1.0f);
 		float32_t tscal = 1.0f + n*0.12f; //n is negativbe here
+		lp_lodamp_k_tmp = n;
 		__disable_irq();
         lp_lodamp_k = n;
         rv_time_scaler = tscal;        // limit the max reverb time, otherwise it will clip
 		__enable_irq();
 	}
-
+	/**
+	 * @brief Output lowpass filter
+	 * 
+	 * @param n 0.0f to 1.0f
+	 */
     void lowpass(float n)
     {
         n = 1.0f - constrain(n, 0.0f, 1.0f);
@@ -99,6 +114,11 @@ public:
 		master_lp_k = n;
 		__enable_irq();
     }
+	/**
+	 * @brief Output highpass filter
+	 * 
+	 * @param n 0.0f 1.0f
+	 */
     void hipass(float n)
 	{
 		n = -constrain(n, 0.0f, 1.0f);
@@ -106,6 +126,12 @@ public:
 		master_hp_k = n;
 		__enable_irq();
 	}
+	/**
+	 * @brief reverb tail diffusion, 
+	 * 	lower values produce more single repeats, echos
+	 * 
+	 * @param n 0.0f - 1.0f
+	 */
     void diffusion(float n)
     {
         n = constrain(n, 0.0f, 1.0f);
@@ -115,9 +141,15 @@ public:
         loop_allp_k = n;
 		__enable_irq();
     }
-
+	/**
+	 * @brief Freeze option On/Off. Freeze sets the reverb
+	 * 	time to infinity and mutes (almost) the input signal
+	 * 
+	 * @param state 
+	 */
     void freeze(bool state)
     {
+		if (flags.freeze == state || flags.bypass) return;
         flags.freeze = state;
         if (state)
         {
@@ -126,7 +158,7 @@ public:
             lp_hidamp_k_tmp = lp_hidamp_k;
             __disable_irq();
             rv_time_k = freeze_rvtime_k;                                      
-            input_attn = freeze_ingain;
+            inputGainSet = freeze_ingain;
             rv_time_scaler = 1.0f;
             lp_lodamp_k = freeze_lodamp_k;
             lp_hidamp_k = freeze_hidamp_k;
@@ -136,11 +168,14 @@ public:
         }
         else
         {
-            //float attn = map(rv_time_k_tmp, 0.0f, rv_time_k_max, 0.5f, 0.25f);    // recalc the in attenuation
             float sc = 1.0f - lp_lodamp_k_tmp * 0.12f;									// scale up the reverb time due to bass loss
 			__disable_irq();
             rv_time_k = rv_time_k_tmp;                                      // restore the value
-            input_attn = 0.5f;
+            if (!flags.bypass)
+			{
+				inputGainSet = 0.5f;
+				inputGain_tmp = 0.5f;
+			}
             rv_time_scaler = sc;
             lp_hidamp_k = lp_hidamp_k_tmp;
             lp_lodamp_k = lp_lodamp_k_tmp;
@@ -160,9 +195,13 @@ public:
 		b = constrain(b, 0.0f, 1.0f);
 		b = map(b, 0.0f, 1.0f, 0.0f, 0.1f);
 		freeze_ingain = b;
-		if (flags.freeze) input_attn = b; // update input gain if freeze is enabled
+		if (flags.freeze) inputGainSet = b; // update input gain if freeze is enabled
 	}
-
+	/**
+	 * @brief Internal Dry / Wet mixer
+	 * 
+	 * @param m 0.0f (full dry) - 1.0f (full wet)
+	 */
     void mix(float m)
     {
 		float32_t dry, wet;
@@ -173,7 +212,11 @@ public:
 		dry_gain = dry;
 		__enable_irq();
 	}
-	
+	/**
+	 * @brief wet signal volume
+	 * 
+	 * @param wet 0.0f - 1.0f
+	 */
     void wet_level(float wet)
     {
 		wet = constrain(wet, 0.0f, 6.0f);
@@ -181,7 +224,11 @@ public:
 		wet_gain = wet;
 		__enable_irq();
     }
-
+	/**
+	 * @brief dry signal volume
+	 * 
+	 * @param dry 0.0f - 1.0f
+	 */
     void dry_level(float dry)
     {
         dry = constrain(dry, 0.0f, 1.0f);
@@ -189,21 +236,51 @@ public:
 		dry_gain = dry;
 		__enable_irq();
     }
-
+	/**
-    bool freeze_tgl() {flags.freeze ^= 1; freeze(flags.freeze); return flags.freeze;}
+	 * @brief toogle the Freeze mode
-    
+	 * 
+	 * @return true 
+	 * @return false 
+	 */
+    bool freeze_tgl() {freeze(flags.freeze^1); return flags.freeze;}
+    /**
+     * @brief return the Freeze mode state
+     * 
+     * @return true 
+     * @return false 
+     */
     bool freeze_get() {return flags.freeze;}
  	
+	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;
+	/**
+	 * @brief sets the bypass mode (see above)
+	 * 
+	 * @param m 
+	 */
+	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 flags.bypass;}
     void bypass_set(bool state) 
     {
         flags.bypass = state;
-        if (state) freeze(false);       // disable freeze in bypass mode
+        if (state) 
+		{
+			if (bp_mode == BYPASS_MODE_TRAILS) inputGainSet = 0.0f;
+			freeze(false);       // disable freeze in bypass mode
+		}
+		else inputGainSet = inputGain_tmp;
     }
     bool bypass_tgl(void) 
     {
-        flags.bypass ^= 1; 
+		bypass_set(flags.bypass^1);
-        if (flags.bypass) freeze(false);       // disable freeze in bypass mode
         return flags.bypass;
     }
 
@@ -219,9 +296,9 @@ public:
 	}
 
 	/**
-	 * @brief 
+	 * @brief Contriols the amount of shimmer effect
 	 * 
-	 * @param s 
+	 * @param s 0.0f - 1.0f
 	 */
 	void shimmer(float s)
 	{
@@ -232,11 +309,21 @@ public:
 		pitchShimR.setMix(s);
 		shimmerRatio = s;
 	}
+	/**
+	 * @brief Sets the pitch of the shimmer effect
+	 * 
+	 * @param ratio pitch up (>1.0f)  or down (<1.0f) ratio
+	 */
 	void shimmerPitch(float ratio)
 	{
 		pitchShimL.setPitch(ratio);
 		pitchShimR.setPitch(ratio);
 	}
+	/**
+	 * @brief Sets the shimmer effect pitch in semitones
+	 * 
+	 * @param semitones 
+	 */
 	void shimmerPitchSemitones(int8_t semitones)
 	{
 		pitchShimL.setPitchSemintone(semitones);
@@ -252,6 +339,11 @@ public:
 		pitchL.setPitchSemintone(semitones);
 		pitchR.setPitchSemintone(semitones);
 	}
+	/**
+	 * @brief Reverb pitch shifter dry/wet mixer
+	 * 
+	 * @param s 0.0f(dry reverb) - 1.0f (100% pitch shifter out)
+	 */
 	void pitchMix(float s)
 	{
 		s = constrain(s, 0.0f, 1.0f);
@@ -268,6 +360,7 @@ private:
         unsigned shimmer:           1;
         unsigned cleanup_done:      1;
     }flags;
+	bypass_mode_t bp_mode = BYPASS_MODE_PASS;
     audio_block_f32_t *inputQueueArray_f32[2];
 
 	static const uint16_t IN_ALLP1_BUFL_LEN = 224u;
@@ -320,7 +413,9 @@ private:
 	AudioBasicLfo lfo1 = AudioBasicLfo(1.35f, LFO_AMPL);
 	AudioBasicLfo lfo2 = AudioBasicLfo(1.57f, LFO_AMPL);
 
-    float input_attn;
+    float inputGain;
+	float inputGainSet;
+	float inputGain_tmp;
     float wet_gain;
     float dry_gain;
 

src/effect_reverbsc_F32.cpp

@@ -1,7 +1,7 @@
 #include "effect_reverbsc_F32.h"
 
 
-#define REVERBSC_DLYBUF_SIZE 98936
+
 #define DELAYPOS_SHIFT 28
 #define DELAYPOS_SCALE 0x10000000
 #define DELAYPOS_MASK 0x0FFFFFFF
@@ -30,6 +30,8 @@ static int DelayLineBytesAlloc(float32_t sr, float32_t i_pitch_mod, int n);
 static const float32_t kOutputGain = 0.35f;
 static const float32_t kJpScale = 0.25f;
 
+extern uint8_t external_psram_size;
+
 AudioEffectReverbSc_F32::AudioEffectReverbSc_F32(bool use_psram) : AudioStream_F32(2, inputQueueArray_f32)
 {
 	sample_rate_ = AUDIO_SAMPLE_RATE_EXACT;
@@ -37,13 +39,27 @@ AudioEffectReverbSc_F32::AudioEffectReverbSc_F32(bool use_psram) : AudioStream_F
 	lpfreq_ = 10000;
 	i_pitch_mod_ = 1;
 	damp_fact_ = 0.195847f; // ~16kHz
+	flags.mem_fail = 0;
+	flags.bypass = 0;
+	flags.freeze = 0;
+	flags.cleanup_done = 1;
+	flags.memsetup_done = 0;
 	
 	int i, n_bytes = 0;
 	n_bytes = 0;
-	if (use_psram)	aux_ = (float32_t *) extmem_malloc(REVERBSC_DLYBUF_SIZE*sizeof(float32_t));
+	if (use_psram)	
+	{
+		// no PSRAM detected - enter the memoery failsafe mode = fixed bypass
+		if (external_psram_size == 0) 
+		{
+			flags.mem_fail = 1;
+			return;
+		}
+		aux_ = (float32_t *) extmem_malloc(aux_size_bytes);
+	}
 	else			
 	{
-		aux_ = (float32_t *) malloc(REVERBSC_DLYBUF_SIZE*sizeof(float32_t));
+		aux_ = (float32_t *) malloc(aux_size_bytes);
 		flags.memsetup_done = 1; 
 	}
 	if (!aux_) return;
@@ -57,8 +73,7 @@ AudioEffectReverbSc_F32::AudioEffectReverbSc_F32(bool use_psram) : AudioStream_F
 		n_bytes += DelayLineBytesAlloc(AUDIO_SAMPLE_RATE_EXACT, 1, i);
 	}
 	mix(0.5f);
-	flags.bypass = 0;
+
-	flags.freeze = 0;
 	initialised = true;
 }
 
@@ -135,14 +150,6 @@ void AudioEffectReverbSc_F32::InitDelayLine(ReverbScDl_t *lp, int n)
 void AudioEffectReverbSc_F32::update()
 {
 #if defined(__IMXRT1062__)
-	if (!initialised) return;
-	if ( !flags.memsetup_done)
-	{
-		memset(aux_, 0, REVERBSC_DLYBUF_SIZE*sizeof(float32_t));
-		arm_dcache_flush_delete(aux_, REVERBSC_DLYBUF_SIZE*sizeof(float32_t));
-		flags.memsetup_done = 1;
-		return;
-	}
 	audio_block_f32_t *blockL, *blockR;
 	int16_t i;
 	float32_t a_in_l, a_in_r, a_out_l, a_out_r, dryL, dryR;
@@ -153,10 +160,39 @@ void AudioEffectReverbSc_F32::update()
 	int buffer_size; /* Local copy */
 	float32_t damp_fact = damp_fact_;
 	
+	if (!initialised) return;
+	// special case if memory allocation failed, pass the input signal directly to the output
+	if (flags.mem_fail)
+	{
+		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;
+	}
+
+	if ( !flags.memsetup_done)
+	{
+		flags.memsetup_done  = memCleanup();
+		return;
+	}
 	if (flags.bypass)
     {
-        if (dry_gain > 0.0f) 		// if dry/wet mixer is used
+		if (!flags.cleanup_done && bp_mode != BYPASS_MODE_TRAILS)
 		{
+			flags.cleanup_done = memCleanup();
+		}
+        switch(bp_mode)
+		{
+			case BYPASS_MODE_PASS:
 				blockL = AudioStream_F32::receiveReadOnly_f32(0);
 				blockR = AudioStream_F32::receiveReadOnly_f32(1);
 				if (!blockL || !blockR) 
@@ -170,7 +206,8 @@ void AudioEffectReverbSc_F32::update()
 				AudioStream_F32::release(blockL);
 				AudioStream_F32::release(blockR);
 				return;
-		}
+				break;
+			case BYPASS_MODE_OFF:
 				blockL = AudioStream_F32::allocate_f32();
 				if (!blockL) return;
 				memset(&blockL->data[0], 0, blockL->length*sizeof(float32_t));
@@ -178,6 +215,12 @@ void AudioEffectReverbSc_F32::update()
 				AudioStream_F32::transmit(blockL, 1);
 				AudioStream_F32::release(blockL);	
 				return;
+				break;
+			case BYPASS_MODE_TRAILS:
+				break;
+			default:
+				break;
+		}
     }
 	blockL = AudioStream_F32::receiveWritable_f32(0);
 	blockR = AudioStream_F32::receiveWritable_f32(1);
@@ -189,8 +232,10 @@ void AudioEffectReverbSc_F32::update()
 			AudioStream_F32::release(blockR);
 		return;
 	}
+	flags.cleanup_done = 0;
 	for (i = 0; i < blockL->length; i++)
 	{
+		input_gain += (input_gain_set - input_gain) * 0.25f;
 		/* calculate "resultant junction pressure" and mix to input signals */
 		a_in_l = a_out_l = a_out_r = 0.0f;
 		dryL = blockL->data[i] * input_gain;
@@ -290,15 +335,16 @@ void AudioEffectReverbSc_F32::update()
 
 void AudioEffectReverbSc_F32::freeze(bool state)
 {
+	if (flags.freeze == state) return;
 	flags.freeze = state;
 	if (state)
 	{
 		feedback_tmp = feedback_;      // store the settings
 		damp_fact_tmp = damp_fact_;
-		input_gain_tmp = input_gain;
+		input_gain_tmp = input_gain_set;
 		__disable_irq();
 		feedback_ = 1.0f; // infinite reverb                                      
-		input_gain = freeze_ingain;
+		input_gain_set = freeze_ingain;
 		__enable_irq();
 	}
 	else
@@ -306,7 +352,39 @@ void AudioEffectReverbSc_F32::freeze(bool state)
 		__disable_irq();
 		feedback_ = feedback_tmp;
 		damp_fact_ = damp_fact_tmp;
-		input_gain = input_gain_tmp;
+		if (!flags.bypass)
+		{
+			input_gain_set = input_gain_tmp;
+		}
 		__enable_irq();
 	}
 }
+
+/**
+ * @brief Partial memory clear
+ * 	Clearing all the delay buffers at once, esp. if 
+ * 	the PSRAM is used takes too long for the audio ISR.
+ *  Hence the buffer clear is done in configurable portions
+ *  spread over a few audio update routines.
+ * 
+ * @return true 	Memory clean is complete
+ * @return false 	Memory clean still in progress
+ */
+bool AudioEffectReverbSc_F32::memCleanup()
+{
+	bool result = false;
+	if (memCleanupEnd > REVERBSC_DLYBUF_SIZE) // last segment
+	{
+		memCleanupEnd = REVERBSC_DLYBUF_SIZE;
+		result = true;	
+	}
+	uint32_t l = (memCleanupEnd - memCleanupStart) * sizeof(float32_t);
+	uint8_t* memPtr = (uint8_t *)&aux_[0]+(memCleanupStart*sizeof(float32_t));
+	memset(memPtr, 0, l);
+	arm_dcache_flush_delete(memPtr, l);
+
+	memCleanupStart = memCleanupEnd;
+	memCleanupEnd += memCleanupStep;
+	
+	return result;
+}

src/effect_reverbsc_F32.h

@@ -8,13 +8,9 @@
  * Year: 1999, 2005
  * Ported to soundpipe by:  Paul Batchelor
  * 
- * Ported to Teensy4 and OpenAudio_ArduinoLibrary: 
+ * Ported/upgraded to Teensy4 and OpenAudio_ArduinoLibrary: 
  * 01.2024 Piotr Zapart www.hexefx.com 
  * 
- * Fixes, changes:
- * - In the original code the reverb level is affected by the feedback control, fixed
- * - Optional 
- * 
  */
 
 #ifndef _EFFECT_REVERBSC_F32_H_
@@ -27,6 +23,8 @@
 #include "arm_math.h"
 #include "basic_DSPutils.h"
 
+#define REVERBSC_DLYBUF_SIZE 98936
+
 class AudioEffectReverbSc_F32 : public AudioStream_F32
 {
 public:
@@ -57,7 +55,7 @@ public:
 		feedback_tmp = feedb;
 		inGain = map(feedb, 0.1f, feedb_max, 0.5f, 0.2f);
 		__disable_irq();
-		input_gain = inGain;
+		input_gain_set = inGain;
 		feedback_ = feedb;
 		__enable_irq();
 	}
@@ -89,44 +87,70 @@ public:
 
     void wet_level(float32_t wet)
     {
-        wet_gain = constrain(wet, 0.0f, 1.0f);
+		wet = constrain(wet, 0.0f, 1.0f);
+		__disable_irq();
+        wet_gain = wet;
+		__enable_irq();
     }
 
     void dry_level(float32_t dry)
     {
-        dry_gain = constrain(dry, 0.0f, 1.0f);
+		dry = constrain(dry, 0.0f, 1.0f);
+		__disable_irq();
+        dry_gain = dry;
+		__enable_irq();
     }	
 	void freeze(bool state);
-    bool freeze_tgl() {flags.freeze ^= 1; freeze(flags.freeze); return flags.freeze;}
+    bool freeze_tgl() {freeze(flags.freeze^1); return flags.freeze;}
     bool freeze_get() {return flags.freeze;}
-    
+ 	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) 
+		{
+			__disable_irq();
+			bp_mode = m;
+			__enable_irq();
+		}
+	}
+	bypass_mode_t bypass_geMode() {return bp_mode;}
     bool bypass_get(void) {return flags.bypass;}
     void bypass_set(bool state) 
     {
+		if (flags.mem_fail) return;
         flags.bypass = state;
-        if (state) freeze(false);       // disable freeze in bypass mode
+        if (state) 
+		{
+			if (bp_mode == BYPASS_MODE_TRAILS) input_gain_set = 0.0f;
+			freeze(false);       // disable freeze in bypass mode
+			__disable_irq();
+			memCleanupStart = 0;
+			memCleanupEnd = memCleanupStep;
+			__enable_irq();
+		}
+		else input_gain_set = input_gain_tmp;
     }
     bool bypass_tgl(void) 
     {
-        flags.bypass ^= 1; 
+		bypass_set(flags.bypass^1);
-        if (flags.bypass) freeze(false);       // disable freeze in bypass mode
         return flags.bypass;
     }
 
-	uint32_t getBfAddr()
-	{
-		float32_t *addr = aux_;
-		return (uint32_t)addr;
-	}
-
 private:
     struct flags_t
     {
         unsigned bypass:            1;
         unsigned freeze:            1;
+		unsigned cleanup_done:		1;
 		unsigned memsetup_done:		1;
+		unsigned mem_fail:			1;
     }flags = {0, 0, 0};
-
+	bypass_mode_t bp_mode;
 	audio_block_f32_t *inputQueueArray_f32[2];
     void NextRandomLineseg(ReverbScDl_t *lp, int n);
     void InitDelayLine(ReverbScDl_t *lp, int n);
@@ -138,12 +162,21 @@ private:
     bool initialised = false;
     ReverbScDl_t delay_lines_[8];
     float32_t *aux_; // main delay line storage buffer, placed either in RAM2 or PSRAM
+	const uint32_t aux_size_bytes = REVERBSC_DLYBUF_SIZE*sizeof(float32_t);
 	float32_t dry_gain = 0.5f;
 	float32_t wet_gain = 0.5f;
 
+	float32_t input_gain_set = 0.5f;
 	float32_t input_gain = 0.5f;
 	float32_t input_gain_tmp = 0.5f;
 	float32_t freeze_ingain = 0.05f;
 	static constexpr float32_t feedb_max = 0.99f;
+
+	bool memCleanup(void);
+	const uint32_t memCleanupStep = 512;
+	uint32_t memCleanupStart = 0;
+	uint32_t memCleanupEnd = memCleanupStep;
+
+
 };
 #endif // _EFFECT_REVERBSC_H_

src/effect_springreverb_F32.cpp

@@ -39,7 +39,7 @@
 
 AudioEffectSpringReverb_F32::AudioEffectSpringReverb_F32() : AudioStream_F32(2, inputQueueArray)
 {
-    input_attn = 0.5f;
+    inputGain = 0.5f;
 	rv_time_k = 0.8f;
     in_allp_k = INP_ALLP_COEFF;
 	bool memOK = true;
@@ -94,7 +94,7 @@ void AudioEffectSpringReverb_F32::update()
     if (!initialized) return;
     if (bp)
     {
-		if (!cleanup_done)
+		if (!cleanup_done && bp_mode != BYPASS_MODE_TRAILS)
 		{
 			sp_lp_allp1a.reset();
 			sp_lp_allp1b.reset();
@@ -113,8 +113,9 @@ void AudioEffectSpringReverb_F32::update()
 			cleanup_done = true;
 		}
 
-        if (dry_gain > 0.0f) 		// if dry/wet mixer is used
+		switch(bp_mode)
 		{
+			case BYPASS_MODE_PASS:
 				blockL = AudioStream_F32::receiveReadOnly_f32(0);
 				blockR = AudioStream_F32::receiveReadOnly_f32(1);
 				if (!blockL || !blockR) 
@@ -128,7 +129,8 @@ void AudioEffectSpringReverb_F32::update()
 				AudioStream_F32::release(blockL);
 				AudioStream_F32::release(blockR);
 				return;
-		}
+				break;
+			case BYPASS_MODE_OFF:
 				blockL = AudioStream_F32::allocate_f32();
 				if (!blockL) return;
 				memset(&blockL->data[0], 0, blockL->length*sizeof(float32_t));
@@ -136,6 +138,11 @@ void AudioEffectSpringReverb_F32::update()
 				AudioStream_F32::transmit(blockL, 1);
 				AudioStream_F32::release(blockL);	
 				return;
+				break;
+			case BYPASS_MODE_TRAILS:
+			default:
+				break;
+		}
     }
 	cleanup_done = false;
 	blockL = AudioStream_F32::receiveWritable_f32(0);
@@ -152,13 +159,14 @@ void AudioEffectSpringReverb_F32::update()
 	for (i=0; i < blockL->length; i++) 
     {  
 		lfo.update();
+		inputGain += (inputGainSet - inputGain) * 0.25f;
 		dryL = blockL->data[i];
 		dryR = blockR->data[i];
-		dry_in = (dryL + dryR) * input_attn;
+		dry_in = (dryL + dryR) * inputGain;
 
-		mono_in = flt_in.process(dry_in)* (1.0f + in_BassCut_k*-2.5f); // add highpass gain compaensation?
+		mono_in = flt_in.process(dry_in)* (1.0f + in_BassCut_k*-2.5f);
-		acc = lp_dly1.getTap(0) * rv_time;	// get DLY1 output
+		acc = lp_dly1.getTap(0) * rv_time;
-		lp_out1 = flt_lp1.process(acc);			// filter it
+		lp_out1 = flt_lp1.process(acc);
 
 		acc = sp_lp_allp1a.process(lp_out1); 
 		acc = sp_lp_allp1b.process(acc);

src/effect_springreverb_F32.h

@@ -69,10 +69,11 @@ public:
     {
         n = constrain(n, 0.0f, 1.0f);
         n = map (n, 0.0f, 1.0f, 0.7f, rv_time_k_max);
-        float32_t attn = map(n, 0.0f, rv_time_k_max, 0.5f, 0.2f);
+        float32_t gain = map(n, 0.0f, rv_time_k_max, 0.5f, 0.2f);
+		inputGain_tmp = gain;
         __disable_irq();
         rv_time_k = n;
-        input_attn = attn;
+        inputGainSet = gain;
         __enable_irq();
     }
 
@@ -119,23 +120,46 @@ public:
 		__enable_irq();
     }
     float32_t get_size(void) {return rv_time_k;}
+
+	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) {bp = state;}
+    void bypass_set(bool state) 
+	{
+		bp = state;
+		if (bp && bp_mode==BYPASS_MODE_TRAILS)
+		{
+			inputGainSet = 0.0f;
+		}
+		if (bp == false) inputGainSet = inputGain_tmp;
+	}
     bool bypass_tgl(void) 
     {
-        bp ^= 1; 
+		bypass_set(bp^1);
         return bp;
     } 
 private:
     audio_block_f32_t *inputQueueArray[2];
 
-    float32_t input_attn;  
+	float32_t inputGainSet = 0.5f;
+    float32_t inputGain = 0.5f;
+	float32_t inputGain_tmp = 0.5f;
     float32_t  wet_gain;
     float32_t  dry_gain;
     float32_t in_allp_k; // input allpass coeff (default 0.6)
     float32_t chrp_allp_k[4] = {-0.7f, -0.65f, -0.6f, -0.5f};
 
     bool bp = false;
+	bypass_mode_t bp_mode = BYPASS_MODE_PASS;
 	bool cleanup_done = false;
     uint16_t chrp_alp1_idx[SPRVB_CHIRP_AMNT] = {0};
     uint16_t chrp_alp2_idx[SPRVB_CHIRP_AMNT] = {0};
@@ -168,6 +192,7 @@ private:
 	float32_t lp_TrebleCut_k;
 	float32_t lp_BassCut_k;
 	
+
 	AudioFilterShelvingLPHP flt_in;
 	AudioFilterShelvingLPHP flt_lp1;
 	AudioFilterShelvingLPHP flt_lp2;

src/effect_xfaderStereo_F32.h

@@ -0,0 +1,114 @@
+/**
+ * @file effect_xfaderStereo_F32.h
+ * @author Piotr Zapart
+ * @brief constant power crossfader for two stereo signals
+ * @version 0.1
+ * @date 2024-03-21
+ * 
+ * @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 _EFFECT_XFADERSTEREO_F32_H_
+#define _EFFECT_XFADERSTEREO_F32_H_
+
+#include <arm_math.h> 
+#include <AudioStream_F32.h>
+#include "basic_components.h"
+class AudioEffectXfaderStereo_F32 : public AudioStream_F32
+{
+public:
+	AudioEffectXfaderStereo_F32(void) : AudioStream_F32(4, inputQueueArray_f32){};
+	AudioEffectXfaderStereo_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32){};
+	void mix(float32_t m)
+	{
+		float32_t gA, gB;
+		m = constrain(m, 0.0f, 1.0f);
+		mix_pwr(m, &gB, &gA);
+		__disable_irq()
+		gainA = gA;
+		gainB = gB;
+		__enable_irq();
+	}
+	void update()
+	{
+		audio_block_f32_t *blockLa, *blockRa, *blockLb, *blockRb;
+		audio_block_f32_t *blockOutLa, *blockOutRa,*blockOutLb, *blockOutRb;
+		blockLa = AudioStream_F32::receiveReadOnly_f32(0);
+		blockRa = AudioStream_F32::receiveReadOnly_f32(1);
+		blockLb = AudioStream_F32::receiveReadOnly_f32(2);
+		blockRb = AudioStream_F32::receiveReadOnly_f32(3);
+
+		if (!blockLa || !blockRa || !blockLb || !blockRb)
+		{
+			if (blockLa) AudioStream_F32::release(blockLa);
+			if (blockRa)  AudioStream_F32::release(blockRa);
+			if (blockLb) AudioStream_F32::release(blockLb);
+			if (blockRb)  AudioStream_F32::release(blockRb);			
+			return;
+		}
+		// max A, B mited
+		if (gainA == 1.0f)
+		{
+			AudioStream_F32::transmit(blockLa, 0);
+			AudioStream_F32::transmit(blockRa, 1);
+			AudioStream_F32::release(blockLa);
+			AudioStream_F32::release(blockRa);
+			AudioStream_F32::release(blockLb);
+			AudioStream_F32::release(blockRb);
+			return;
+		}
+		if (gainB == 1.0f)
+		{
+			AudioStream_F32::transmit(blockLb, 0);
+			AudioStream_F32::transmit(blockRb, 1);
+			AudioStream_F32::release(blockLa);
+			AudioStream_F32::release(blockRa);
+			AudioStream_F32::release(blockLb);
+			AudioStream_F32::release(blockRb);
+			return;
+		}
+		blockOutLa = AudioStream_F32::allocate_f32();
+		blockOutRa = AudioStream_F32::allocate_f32();
+		blockOutLb = AudioStream_F32::allocate_f32();
+		blockOutRb = AudioStream_F32::allocate_f32();
+		if (!blockOutLa || !blockOutRa || !blockOutLa || !blockOutRa)
+		{
+			if (blockOutLa) AudioStream_F32::release(blockOutLa);
+			if (blockOutRa)  AudioStream_F32::release(blockOutRa);
+			if (blockOutLb) AudioStream_F32::release(blockOutLb);
+			if (blockOutRb)  AudioStream_F32::release(blockOutRb);		
+			return;
+		}	
+
+		arm_scale_f32(blockLa->data, gainA, blockOutLa->data, blockOutLa->length);
+		arm_scale_f32(blockRa->data, gainA, blockOutRa->data, blockOutRa->length);
+		arm_scale_f32(blockLb->data, gainB, blockOutLb->data, blockOutLb->length);
+		arm_scale_f32(blockRb->data, gainB, blockOutRb->data, blockOutRb->length);
+		arm_add_f32(blockOutLa->data, blockOutLb->data, blockOutLa->data, blockOutLa->length);
+		arm_add_f32(blockOutRa->data, blockOutRb->data, blockOutRa->data, blockOutRa->length);
+		AudioStream_F32::transmit(blockOutLa, 0);
+		AudioStream_F32::transmit(blockOutRa, 1);
+		AudioStream_F32::release(blockLa);
+		AudioStream_F32::release(blockRa);
+		AudioStream_F32::release(blockLb);
+		AudioStream_F32::release(blockRb);
+		AudioStream_F32::release(blockOutLa);
+		AudioStream_F32::release(blockOutRa);
+		AudioStream_F32::release(blockOutLb);
+		AudioStream_F32::release(blockOutRb);	
+	}
+private:
+	audio_block_f32_t *inputQueueArray_f32[4];
+	float32_t gainA = 1.0f;
+	float32_t gainB = 0.0f;
+};
+#endif // _EFFECT_XFADERSTEREO_F32_H_

src/filter_3bandeq.h

@@ -15,6 +15,7 @@
 // - Uses 4 first order filters in series, should give 24dB per octave
 // - Now with P4 Denormal fix :)
 //----------------------------------------------------------------------------
+// Teensy OpenAudio_ArduinoLibrary_F32 port: 01.2024 Piotr Zapart www.hexefx.com 
 
 #ifndef _FILTER_3BANDEQ_H_
 #define _FILTER_3BANDEQ_H_
@@ -23,7 +24,6 @@
 #include "AudioStream_F32.h"
 #include "arm_math.h"
 #include "mathDSP_F32.h"
-#include "basic_components.h"
 
 class AudioFilterEqualizer3band_F32 : public AudioStream_F32
 {

src/filter_biquadStereo_F32.cpp

@@ -0,0 +1,59 @@
+#include "filter_biquadStereo_F32.h"
+
+void AudioFilterBiquadStereo_F32::update(void)
+{
+	audio_block_f32_t *blockL, *blockR, *blockOutL, *blockOutR;
+	blockL = AudioStream_F32::receiveWritable_f32(0);
+	blockR = AudioStream_F32::receiveWritable_f32(1);
+
+	// no input signal
+	if (!blockL || !blockR)
+	{
+		if (blockL) AudioStream_F32::release(blockL);
+		if (blockR) AudioStream_F32::release(blockR);
+		return;
+	}
+	if (bp || gain_dry == 1.0f || !doBiquad)
+	{
+		AudioStream_F32::transmit(blockL, 0);	
+		AudioStream_F32::transmit(blockR, 1);
+		AudioStream_F32::release(blockL);
+		AudioStream_F32::release(blockR);
+		return;
+	}
+
+	blockOutL = AudioStream_F32::allocate_f32();
+	blockOutR = AudioStream_F32::allocate_f32();
+	if (!blockOutL || !blockOutR)
+	{
+		if (blockOutL) AudioStream_F32::release(blockOutL);
+		if (blockOutR) AudioStream_F32::release(blockOutR);
+		AudioStream_F32::release(blockL);
+		AudioStream_F32::release(blockR);		
+		return;
+	}
+	arm_biquad_cascade_df1_f32(&iirL_inst, blockL->data, blockOutL->data, blockOutL->length);
+	arm_biquad_cascade_df1_f32(&iirR_inst, blockR->data, blockOutR->data, blockOutR->length);
+	if (gain_wet != 1.0f)	// transmit wet only
+	{
+		arm_scale_f32(blockL->data, gain_dry, blockL->data, blockL->length); 				// dryL * gain_dry
+		arm_scale_f32(blockR->data, gain_dry, blockR->data, blockR->length); 				// dryR * gain_dry
+		arm_scale_f32(blockOutL->data, gain_wet, blockOutL->data, blockOutL->length); 		// wetL * gain_wet
+		arm_scale_f32(blockOutR->data, gain_wet, blockOutR->data, blockOutR->length); 		// wetR * gain_wet	
+		arm_add_f32(blockL->data, blockOutL->data, blockOutL->data, blockOutL->length);	 	// dryL+wetL
+		arm_add_f32(blockR->data, blockOutR->data, blockOutR->data, blockOutR->length);	 	// dryR+wetR
+	}
+	if (makeup_gain != 1.0f)
+	{
+		arm_scale_f32(blockOutL->data, makeup_gain, blockOutL->data, blockOutL->length); 		// wetL * makeup gain
+		arm_scale_f32(blockOutR->data, makeup_gain, blockOutR->data, blockOutR->length); 		// wetR * makeup gain		
+	}
+	AudioStream_F32::transmit(blockOutL, 0);
+	AudioStream_F32::transmit(blockOutR, 1);
+	AudioStream_F32::release(blockOutL);
+	AudioStream_F32::release(blockOutR);
+	AudioStream_F32::release(blockL);
+	AudioStream_F32::release(blockR);
+
+	
+}

src/filter_biquadStereo_F32.h

@@ -0,0 +1,258 @@
+#ifndef _FILTER_BIQUADSTEREO_F32_H_
+#define _FILTER_BIQUADSTEREO_F32_H_
+
+#include "Arduino.h"
+#include "AudioStream_F32.h"
+#include "arm_math.h"
+#include "basic_DSPutils.h"
+
+// Changed Feb 2021
+#define IIR_STEREO_MAX_STAGES 4
+
+class AudioFilterBiquadStereo_F32 : public AudioStream_F32
+{
+	// GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node
+	// GUI: shortName:IIR2
+public:
+	AudioFilterBiquadStereo_F32(uint8_t stages=IIR_STEREO_MAX_STAGES) : AudioStream_F32(2, inputQueueArray), numStagesUsed(stages)
+	{
+		setSampleRate_Hz(AUDIO_SAMPLE_RATE_EXACT);
+		doClassInit();
+	}
+	AudioFilterBiquadStereo_F32(const AudioSettings_F32 &settings, uint8_t stages=IIR_STEREO_MAX_STAGES) : AudioStream_F32(2, inputQueueArray), numStagesUsed(stages)
+	{
+		setSampleRate_Hz(settings.sample_rate_Hz);
+		doClassInit();
+	}
+
+	void doClassInit(void)
+	{
+		memset(&coeff32[0], 0, 5 * IIR_STEREO_MAX_STAGES * sizeof(coeff32[0]));
+		for (int ii = 0; ii < 4; ii++)
+		{
+			coeff32[5 * ii] = 1.0f; // b0 = 1 for pass through
+		}
+		arm_biquad_cascade_df1_init_f32(&iirL_inst, numStagesUsed, &coeff32[0], &stateL_F32[0]);
+		arm_biquad_cascade_df1_init_f32(&iirR_inst, numStagesUsed, &coeff32[0], &stateR_F32[0]);
+		doBiquad = false;  // This is the way to jump over the biquad
+	}
+
+	// Up to 4 stages are allowed.  Coefficients, either by design function
+	// or from direct setCoefficients() need to be added to the double array
+	// and also to the float
+	void setCoefficients(int iStage, double *cf)
+	{
+		if (iStage > numStagesUsed)
+		{
+			if (Serial)
+			{
+				Serial.print("AudioFilterBiquad_F32: setCoefficients:");
+				Serial.println(" *** MaxStages Error");
+			}
+			return;
+		}
+		
+		for (int ii = 0; ii < 5; ii++)
+		{
+			coeff32[ii + 5 * iStage] = (float)cf[ii]; // and of floats
+		}
+
+		doBiquad = true;
+	}
+
+	void end(void)
+	{
+		doBiquad = false;
+	}
+
+	void setSampleRate_Hz(float _fs_Hz) { sampleRate_Hz = _fs_Hz; }
+
+	//  Deprecated
+	void setBlockDC(void)
+	{
+		// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+		// Use matlab to compute the coeff for HP at 40Hz: [b,a]=butter(2,40/(44100/2),'high'); %assumes fs_Hz = 44100
+		double b[] = {8.173653471988667e-01, -1.634730694397733e+00, 8.173653471988667e-01}; // from Matlab
+		double a[] = {1.000000000000000e+00, -1.601092394183619e+00, 6.683689946118476e-01}; // from Matlab
+		setFilterCoeff_Matlab(b, a);
+	}
+
+	void setFilterCoeff_Matlab(double b[], double a[])
+	{ // one stage of N=2 IIR
+		double coeff[5];
+		// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+		// Use matlab to compute the coeff, such as: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
+		coeff[0] = b[0];
+		coeff[1] = b[1];
+		coeff[2] = b[2]; // here are the matlab "b" coefficients
+		coeff[3] = -a[1];
+		coeff[4] = -a[2]; // the DSP needs the "a" terms to have opposite sign vs Matlab
+		setCoefficients(0, coeff);
+	}
+	// Compute common filter functions
+	// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
+	// void setLowpass(uint32_t stage, float frequency, float q = 0.7071) {
+	void setLowpass(int stage, float frequency, float q)
+	{
+		double coeff[5];
+		double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
+		double sinW0 = sin(w0);
+		double alpha = sinW0 / ((double)q * 2.0);
+		double cosW0 = cos(w0);
+		double scale = 1.0 / (1.0 + alpha); // which is equal to 1.0 / a0
+		/* b0 */ coeff[0] = ((1.0 - cosW0) / 2.0) * scale;
+		/* b1 */ coeff[1] = (1.0 - cosW0) * scale;
+		/* b2 */ coeff[2] = coeff[0];
+		/* a1 */ coeff[3] = -(-2.0 * cosW0) * scale;
+		/* a2 */ coeff[4] = -(1.0 - alpha) * scale;
+		setCoefficients(stage, coeff);
+	}
+
+	void setHighpass(uint32_t stage, float frequency, float q)
+	{
+		double coeff[5];
+		double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
+		double sinW0 = sin(w0);
+		double alpha = sinW0 / ((double)q * 2.0);
+		double cosW0 = cos(w0);
+		double scale = 1.0 / (1.0 + alpha);
+		/* b0 */ coeff[0] = ((1.0 + cosW0) / 2.0) * scale;
+		/* b1 */ coeff[1] = -(1.0 + cosW0) * scale;
+		/* b2 */ coeff[2] = coeff[0];
+		/* a1 */ coeff[3] = -(-2.0 * cosW0) * scale;
+		/* a2 */ coeff[4] = -(1.0 - alpha) * scale;
+		setCoefficients(stage, coeff);
+	}
+
+	void setBandpass(uint32_t stage, float frequency, float q)
+	{
+		double coeff[5];
+		double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
+		double sinW0 = sin(w0);
+		double alpha = sinW0 / ((double)q * 2.0);
+		double cosW0 = cos(w0);
+		double scale = 1.0 / (1.0 + alpha);
+		/* b0 */ coeff[0] = alpha * scale;
+		/* b1 */ coeff[1] = 0;
+		/* b2 */ coeff[2] = (-alpha) * scale;
+		/* a1 */ coeff[3] = -(-2.0 * cosW0) * scale;
+		/* a2 */ coeff[4] = -(1.0 - alpha) * scale;
+		setCoefficients(stage, coeff);
+	}
+
+	// frequency in Hz.  q makes the response stay close to 0.0dB until
+	// close to the notch frequency.  q up to 100 or more seem stable.
+	void setNotch(uint32_t stage, float frequency, float q)
+	{
+		double coeff[5];
+		double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
+		double sinW0 = sin(w0);
+		double alpha = sinW0 / ((double)q * 2.0);
+		double cosW0 = cos(w0);
+		double scale = 1.0 / (1.0 + alpha); // which is equal to 1.0 / a0
+		/* b0 */ coeff[0] = scale;
+		/* b1 */ coeff[1] = (-2.0 * cosW0) * scale;
+		/* b2 */ coeff[2] = coeff[0];
+		/* a1 */ coeff[3] = -(-2.0 * cosW0) * scale;
+		/* a2 */ coeff[4] = -(1.0 - alpha) * scale;
+		setCoefficients(stage, coeff);
+	}
+
+	void setLowShelf(uint32_t stage, float frequency, float gain, float slope)
+	{
+		double coeff[5];
+		double a = pow(10.0, gain / 40.0);
+		double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
+		double sinW0 = sin(w0);
+		// double alpha = (sinW0 * sqrt((a+1/a)*(1/slope-1)+2) ) / 2.0;
+		double cosW0 = cos(w0);
+		// generate three helper-values (intermediate results):
+		double sinsq = sinW0 * sqrt((pow(a, 2.0) + 1.0) * (1.0 / slope - 1.0) + 2.0 * a);
+		double aMinus = (a - 1.0) * cosW0;
+		double aPlus = (a + 1.0) * cosW0;
+		double scale = 1.0 / ((a + 1.0) + aMinus + sinsq);
+		/* b0 */ coeff[0] = a * ((a + 1.0) - aMinus + sinsq) * scale;
+		/* b1 */ coeff[1] = 2.0 * a * ((a - 1.0) - aPlus) * scale;
+		/* b2 */ coeff[2] = a * ((a + 1.0) - aMinus - sinsq) * scale;
+		/* a1 */ coeff[3] = 2.0 * ((a - 1.0) + aPlus) * scale;
+		/* a2 */ coeff[4] = -((a + 1.0) + aMinus - sinsq) * scale;
+		setCoefficients(stage, coeff);
+	}
+
+	void setHighShelf(uint32_t stage, float frequency, float gain, float slope)
+	{
+		double coeff[5];
+		double a = pow(10.0, gain / 40.0);
+		double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
+		double sinW0 = sin(w0);
+		// double alpha = (sinW0 * sqrt((a+1/a)*(1/slope-1)+2) ) / 2.0;
+		double cosW0 = cos(w0);
+		// generate three helper-values (intermediate results):
+		double sinsq = sinW0 * sqrt((pow(a, 2.0) + 1.0) * (1.0 / slope - 1.0) + 2.0 * a);
+		double aMinus = (a - 1.0) * cosW0;
+		double aPlus = (a + 1.0) * cosW0;
+		double scale = 1.0 / ((a + 1.0) - aMinus + sinsq);
+		/* b0 */ coeff[0] = a * ((a + 1.0) + aMinus + sinsq) * scale;
+		/* b1 */ coeff[1] = -2.0 * a * ((a - 1.0) + aPlus) * scale;
+		/* b2 */ coeff[2] = a * ((a + 1.0) + aMinus - sinsq) * scale;
+		/* a1 */ coeff[3] = -2.0 * ((a - 1.0) - aPlus) * scale;
+		/* a2 */ coeff[4] = -((a + 1.0) - aMinus - sinsq) * scale;
+		setCoefficients(stage, coeff);
+	}
+	void update(void);
+
+	void mix(float m)
+	{
+		float g_wet, g_dry;
+		m = constrain(m, 0.0f, 1.0f);
+		mix_pwr(m, &g_wet, &g_dry);
+		__disable_irq();
+		gain_wet = g_wet;
+		gain_dry = g_dry;
+		__enable_irq();
+	}
+	void makeupGain(float g)
+	{
+		__disable_irq();
+		makeup_gain = g;
+		__enable_irq();		
+	}
+
+ 	void bypass_set(bool state) 
+	{
+		__disable_irq();
+		bp = state;
+		__enable_irq();
+	}
+    bool bypass_tgl(void) 
+    {
+		bool bp_new = bp ^ 1;
+		__disable_irq();
+        bp  = bp_new; 
+		__enable_irq();
+        return bp;
+    }
+private:
+	audio_block_f32_t *inputQueueArray[2];
+	bool bp = false;
+	float coeff32[5 * IIR_STEREO_MAX_STAGES]; // Local copies to be transferred with begin()
+	float stateL_F32[4 * IIR_STEREO_MAX_STAGES];
+	float stateR_F32[4 * IIR_STEREO_MAX_STAGES];
+	float sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; // default.  from AudioStream.h??
+	const uint8_t numStagesUsed;
+	bool doBiquad = false;
+	float gain_dry = 0.0f;
+	float gain_wet = 1.0f;
+	float makeup_gain = 1.0f;
+
+	/* Info - The structure from arm_biquad_casd_df1_inst_f32 consists of
+	 *    uint32_t  numStages;
+	 *    const float32_t *pCoeffs;  //Points to the array of coefficients, length 5*numStages.
+	 *    float32_t *pState;         //Points to the array of state variables, length 4*numStages.
+	 */
+	// ARM DSP Math library filter instance.
+	arm_biquad_casd_df1_inst_f32 iirL_inst;
+	arm_biquad_casd_df1_inst_f32 iirR_inst;
+};
+
+#endif // _FILTER_BIQUADSTEREO_F32_H_

src/filter_equalizer_F32.h

@@ -113,9 +113,7 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
  * SOFTWARE.
  *
- * 01.2024 - added
+ * 01.2024 - added bypass system, Piot rZapart www.hexefx.com
- *
- *
  */
 
 #ifndef _FILTEREQUALIZER_F32_H_

src/filter_ir_cabsim_F32.cpp

@@ -56,7 +56,30 @@ void AudioFilterIRCabsim_F32::update()
 		if (blockR) AudioStream_F32::release(blockR);
 		return;
 	}
+#ifdef USE_IR_ISR_LOAD
+	switch(ir_loadState)
+	{
+		case IR_LOAD_START:
+			ptr_fmask = &fmask[0][0];
+			ptr_fftout = &fftout[0];
+			memset(ptr_fftout, 0, nfor*512*4);  // clear fftout array
+			memset(fftin, 0,  512 * 4);  // clear fftin array
+			ir_loadState = IR_LOAD_STEP1;	
+			break;
+		case IR_LOAD_STEP1:
+			init_partitioned_filter_masks(irPtrTable[ir_idx]);
+			ir_loadState = IR_LOAD_STEP2;
+			break;
+		case IR_LOAD_STEP2:
+			delay.reset();
+			ir_loaded = 1;
+			ir_loadState = IR_LOAD_FINISHED;
+			break;
+		case IR_LOAD_FINISHED:
+		default: break;
+	}
 
+#endif
 	if (!ir_loaded) // ir not loaded yet or bypass mode
 	{
 		// bypass clean signal
@@ -66,6 +89,7 @@ void AudioFilterIRCabsim_F32::update()
 		AudioStream_F32::release(blockR);
 		return;
 	}
+
 	if (first_block) // fill real & imaginaries with zeros for the first BLOCKSIZE samples
 	{
 		memset(&fftin[0], 0, blockL->length*sizeof(float32_t)*4);
@@ -136,7 +160,6 @@ void AudioFilterIRCabsim_F32::update()
 	// apply post EQ, restore the channel R phase, reduce the gain a bit
 	if (doubleTrack)  
 	{
-		
 		arm_fir_f32(&FIR_postL, blockL->data, blockL->data, blockL->length);
 		arm_fir_f32(&FIR_postR, blockR->data, blockR->data, blockR->length);
 		arm_scale_f32(blockR->data, -doubler_gainR, blockR->data, blockR->length);
@@ -156,6 +179,7 @@ void AudioFilterIRCabsim_F32::ir_register(const float32_t *irPtr, uint8_t positi
 	irPtrTable[position] = irPtr;
 }
 
+
 void AudioFilterIRCabsim_F32::ir_load(uint8_t idx)
 {
 	const float32_t *newIrPtr = NULL;
@@ -173,6 +197,17 @@ void AudioFilterIRCabsim_F32::ir_load(uint8_t idx)
 	{
 		return;
 	}
+#ifdef USE_IR_ISR_LOAD
+	nc = newIrPtr[0];
+	uint32_t _nfor = nc / IR_BUFFER_SIZE;
+	if (_nfor > nforMax) _nfor = nforMax;
+	__disable_irq()
+	nfor = _nfor;
+	ir_loadState = IR_LOAD_START;
+	__enable_irq();
+	ir_length_ms =  (1000.0f * nfor * (float32_t)AUDIO_BLOCK_SAMPLES) / AUDIO_SAMPLE_RATE_EXACT;
+#else
+	
 	AudioNoInterrupts();
 	nc = newIrPtr[0];
 	nfor = nc / IR_BUFFER_SIZE;
@@ -188,8 +223,11 @@ void AudioFilterIRCabsim_F32::ir_load(uint8_t idx)
 	delay.reset();
 	ir_loaded = 1;
 	AudioInterrupts();
+	#endif
+
 }
 
+
 void AudioFilterIRCabsim_F32::init_partitioned_filter_masks(const float32_t *irPtr)
 {
 	const static arm_cfft_instance_f32 *maskS;

src/filter_ir_cabsim_F32.h

@@ -41,6 +41,9 @@
 #define IR_N_B          (1)
 #define IR_MAX_REG_NUM  11       // max number of registered IRs
 
+
+#define USE_IR_ISR_LOAD
+
 class AudioFilterIRCabsim_F32 : public AudioStream_F32
 {
 public:
@@ -155,6 +158,17 @@ private:
 		 0.154734746f,     0.35352844f,     0.441179603f,     0.35352844f,      0.154734746f,   -0.0208595414f,
 		-0.0834295526f,   -0.0470990688f,   0.0108086104f,    0.0319586769f,    0.0158470348f,  -0.00560652791f, 
 		-0.0112718018f,   -0.00476867426f,  0.0013392308f,    0.00207542209f,   0.000503875781f, 0.0f };
+
+
+		typedef enum
+		{
+			IR_LOAD_START,
+			IR_LOAD_STEP1,
+			IR_LOAD_STEP2,
+			IR_LOAD_FINISHED
+		}ir_loadState_t;
+		ir_loadState_t ir_loadState = IR_LOAD_FINISHED;
+
 }; 
 
 

src/hexefx_audio_F32.h

@@ -1,13 +1,19 @@
 #ifndef _HEXEFX_AUDIO_H
 #define _HEXEFX_AUDIO_H
 
+#include "control_WM8731_F32.h"
+#include "control_SGTL5000_F32.h"
+
 #include "input_i2s2_F32.h"
 #include "output_i2s2_F32.h"
 
+#include "switch_selectorStereo_F32.h"
+
 #include "filter_ir_cabsim_F32.h"
 #include "filter_tonestackStereo_F32.h"
 #include "filter_equalizer_F32.h"
 #include "filter_3bandeq.h"
+#include "filter_biquadStereo_F32.h"
 
 #include "effect_gainStereo_F32.h"
 #include "effect_platereverb_F32.h"
@@ -17,6 +23,9 @@
 #include "effect_infphaser_F32.h"
 #include "effect_phaserStereo_F32.h"
 #include "effect_noiseGateStereo_F32.h"
-#include "effect_delaystereo.h"
+#include "effect_delaystereo_F32.h"
+#include "effect_compressorStereo_F32.h"
+#include "effect_guitarBooster_F32.h"
+#include "effect_xfaderStereo_F32.h"
 
 #endif // _HEXEFX_AUDIO_H

src/input_i2s2_F32.cpp

@@ -36,11 +36,11 @@
 #include <Arduino.h> //do we really need this? (Chip: 2020-10-31)
 #include "input_i2s2_F32.h"
 #include "output_i2s2_F32.h"
-
+#include "basic_DSPutils.h"
 #include <arm_math.h>
 
 // DMAMEM __attribute__((aligned(32)))
-static uint32_t i2s2_rx_buffer[AUDIO_BLOCK_SAMPLES]; //good for 16-bit audio samples coming in from teh AIC.  32-bit transfers will need this to be bigger.
+static uint64_t i2s2_rx_buffer[AUDIO_BLOCK_SAMPLES] __attribute__((aligned(32))); // good for 16-bit audio samples coming in from teh AIC.  32-bit transfers will need this to be bigger.
 audio_block_f32_t *AudioInputI2S2_F32::block_left_f32 = NULL;
 audio_block_f32_t *AudioInputI2S2_F32::block_right_f32 = NULL;
 uint16_t AudioInputI2S2_F32::block_offset = 0;
@@ -53,144 +53,100 @@ unsigned long AudioInputI2S2_F32::update_counter = 0;
 float AudioInputI2S2_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE;
 int AudioInputI2S2_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES;
 
-//#for 16-bit transfers
 #define I2S2_BUFFER_TO_USE_BYTES (AudioOutputI2S2_F32::audio_block_samples * sizeof(i2s2_rx_buffer[0]))
 
-//#for 32-bit transfers
+// --------------------------------------------------------------------------------
-//#define I2S2_BUFFER_TO_USE_BYTES (AudioOutputI2S_F32::audio_block_samples*2*sizeof(i2s_rx_buffer[0]))
+void AudioInputI2S2_F32::begin()
-
+{
-void AudioInputI2S2_F32::begin(void) {
-	bool transferUsing32bit = false;
-	begin(transferUsing32bit);
-}
-
-void AudioInputI2S2_F32::begin(bool transferUsing32bit) {
 	dma.begin(true); // Allocate the DMA channel first
 
 	AudioOutputI2S2_F32::sample_rate_Hz = sample_rate_Hz;			// these were given in the AudioSettings in the contructor
 	AudioOutputI2S2_F32::audio_block_samples = audio_block_samples; // these were given in the AudioSettings in the contructor
 
-	//block_left_1st = NULL;
-	//block_right_1st = NULL;
-
 	// TODO: should we set & clear the I2S_RCSR_SR bit here?
-	AudioOutputI2S2_F32::config_i2s(transferUsing32bit);
+	AudioOutputI2S2_F32::config_i2s();
 
-#if defined(__IMXRT1062__)
 	CORE_PIN5_CONFIG = 2;				   // EMC_08, 2=SAI2_RX_DATA, page 434
 	IOMUXC_SAI2_RX_DATA0_SELECT_INPUT = 0; // 0=GPIO_EMC_08_ALT2, page 876
 
-	dma.TCD->SADDR = (void *)((uint32_t)&I2S2_RDR0 + 2);
+	dma.TCD->SADDR = (void *)((uint32_t)&I2S2_RDR0);
 	dma.TCD->SOFF = 0;
-	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
+	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2);
-	dma.TCD->NBYTES_MLNO = 2;
+	dma.TCD->NBYTES_MLNO = 4;
 	dma.TCD->SLAST = 0;
 	dma.TCD->DADDR = i2s2_rx_buffer;
-	dma.TCD->DOFF = 2;
+	dma.TCD->DOFF = 4;
-	//dma.TCD->CITER_ELINKNO = sizeof(i2s_rx_buffer) / 2; //original from Teensy Audio Library
+	dma.TCD->CITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 4;
-	dma.TCD->CITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 2;
-	//dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer); //original from Teensy Audio Library
 	dma.TCD->DLASTSGA = -I2S2_BUFFER_TO_USE_BYTES;
-	//dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_buffer) / 2; //original from Teensy Audio Library
+	dma.TCD->BITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 4;
-	dma.TCD->BITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 2;
 	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI2_RX);
 
 	I2S2_RCSR = I2S_RCSR_RE | I2S_RCSR_BCE | I2S_RCSR_FRDE | I2S_RCSR_FR; // page 2099
-	I2S2_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE; // page 2087
+
-#endif
 	update_responsibility = update_setup();
 	dma.enable();
 	dma.attachInterrupt(isr);
 
 	update_counter = 0;
 }
-
+// ------------------------------ RX DMA ISR ------------------------------------------
-
 void AudioInputI2S2_F32::isr(void)
 {
 	uint32_t daddr, offset;
-	const int16_t *src, *end;
+	const int32_t *src, *end;
-	//int16_t *dest_left, *dest_right;
-	//audio_block_t *left, *right;
 	float32_t *dest_left_f32, *dest_right_f32;
 	audio_block_f32_t *left_f32, *right_f32;
 
-#if defined(KINETISK) || defined(__IMXRT1062__)
 	daddr = (uint32_t)(dma.TCD->DADDR);
-#endif
 	dma.clearInterrupt();
-	//Serial.println("isr");
+	if (daddr < (uint32_t)i2s2_rx_buffer + I2S2_BUFFER_TO_USE_BYTES / 2)
-
+	{
-	//if (daddr < (uint32_t)i2s_rx_buffer + sizeof(i2s_rx_buffer) / 2) { //original Teensy Audio Library
-	if (daddr < (uint32_t)i2s2_rx_buffer + I2S2_BUFFER_TO_USE_BYTES / 2) {
 		// DMA is receiving to the first half of the buffer
 		// need to remove data from the second half
-		//src = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original Teensy Audio Library
+		src = (int32_t *)&i2s2_rx_buffer[audio_block_samples / 2];
-		//end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES]; //original Teensy Audio Library
+		end = (int32_t *)&i2s2_rx_buffer[audio_block_samples];
-		src = (int16_t *)&i2s2_rx_buffer[audio_block_samples/2];
-		end = (int16_t *)&i2s2_rx_buffer[audio_block_samples];			
 		update_counter++; // let's increment the counter here to ensure that we get every ISR resulting in audio
 		if (AudioInputI2S2_F32::update_responsibility) AudioStream_F32::update_all();
-	} else {
+	}
+	else
+	{
 		// DMA is receiving to the second half of the buffer
 		// need to remove data from the first half
-		src = (int16_t *)&i2s2_rx_buffer[0];
+		src = (int32_t *)&i2s2_rx_buffer[0];
-		//end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original Teensy Audio Library
+		end = (int32_t *)&i2s2_rx_buffer[audio_block_samples / 2];
-		end = (int16_t *)&i2s2_rx_buffer[audio_block_samples/2];
 	}
 	left_f32 = AudioInputI2S2_F32::block_left_f32;
 	right_f32 = AudioInputI2S2_F32::block_right_f32;
-	if (left_f32 != NULL && right_f32 != NULL) {
+	if (left_f32 != NULL && right_f32 != NULL)
+	{
 		offset = AudioInputI2S2_F32::block_offset;
-		//if (offset <= (uint32_t)(AUDIO_BLOCK_SAMPLES/2)) { //original Teensy Audio Library
+		if (offset <= ((uint32_t)audio_block_samples / 2))
-		if (offset <= ((uint32_t) audio_block_samples/2)) {
+		{
 			dest_left_f32 = &(left_f32->data[offset]);
 			dest_right_f32 = &(right_f32->data[offset]);
-			//AudioInputI2S2_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES/2; //original Teensy Audio Library
 			AudioInputI2S2_F32::block_offset = offset + audio_block_samples / 2;
-			do {
+			do
-				//Serial.println(*src);
+			{
-				//n = *src++;
-				//*dest_left++ = (int16_t)n;
-				//*dest_right++ = (int16_t)(n >> 16);
 				*dest_left_f32++ = (float32_t)*src++;
 				*dest_right_f32++ = (float32_t)*src++;
 			} while (src < end);
 		}
 	}
 }
-
+// --------------------------------------------------------------------------------
-#define I16_TO_F32_NORM_FACTOR (3.051850947599719e-05)  //which is 1/32767 
+void AudioInputI2S2_F32::update_1chan(int chan, audio_block_f32_t *&out_f32)
-void AudioInputI2S2_F32::scale_i16_to_f32( float32_t *p_i16, float32_t *p_f32, int len) {
+{
-	for (int i=0; i<len; i++) { *p_f32++ = ((*p_i16++) * I16_TO_F32_NORM_FACTOR); }
+	if (!out_f32)
-}
+		return;
-#define I24_TO_F32_NORM_FACTOR (1.192093037616377e-07)   //which is 1/(2^23 - 1)
-void AudioInputI2S2_F32::scale_i24_to_f32( float32_t *p_i24, float32_t *p_f32, int len) {
-	for (int i=0; i<len; i++) { *p_f32++ = ((*p_i24++) * I24_TO_F32_NORM_FACTOR); }
-}
-#define I32_TO_F32_NORM_FACTOR (4.656612875245797e-10)   //which is 1/(2^31 - 1)
-void AudioInputI2S2_F32::scale_i32_to_f32( float32_t *p_i32, float32_t *p_f32, int len) {
-	for (int i=0; i<len; i++) { *p_f32++ = ((*p_i32++) * I32_TO_F32_NORM_FACTOR); }
-}
-
- void AudioInputI2S2_F32::update_1chan(int chan, audio_block_f32_t *&out_f32) {
-	 if (!out_f32) return;
-	 
 	// scale the float values so that the maximum possible audio values span -1.0 to + 1.0
-	//scale_i32_to_f32(out_f32->data, out_f32->data, audio_block_samples);
+	arm_scale_f32(out_f32->data, I32_TO_F32_NORM_FACTOR, out_f32->data, audio_block_samples);
-	scale_i16_to_f32(out_f32->data, out_f32->data, audio_block_samples);
-
 	// prepare to transmit by setting the update_counter (which helps tell if data is skipped or out-of-order)
 	out_f32->id = update_counter;
-		
+	AudioStream_F32::transmit(out_f32, chan);	// transmit the f32 data!
-	//transmit the f32 data!
+	AudioStream_F32::release(out_f32);			// release the memory blocks
-	AudioStream_F32::transmit(out_f32,chan);
-
-	//release the memory blocks
-	AudioStream_F32::release(out_f32);
 }
- 
+// --------------------------------------------------------------------------------
 void AudioInputI2S2_F32::update(void)
 {
 	static bool flag_beenSuccessfullOnce = false;
@@ -198,19 +154,22 @@ void AudioInputI2S2_F32::update(void)
 
 	new_left = AudioStream_F32::allocate_f32();
 	new_right = AudioStream_F32::allocate_f32();
-	if ((!new_left) || (!new_right)) {
+	if ((!new_left) || (!new_right))
+	{
 		// ran out of memory.  Clear and return!
 		if (new_left) 	AudioStream_F32::release(new_left);
 		if (new_right)	AudioStream_F32::release(new_right);
-		new_left = NULL;  new_right = NULL;
+		new_left = NULL;
+		new_right = NULL;
 		flag_out_of_memory = 1;
-		if (flag_beenSuccessfullOnce) Serial.println("Input_I2S_F32: update(): WARNING!!! Out of Memory.");
+		if (flag_beenSuccessfullOnce)
-	} else {
+			Serial.println("Input_I2S_F32: update(): WARNING!!! Out of Memory.");
-		flag_beenSuccessfullOnce = true;
 	}
+	else {flag_beenSuccessfullOnce = true; }
 
 	__disable_irq();
-	if (block_offset >= audio_block_samples) {
+	if (block_offset >= audio_block_samples)
+	{
 		// the DMA filled 2 blocks, so grab them and get the
 		// 2 new blocks to the DMA, as quickly as possible
 		out_left = block_left_f32;
@@ -219,28 +178,31 @@ void AudioInputI2S2_F32::update(void)
 		block_right_f32 = new_right;
 		block_offset = 0;
 		__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(1, out_right); // uses audio_block_samples and update_counter
-		
+	}
-		
+	else if (new_left != NULL)
-	} else if (new_left != NULL) {
+	{
 		// the DMA didn't fill blocks, but we allocated blocks
-		if (block_left_f32 == NULL) {
+		if (block_left_f32 == NULL)
+		{
 			// the DMA doesn't have any blocks to fill, so
 			// give it the ones we just allocated
 			block_left_f32 = new_left;
 			block_right_f32 = new_right;
 			block_offset = 0;
 			__enable_irq();
-		} else {
+		}
+		else
+		{
 			// the DMA already has blocks, doesn't need these
 			__enable_irq();
 			AudioStream_F32::release(new_left);
 			AudioStream_F32::release(new_right);
 		}
-	} else {
+	}
+	else
+	{
 		// The DMA didn't fill blocks, and we could not allocate
 		// memory... the system is likely starving for memory!
 		// Sadly, there's nothing we can do.
@@ -248,13 +210,11 @@ void AudioInputI2S2_F32::update(void)
 	}
 }
 
-/******************************************************************/
+// --------------------------------------------------------------------------------
-
 
 void AudioInputI2S2slave_F32::begin(void)
 {
 	dma.begin(true); // Allocate the DMA channel first
 
 	AudioOutputI2S2slave_F32::config_i2s();
-
 }

src/input_i2s2_F32.h

@@ -39,9 +39,10 @@
 #include <Arduino.h>
 #include <arm_math.h> 
 #include "AudioStream_F32.h"
-#include "AudioStream.h"   //Do we really need this?? (Chip, 2020-10-31)
 #include "DMAChannel.h"
 
+
+
 class AudioInputI2S2_F32 : public AudioStream_F32
 {
 //GUI: inputs:0, outputs:2  //this line used for automatic generation of GUI nodes
@@ -54,21 +55,15 @@ public:
 	}
 	
 	virtual void update(void);
-	static void scale_i16_to_f32( float32_t *p_i16, float32_t *p_f32, int len) ;
+
-	static void scale_i24_to_f32( float32_t *p_i24, float32_t *p_f32, int len) ;
-	static void scale_i32_to_f32( float32_t *p_i32, float32_t *p_f32, int len);
 	void begin(void);
-	void begin(bool);
-	void sub_begin_i32(void);
-	//void sub_begin_i16(void);
 	int get_isOutOfMemory(void) { return flag_out_of_memory; }
 	void clear_isOutOfMemory(void) { flag_out_of_memory = 0; }
-	//friend class AudioOutputI2S_F32;
+	bool get_update_responsibility() { return update_responsibility;}
 protected:	
 	AudioInputI2S2_F32(int dummy): AudioStream_F32(0, NULL) {} // to be used only inside AudioInputI2Sslave !!
 	static bool update_responsibility;
 	static DMAChannel dma;
-	static void isr_32(void);
 	static void isr(void);
 	virtual void update_1chan(int, audio_block_f32_t *&);
 private:
@@ -86,7 +81,6 @@ class AudioInputI2S2slave_F32 : public AudioInputI2S2_F32
 public:
 	AudioInputI2S2slave_F32(void) : AudioInputI2S2_F32(0) { begin(); }
 	void begin(void);
-	friend void dma_ch1_isr(void);
 };
 
 #endif // _INPUT_I2S_F32_H_

src/output_i2s2_F32.cpp

@@ -35,13 +35,8 @@
 // Ported to I2S2, 12.2023 by Piotr Zapart www.hexefx.com - for teensy4.x only!
 
 #include "output_i2s2_F32.h"
-#include <arm_math.h>
+#include "basic_DSPutils.h"
-#include <Audio.h> //to get access to Audio/utlity/imxrt_hw.h...do we really need this??? WEA 2020-10-31
 
-float AudioOutputI2S2_F32::setI2SFreq_T3(const float freq_Hz)
-{
-	return 0.0f;
-}
 
 audio_block_f32_t *AudioOutputI2S2_F32::block_left_1st = NULL;
 audio_block_f32_t *AudioOutputI2S2_F32::block_right_1st = NULL;
@@ -51,7 +46,7 @@ uint16_t AudioOutputI2S2_F32::block_left_offset = 0;
 uint16_t AudioOutputI2S2_F32::block_right_offset = 0;
 bool AudioOutputI2S2_F32::update_responsibility = false;
 DMAChannel AudioOutputI2S2_F32::dma(false);
-DMAMEM __attribute__((aligned(32))) static uint32_t i2s2_tx_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint64_t i2s2_tx_buffer[AUDIO_BLOCK_SAMPLES];
 
 float AudioOutputI2S2_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE;
 int AudioOutputI2S2_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES;
@@ -60,80 +55,66 @@ int AudioOutputI2S2_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES;
 #include <utility/imxrt_hw.h> //from Teensy Audio library.  For set_audioClock()
 #endif
 
-// #for 16-bit transfers
 #define I2S2_BUFFER_TO_USE_BYTES (AudioOutputI2S2_F32::audio_block_samples * sizeof(i2s2_tx_buffer[0]))
-
+//  --------------------------------------------------------------------------------
-// #for 32-bit transfers
-// #define I2S_BUFFER_TO_USE_BYTES (AudioOutputI2S2_F32::audio_block_samples*2*sizeof(i2s_tx_buffer[0]))
-
 void AudioOutputI2S2_F32::begin(void)
 {
-	bool transferUsing32bit = false;
+	bool transferUsing32bit = true;
 	begin(transferUsing32bit);
 }
-
+//  --------------------------------------------------------------------------------
 void AudioOutputI2S2_F32::begin(bool transferUsing32bit)
 {
-
 	dma.begin(true); // Allocate the DMA channel first
-
 	block_left_1st = NULL;
 	block_right_1st = NULL;
-
 	AudioOutputI2S2_F32::config_i2s(transferUsing32bit, sample_rate_Hz);
 
 #if defined(__IMXRT1062__)
 	CORE_PIN2_CONFIG = 2; // EMC_04, 2=SAI2_TX_DATA, page 428
 
 	dma.TCD->SADDR = i2s2_tx_buffer;
-	dma.TCD->SOFF = 2;
+	dma.TCD->SOFF = 4;
-	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
+	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2);
-	dma.TCD->NBYTES_MLNO = 2;
+	dma.TCD->NBYTES_MLNO = 4;
-	// dma.TCD->SLAST = -sizeof(i2s_tx_buffer);//orig from Teensy Audio Library 2020-10-31
 	dma.TCD->SLAST = -I2S2_BUFFER_TO_USE_BYTES;
 	dma.TCD->DOFF = 0;
-	// dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //orig from Teensy Audio Library 2020-10-31
+	dma.TCD->CITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 4;
-	dma.TCD->CITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 2;
 	dma.TCD->DLASTSGA = 0;
-	// dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;//orig from Teensy Audio Library 2020-10-31
+	dma.TCD->BITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 4;
-	dma.TCD->BITER_ELINKNO = I2S2_BUFFER_TO_USE_BYTES / 2;
 	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
-	dma.TCD->DADDR = (void *)((uint32_t)&I2S2_TDR0 + 2);
+	dma.TCD->DADDR = (void *)((uint32_t)&I2S2_TDR0);
 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI2_TX);
 	dma.enable(); // newer location of this line in Teensy Audio library
 
-	// I2S2_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE;
+	I2S2_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE;
-	I2S2_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE | I2S_TCSR_FR;
+	I2S2_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
 #endif
 	update_responsibility = update_setup();
 	dma.attachInterrupt(AudioOutputI2S2_F32::isr);
 	enabled = 1;
 }
-
+//  --------------------------------------------------------------------------------
 void AudioOutputI2S2_F32::isr(void)
 {
-#if defined(KINETISK) || defined(__IMXRT1062__)
+	int32_t *dest;
-	int16_t *dest;
 	audio_block_f32_t *blockL, *blockR;
 	uint32_t saddr, offsetL, offsetR;
 
 	saddr = (uint32_t)(dma.TCD->SADDR);
 	dma.clearInterrupt();
-	// if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {	//original 16-bit
 	if (saddr < (uint32_t)i2s2_tx_buffer + I2S2_BUFFER_TO_USE_BYTES / 2)
 	{ // are we transmitting the first half or second half of the buffer?
 		// DMA is transmitting the first half of the buffer
 		// so we must fill the second half
-		// dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original Teensy Audio
+		dest = (int32_t *)&i2s2_tx_buffer[audio_block_samples / 2];
-		dest = (int16_t *)&i2s2_tx_buffer[audio_block_samples / 2]; // this will be diff if we were to do 32-bit samples
+		if (AudioOutputI2S2_F32::update_responsibility) AudioStream_F32::update_all();
-		if (AudioOutputI2S2_F32::update_responsibility)
-			AudioStream_F32::update_all();
 	}
 	else
 	{
 		// DMA is transmitting the second half of the buffer
 		// so we must fill the first half
-		dest = (int16_t *)i2s2_tx_buffer;
+		dest = (int32_t *)&i2s2_tx_buffer[0];
 	}
 
 	blockL = AudioOutputI2S2_F32::block_left_1st;
@@ -141,52 +122,45 @@ void AudioOutputI2S2_F32::isr(void)
 	offsetL = AudioOutputI2S2_F32::block_left_offset;
 	offsetR = AudioOutputI2S2_F32::block_right_offset;
 
-	int16_t *d = dest;
+	int32_t *d = dest;
 	if (blockL && blockR)
 	{
-		// memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
-		// memcpy_tointerleaveLRwLen(dest, blockL->data + offsetL, blockR->data + offsetR, audio_block_samples/2);
 		float32_t *pL = blockL->data + offsetL;
 		float32_t *pR = blockR->data + offsetR;
 		for (int i = 0; i < audio_block_samples / 2; i++)
 		{
-			*d++ = (int16_t)*pL++;
+			*d++ = (int32_t)*pL++;
-			*d++ = (int16_t)*pR++; // interleave
+			*d++ = (int32_t)*pR++; // interleave
-			//*d++ = 0;
-			//*d++ = 0;
 		}
 		offsetL += audio_block_samples / 2;
 		offsetR += audio_block_samples / 2;
 	}
 	else if (blockL)
 	{
-		// memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
 		float32_t *pL = blockL->data + offsetL;
-		for (int i = 0; i < audio_block_samples / 2 * 2; i += 2)
+		for (int i = 0; i < audio_block_samples; i += 2)
 		{
-			*(d + i) = (int16_t)*pL++;
+			*(d + i) = (int32_t)*pL++;
-		} // interleave
+		}
 		offsetL += audio_block_samples / 2;
 	}
 	else if (blockR)
 	{
 		float32_t *pR = blockR->data + offsetR;
-		for (int i = 0; i < audio_block_samples / 2 * 2; i += 2)
+		for (int i = 0; i < audio_block_samples; i += 2)
 		{
-			*(d + i) = (int16_t)*pR++;
+			*(d + i) = (int32_t)*pR++;
-		} // interleave
+		} 
 		offsetR += audio_block_samples / 2;
 	}
 	else
 	{
-		// memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
+		memset(dest, 0, audio_block_samples * 4);
-		memset(dest, 0, audio_block_samples * 2);
 		return;
 	}
 
 	arm_dcache_flush_delete(dest, sizeof(i2s2_tx_buffer) / 2);
 
-	// if (offsetL < AUDIO_BLOCK_SAMPLES) { //orig Teensy Audio
 	if (offsetL < (uint16_t)audio_block_samples)
 	{
 		AudioOutputI2S2_F32::block_left_offset = offsetL;
@@ -198,7 +172,6 @@ void AudioOutputI2S2_F32::isr(void)
 		AudioOutputI2S2_F32::block_left_1st = AudioOutputI2S2_F32::block_left_2nd;
 		AudioOutputI2S2_F32::block_left_2nd = NULL;
 	}
-	// if (offsetR < AUDIO_BLOCK_SAMPLES) { //orig Teensy Audio
 	if (offsetR < (uint16_t)audio_block_samples)
 	{
 		AudioOutputI2S2_F32::block_right_offset = offsetR;
@@ -210,59 +183,24 @@ void AudioOutputI2S2_F32::isr(void)
 		AudioOutputI2S2_F32::block_right_1st = AudioOutputI2S2_F32::block_right_2nd;
 		AudioOutputI2S2_F32::block_right_2nd = NULL;
 	}
-#endif
-}
-
-#define F32_TO_I16_NORM_FACTOR (32767) // which is 2^15-1
-void AudioOutputI2S2_F32::scale_f32_to_i16(float32_t *p_f32, float32_t *p_i16, int len)
-{
-	for (int i = 0; i < len; i++)
-	{
-		*p_i16++ = max(-F32_TO_I16_NORM_FACTOR, min(F32_TO_I16_NORM_FACTOR, (*p_f32++) * F32_TO_I16_NORM_FACTOR));
-	}
-}
-#define F32_TO_I24_NORM_FACTOR (8388607) // which is 2^23-1
-void AudioOutputI2S2_F32::scale_f32_to_i24(float32_t *p_f32, float32_t *p_i24, int len)
-{
-	for (int i = 0; i < len; i++)
-	{
-		*p_i24++ = max(-F32_TO_I24_NORM_FACTOR, min(F32_TO_I24_NORM_FACTOR, (*p_f32++) * F32_TO_I24_NORM_FACTOR));
-	}
-}
-#define F32_TO_I32_NORM_FACTOR (2147483647) // which is 2^31-1
-// define F32_TO_I32_NORM_FACTOR (8388607)   //which is 2^23-1
-void AudioOutputI2S2_F32::scale_f32_to_i32(float32_t *p_f32, float32_t *p_i32, int len)
-{
-	for (int i = 0; i < len; i++)
-	{
-		*p_i32++ = max(-F32_TO_I32_NORM_FACTOR, min(F32_TO_I32_NORM_FACTOR, (*p_f32++) * F32_TO_I32_NORM_FACTOR));
-	}
-	// for (int i=0; i<len; i++) { *p_i32++ = (*p_f32++) * F32_TO_I32_NORM_FACTOR + 512.f*8388607.f; }
 }
 
+//  --------------------------------------------------------------------------------
 // update has to be carefully coded so that, if audio_blocks are not available, the code exits
 // gracefully and won't hang.  That'll cause the whole system to hang, which would be very bad.
 // static int count = 0;
 void AudioOutputI2S2_F32::update(void)
 {
-	// null audio device: discard all incoming data
-	// if (!active) return;
-	// audio_block_t *block = receiveReadOnly();
-	// if (block) release(block);
-
 	audio_block_f32_t *block_f32;
 	audio_block_f32_t *block_f32_scaled = AudioStream_F32::allocate_f32();
 	audio_block_f32_t *block2_f32_scaled = AudioStream_F32::allocate_f32();
 	if ((!block_f32_scaled) || (!block2_f32_scaled))
 	{
 		// couldn't get some working memory.  Return.
-		if (block_f32_scaled)
+		if (block_f32_scaled)  AudioStream_F32::release(block_f32_scaled);
-			AudioStream_F32::release(block_f32_scaled);
+		if (block2_f32_scaled) AudioStream_F32::release(block2_f32_scaled);
-		if (block2_f32_scaled)
-			AudioStream_F32::release(block2_f32_scaled);
 		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
 	if (block_f32)
@@ -274,22 +212,8 @@ void AudioOutputI2S2_F32::update(void)
 			Serial.print(", but I2S settings want it to be = ");
 			Serial.println(audio_block_samples);
 		}
-		// Serial.print("AudioOutputI2S2_F32: audio_block_samples = ");
-		// Serial.println(audio_block_samples);
-
-		// scale F32 to Int32
-		// block_f32_scaled = AudioStream_F32::allocate_f32();
-		// scale_f32_to_i32(block_f32->data, block_f32_scaled->data, audio_block_samples);
-		scale_f32_to_i16(block_f32->data, block_f32_scaled->data, audio_block_samples);
 
-		// count++;
+		scale_float_to_int32range(block_f32->data, block_f32_scaled->data, audio_block_samples);
-		// if (count > 100) {
-		//	Serial.print("AudioOutputI2S2_F32::update() orig, scaled = ");
-		//	Serial.print(block_f32->data[30]);
-		//	Serial.print(", ");
-		//	Serial.println(block_f32_scaled->data[30]);
-		//	count=0;
-		// }
 
 		// now process the data blocks
 		__disable_irq();
@@ -326,10 +250,8 @@ void AudioOutputI2S2_F32::update(void)
 	block_f32 = receiveReadOnly_f32(1);	  // input 1 = right channel
 	if (block_f32)
 	{
-		// scale F32 to Int32
+		arm_scale_f32(block_f32->data, (float32_t)F32_TO_I32_NORM_FACTOR, 
-		// block_f32_scaled = AudioStream_F32::allocate_f32();
+						block_f32_scaled->data,audio_block_samples);
-		// scale_f32_to_i32(block_f32->data, block_f32_scaled->data, audio_block_samples);
-		scale_f32_to_i16(block_f32->data, block_f32_scaled->data, audio_block_samples);
 
 		__disable_irq();
 		if (block_right_1st == NULL)
@@ -361,8 +283,6 @@ void AudioOutputI2S2_F32::update(void)
 		AudioStream_F32::release(block_f32_scaled);
 	}
 }
-
-
 void AudioOutputI2S2_F32::config_i2s(void) { config_i2s(false, AudioOutputI2S2_F32::sample_rate_Hz); }
 void AudioOutputI2S2_F32::config_i2s(bool transferUsing32bit) { config_i2s(transferUsing32bit, AudioOutputI2S2_F32::sample_rate_Hz); }
 void AudioOutputI2S2_F32::config_i2s(float fs_Hz) { config_i2s(false, fs_Hz); }
@@ -373,12 +293,9 @@ void AudioOutputI2S2_F32::config_i2s(bool transferUsing32bit, float fs_Hz)
 	CCM_CCGR5 |= CCM_CCGR5_SAI2(CCM_CCGR_ON);
 
 	// if either transmitter or receiver is enabled, do nothing
-	if (I2S2_TCSR & I2S_TCSR_TE)
+	if (I2S2_TCSR & I2S_TCSR_TE) return;
-		return;
+	if (I2S2_RCSR & I2S_RCSR_RE) return;
-	if (I2S2_RCSR & I2S_RCSR_RE)
-		return;
 	// PLL:
-	// int fs = AUDIO_SAMPLE_RATE_EXACT; //original from Teensy Audio Library
 	int fs = fs_Hz;
 
 	// PLL between 27*24 = 648MHz und 54*24=1296MHz
@@ -404,7 +321,6 @@ void AudioOutputI2S2_F32::config_i2s(bool transferUsing32bit, float fs_Hz)
 	int tsync = 1;
 
 	I2S2_TMR = 0;
-	// I2S1_TCSR = (1<<25); //Reset
 	I2S2_TCR1 = I2S_TCR1_RFW(1);
 	I2S2_TCR2 = I2S_TCR2_SYNC(tsync) | I2S_TCR2_BCP // sync=0; tx is async;
 				| (I2S_TCR2_BCD | I2S_TCR2_DIV((1)) | I2S_TCR2_MSEL(1));
@@ -413,7 +329,6 @@ void AudioOutputI2S2_F32::config_i2s(bool transferUsing32bit, float fs_Hz)
 	I2S2_TCR5 = I2S_TCR5_WNW((32 - 1)) | I2S_TCR5_W0W((32 - 1)) | I2S_TCR5_FBT((32 - 1));
 
 	I2S2_RMR = 0;
-	// I2S1_RCSR = (1<<25); //Reset
 	I2S2_RCR1 = I2S_RCR1_RFW(1);
 	I2S2_RCR2 = I2S_RCR2_SYNC(rsync) | I2S_RCR2_BCP // sync=0; rx is async;
 				| (I2S_RCR2_BCD | I2S_RCR2_DIV((1)) | I2S_RCR2_MSEL(1));
@@ -442,17 +357,17 @@ void AudioOutputI2S2slave_F32::begin(void)
 #if defined(__IMXRT1062__)
 	CORE_PIN7_CONFIG = 3; // 1:TX_DATA0
 	dma.TCD->SADDR = i2s2_tx_buffer;
-	dma.TCD->SOFF = 2;
+	dma.TCD->SOFF = 4;
-	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
+	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2);
-	dma.TCD->NBYTES_MLNO = 2;
+	dma.TCD->NBYTES_MLNO = 4;
 	dma.TCD->SLAST = -sizeof(i2s2_tx_buffer);
 	// dma.TCD->DADDR = (void *)((uint32_t)&I2S1_TDR1 + 2);
 	dma.TCD->DOFF = 0;
-	dma.TCD->CITER_ELINKNO = sizeof(i2s2_tx_buffer) / 2;
+	dma.TCD->CITER_ELINKNO = sizeof(i2s2_tx_buffer) / 4;
 	dma.TCD->DLASTSGA = 0;
-	dma.TCD->BITER_ELINKNO = sizeof(i2s2_tx_buffer) / 2;
+	dma.TCD->BITER_ELINKNO = sizeof(i2s2_tx_buffer) / 4;
 	// dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI2_TX);
-	dma.TCD->DADDR = (void *)((uint32_t)&I2S2_TDR0 + 2);
+	dma.TCD->DADDR = (void *)((uint32_t)&I2S2_TDR0);
 	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI2_TX);
 	dma.enable();

src/output_i2s2_F32.h

@@ -58,22 +58,11 @@ public:
 	virtual void update(void);
 	void begin(void);
 	void begin(bool);
-	void sub_begin_i32(void);
-	void sub_begin_i16(void);
 	friend class AudioInputI2S2_F32;
 
-	//friend class AudioInputI2S_F32;
-	#if defined(__IMXRT1062__)
 	friend class AudioOutputI2SQuad_F32;
 	friend class AudioInputI2SQuad_F32;
-
+	bool get_update_responsibility() { return update_responsibility;}
-	#endif
-
-	static void scale_f32_to_i16( float32_t *p_f32, float32_t *p_i16, int len) ;
-	static void scale_f32_to_i24( float32_t *p_f32, float32_t *p_i16, int len) ;
-	static void scale_f32_to_i32( float32_t *p_f32, float32_t *p_i32, int len) ;
-	
-	static float setI2SFreq_T3(const float);  // I2S clock for T3,x
 protected:
 	AudioOutputI2S2_F32(int dummy): AudioStream_F32(2, inputQueueArray) {} // to be used only inside AudioOutputI2Sslave !!
 	static void config_i2s(void);
@@ -85,8 +74,6 @@ protected:
 	static audio_block_f32_t *block_right_1st;
 	static bool update_responsibility;
 	static DMAChannel dma;
-	static void isr_16(void);
-	static void isr_32(void);
 	static void isr(void);
 private:
 	static audio_block_f32_t *block_left_2nd;
@@ -105,7 +92,6 @@ public:
 	AudioOutputI2S2slave_F32(void) : AudioOutputI2S2_F32(0) { begin(); } ;
 	void begin(void);
 	friend class AudioInputI2S2slave_F32;
-	friend void dma_ch0_isr(void);
 protected:
 	static void config_i2s(void);
 };

src/switch_selectorStereo_F32.h

@@ -0,0 +1,93 @@
+/**
+ * @file switch_selectorStereo_F32.h
+ * @author Piotr Zapart
+ * @brief Signal selector for routing mono to stereo 
+ * @version 0.1
+ * @date 2024-03-21
+ * 
+ * @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 _SWITCH_SELECTORSTEREO_F32_H_
+#define _SWITCH_SELECTORSTEREO_F32_H_
+#include <AudioStream_F32.h>
+#include <arm_math.h>
+
+class AudioSwitchSelectorStereo : public AudioStream_F32
+{
+public:
+	AudioSwitchSelectorStereo(void) : AudioStream_F32(2, inputQueueArray){};
+	typedef enum
+	{
+		SIGNAL_SELECT_LR,	 	// default stereo operation
+		SIGNAL_SELECT_L,		// left input as mono input 
+		SIGNAL_SELECT_R			// right input as mono input
+	}selector_mode_t;
+
+	selector_mode_t setMode(selector_mode_t m)
+	{
+		if (m <= 2)
+		{
+			__disable_irq();
+			mode = m;
+			__enable_irq();
+		}
+		return mode;
+	}
+	selector_mode_t getMode() {return mode;};
+	void update()
+	{
+		audio_block_f32_t *blockL, *blockR, *outL, *outR;
+		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;
+		}
+		switch(mode)
+		{
+			case SIGNAL_SELECT_LR:
+				outL = blockL;
+				outR = blockR;
+				break;
+			case SIGNAL_SELECT_L:
+				outL = blockL;
+				outR = blockL;
+				break;
+			case SIGNAL_SELECT_R:
+				outL = blockR;
+				outR = blockR;
+				break;
+			default:
+				outL = blockL;
+				outR = blockR;
+				break;			
+		}
+		AudioStream_F32::transmit(outL, 0);
+		AudioStream_F32::transmit(outR, 1);
+		AudioStream_F32::release(blockL);
+		AudioStream_F32::release(blockR);		
+	}
+
+private:
+	audio_block_f32_t *inputQueueArray[2];
+	selector_mode_t mode = SIGNAL_SELECT_LR;
+};
+
+#endif // _SWITCH_SELECTORSTEREO_F32_H_

src/wavetables.c

@@ -81,4 +81,3 @@ const float music_intevals[37] =
 	 3.174802f,  3.363586f,  3.563595f,  3.775497f,  4.000000f
 };
 
-