diff --git a/firmware/application/apps/replay_app.cpp b/firmware/application/apps/replay_app.cpp
index 9147c46f..4313b8e7 100644
--- a/firmware/application/apps/replay_app.cpp
+++ b/firmware/application/apps/replay_app.cpp
@@ -201,6 +201,16 @@ ReplayAppView::ReplayAppView(
 	tx_gain = 35;field_rfgain.set_value(tx_gain);  // Initial default  value (-12 dB's max ).
 	field_rfamp.set_value(rf_amp ? 14 : 0);  // Initial default value True. (TX RF amp on , +14dB's)
 
+	field_rfamp.on_change = [this](int32_t v) {	// allow initial value change just after opened file.	
+		rf_amp = (bool)v;
+	};
+	field_rfamp.set_value(rf_amp ? 14 : 0);
+
+    field_rfgain.on_change = [this](int32_t v) { // allow initial value change just after opened file.
+		tx_gain = v;
+	};  
+	field_rfgain.set_value(tx_gain);
+
 	baseband::run_image(portapack::spi_flash::image_tag_replay);
 
 	add_children({
diff --git a/firmware/application/apps/ui_mictx.cpp b/firmware/application/apps/ui_mictx.cpp
index bda0f278..68d98212 100644
--- a/firmware/application/apps/ui_mictx.cpp
+++ b/firmware/application/apps/ui_mictx.cpp
@@ -186,7 +186,7 @@ void MicTXView::rxaudio(bool is_on) {
 
 		baseband::run_image(portapack::spi_flash::image_tag_mic_tx);
 		audio::output::stop();
-		audio::input::start(ak4951_alc_GUI_selected);	// set up audio input =  mic config of any audio coded AK4951/WM8731, (in WM8731 parameter will be ignored)
+		audio::input::start();	// set up audio input =  mic config of any audio coded AK4951/WM8731, (in WM8731 parameter will be ignored)
 		portapack::pin_i2s0_rx_sda.mode(3);
 		configure_baseband();
 	}
@@ -212,7 +212,7 @@ MicTXView::MicTXView(
 	baseband::run_image(portapack::spi_flash::image_tag_mic_tx);
 	
 	
-	if ( audio_codec_wm8731.detected() ) {
+	if (true ) {       // Temporary , disabling ALC feature , (pending to solve -No Audio in Mic app ,in some H2/H2+ WM /QFP100 CPLS users-  if ( audio_codec_wm8731.detected() ) {
 	add_children({		
 		&labels_WM8731,		// we have audio codec WM8731, same MIC menu  as original.
 		&vumeter,
@@ -285,7 +285,7 @@ MicTXView::MicTXView(
 	
 	options_ak4951_alc_mode.on_change = [this](size_t, int8_t v) {
 	      ak4951_alc_GUI_selected  = v;
-		  audio::input::start(ak4951_alc_GUI_selected);
+		  audio::input::start();
 	};
 	// options_ak4951_alc_mode.set_selected_index(0);
 	 
@@ -341,23 +341,37 @@ MicTXView::MicTXView(
 				enable_dsb = false;
 				field_bw.set_value(transmitter_model.channel_bandwidth() / 1000);
 				//if (rx_enabled)
-				rxaudio(rx_enabled); //Update now if we have RX audio on
+				rxaudio(rx_enabled); 					//Update now if we have RX audio on
+				options_tone_key.hidden(0);				// we are in FM mode , we should have active the Key-tones & CTCSS option.
+				field_rxbw.hidden(0);					// we are in FM mode,  we need to allow the user set up of the RX NFM BW selection (8K5, 11K, 16K) 
 				break;
 			case 1:
 				enable_am = true;
-				rxaudio(rx_enabled); //Update now if we have RX audio on
+				rxaudio(rx_enabled); 					//Update now if we have RX audio on
+				options_tone_key.set_selected_index(0);	// we are NOT in FM mode ,  we reset the possible previous key-tones &CTCSS selection.
+				set_dirty();							// Refresh display	
+				options_tone_key.hidden(1);				// we hide that Key-tones & CTCSS input selecction, (no meaning in AM/DSB/SSB).
+				field_rxbw.hidden(1);					// we hide the NFM BW selection in other modes non_FM  (no meaning in AM/DSB/SSB)
+				check_rogerbeep.hidden(0);				// make visible again the "rogerbeep" selection.
 				break;
 			case 2:
 				enable_usb = true;
-				rxaudio(rx_enabled); //Update now if we have RX audio on
+				rxaudio(rx_enabled); 					//Update now if we have RX audio on
+				check_rogerbeep.set_value(false);		// reset the possible activation of roger beep, because it is not compatible with SSB , by now. 
+				check_rogerbeep.hidden(1);				// hide that roger beep selection. 
+				set_dirty();							// Refresh display
 				break;
 			case 3:
 				enable_lsb = true;
-				rxaudio(rx_enabled); //Update now if we have RX audio on
+				rxaudio(rx_enabled); 					//Update now if we have RX audio on
+				check_rogerbeep.set_value(false);		// reset the possible activation of roger beep, because it is not compatible with SSB , by now. 
+				check_rogerbeep.hidden(1);				// hide that roger beep selection. 
+				set_dirty();							// Refresh display
 				break;
 			case 4:
 				enable_dsb = true;
-				rxaudio(rx_enabled); //Update now if we have RX audio on
+				rxaudio(rx_enabled); 					//Update now if we have RX audio on
+				check_rogerbeep.hidden(0);				// make visible again the "rogerbeep" selection.
 				break;
 		}
 		//configure_baseband();
@@ -525,7 +539,7 @@ MicTXView::MicTXView(
 	set_tx(false);
 	
 	audio::set_rate(audio::Rate::Hz_24000);
-	audio::input::start(ak4951_alc_GUI_selected);		// originally , audio::input::start();  (we added parameter)
+	audio::input::start();		// originally , audio::input::start();  (we added parameter)
 }
 
 MicTXView::~MicTXView() {
diff --git a/firmware/application/audio.cpp b/firmware/application/audio.cpp
index ce32076c..4ba23acc 100644
--- a/firmware/application/audio.cpp
+++ b/firmware/application/audio.cpp
@@ -168,8 +168,8 @@ void speaker_mute() {
 
 namespace input {
 
-void start(int8_t alc_mode) {
-	audio_codec->microphone_enable(alc_mode);   // added user-GUI selection for AK4951, ALC mode parameter.
+void start() {
+	audio_codec->microphone_enable();
 	i2s::i2s0::rx_start();
 }
 
diff --git a/firmware/application/audio.hpp b/firmware/application/audio.hpp
index 56301ee8..b02e42aa 100644
--- a/firmware/application/audio.hpp
+++ b/firmware/application/audio.hpp
@@ -49,7 +49,7 @@ public:
 	virtual volume_range_t headphone_gain_range() const = 0;
 	virtual void set_headphone_volume(const volume_t volume) = 0;
 
-	virtual void microphone_enable(int8_t alc_mode) = 0;	// added user-GUI  AK4951 ,selected ALC mode.
+	virtual void microphone_enable() = 0;
 	virtual void microphone_disable() = 0;
 
 	virtual size_t reg_count() const = 0;
@@ -59,7 +59,7 @@ public:
 
 namespace output {
 
-void start();		// this  other start(),no changed. ,in namespace output , used to config audio playback mode, 
+void start();
 void stop();
 
 void mute();
@@ -72,7 +72,7 @@ void speaker_unmute();
 
 namespace input {
 
-void start(int8_t alc_mode);  // added parameter user-GUI select AK4951-ALC mode for config mic path,(recording mode in datasheet),
+void start();
 void stop();
 
 } /* namespace input */
diff --git a/firmware/baseband/dsp_modulate.cpp b/firmware/baseband/dsp_modulate.cpp
index 31dd3f85..04102800 100644
--- a/firmware/baseband/dsp_modulate.cpp
+++ b/firmware/baseband/dsp_modulate.cpp
@@ -21,6 +21,8 @@
 
 #include "dsp_modulate.hpp"
 #include "sine_table_int8.hpp"
+#include "portapack_shared_memory.hpp"
+#include "tonesets.hpp"
 
 namespace dsp {
 namespace modulate {
@@ -40,13 +42,42 @@ void Modulator::set_over(uint32_t new_over) {
     over = new_over;
 }
 
+void Modulator::set_gain_vumeter_beep(float new_audio_gain , bool new_play_beep ) {
+      audio_gain = new_audio_gain ;    
+	  play_beep = new_play_beep; 
+
+}
+
+int32_t Modulator::apply_beep(int32_t sample_in, bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message ) {
+
+	if (play_beep) {	   		// We need to add audio beep sample.
+			if (new_beep_timer) {
+				new_beep_timer--;
+			} else {
+				new_beep_timer = baseband_fs * 0.05;			// 50ms
+				
+				if (new_beep_index == BEEP_TONES_NB) {
+					configured_in = false;
+					shared_memory.application_queue.push(new_txprogress_message);
+				} else {
+					beep_gen.configure(beep_deltas[new_beep_index], 1.0); // config sequentially the audio beep tone.
+					new_beep_index++;
+				}
+			}
+		    sample_in = beep_gen.process(0);    // Get sample of the selected sequence of 6  beep tones , and overwrite audio sample. Mix 0%.
+		}
+	return sample_in;    // Return audio mic scaled with gain , 8 bit sample or audio beep sample.		
+}
+
+
 ///
 
 SSB::SSB() : hilbert() {
 	mode = Mode::LSB;
 }
 
-void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
+void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& configured_in,  uint32_t& new_beep_index, uint32_t& new_beep_timer,TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message,  uint32_t& new_power_acc_count, uint32_t& new_divider ) {
+	// No way to activate correctly  the roger beep in this option, Maybe not enough M4 CPU power , Let's  block roger beep in SSB  selection by now . 
 	int32_t		sample = 0;
 	int8_t		re = 0, im = 0;
 	
@@ -55,7 +86,9 @@ void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
 			float	i = 0.0, q = 0.0;
 
 		    sample = audio.p[counter / over] >> 2;
-			//switch (mode) {
+			sample *= audio_gain;      //  Apply GAIN  Scale factor to the audio TX modulation.
+	
+				//switch (mode) {
 				//case Mode::LSB:	
 			hilbert.execute(sample / 32768.0f, i, q);
 				//case Mode::USB:	hilbert.execute(sample / 32768.0f, q, i);
@@ -72,9 +105,23 @@ void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
 			//re = q;
 			//im = i;
 			//break;
+				
 		}
 		
 		buffer.p[counter] = { re, im };
+		
+		// Update vu-meter bar in the LCD screen.
+	    	power_acc += (sample < 0) ? -sample : sample;	// Power average for UI vu-meter
+	
+			if (new_power_acc_count) {
+				new_power_acc_count--;
+			} else {				// power_acc_count = 0
+				new_power_acc_count = new_divider;
+				new_level_message.value = power_acc / (new_divider *8);	// Why ?  . This division is to adj vu-meter sentitivity, to match saturation point to red-muter .
+				shared_memory.application_queue.push(new_level_message);
+				power_acc = 0;
+			}	
+	
 	}
 }
 
@@ -88,15 +135,39 @@ void FM::set_fm_delta(uint32_t new_delta) {
 	fm_delta = new_delta;
 }
 
-void FM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
+void FM::set_tone_gen_configure(const uint32_t set_delta, const float set_tone_mix_weight) {
+	 tone_gen.configure(set_delta, set_tone_mix_weight);
+}	
+
+void FM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message, uint32_t& new_power_acc_count, uint32_t& new_divider  ) {
 	int32_t		sample = 0;
 	int8_t		re, im;
 
 	for (size_t counter = 0; counter < buffer.count; counter++) {
-		if (counter % over == 0) {
-		    sample = audio.p[counter / over] >> 8;
-		    delta = sample * fm_delta;
-		}
+	 
+	    sample = audio.p[counter>>6] >> 8;			// 	sample = audio.p[counter / over] >> 8;   (not enough efficient running code, over = 1536000/240000= 64 )
+		sample *= audio_gain;      					// Apply GAIN  Scale factor to the audio TX modulation.
+
+		if (play_beep) {
+			sample = apply_beep(sample, configured_in, new_beep_index, new_beep_timer, new_txprogress_message ); 	// Apply beep -if selected - atom ,sample by sample.
+		} else {
+			// Update vu-meter bar in the LCD screen.
+	    	power_acc += (sample < 0) ? -sample : sample;	// Power average for UI vu-meter
+	
+			if (new_power_acc_count) {
+				new_power_acc_count--;
+			} else {				// power_acc_count = 0
+				new_power_acc_count = new_divider;
+				new_level_message.value = power_acc / (new_divider / 4);	// Why ?  . This division is to adj vu-meter sentitivity, to match saturation point to red-muter .
+				shared_memory.application_queue.push(new_level_message);
+				power_acc = 0;
+			}	
+			// TODO: pending to optimize CPU running code. 
+			// So far , we can not handle all 3 issues at the same time (vu-meter , CTCSS, beep).
+			sample = tone_gen.process(sample);			// Add selected Key_Tone or CTCSS subtone , atom function() , sample by sample.
+		}	
+  
+	    delta = sample * fm_delta;						// Modulate FM
 
 		phase += delta;
 		sphase = phase >> 24;
@@ -112,16 +183,33 @@ AM::AM() {
 	mode = Mode::AM;
 }
 
-void AM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
+void AM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message, uint32_t& new_power_acc_count, uint32_t& new_divider ) {
 	int32_t         sample = 0;
 	int8_t          re = 0, im = 0;
-	float		q = 0.0;
+	float			q = 0.0;
 	
 	for (size_t counter = 0; counter < buffer.count; counter++) {
 		if (counter % 128 == 0) {
 			sample = audio.p[counter / over] >> 2;
+			sample *= audio_gain;      					 // Apply GAIN  Scale factor to the audio TX modulation.
 		}
 
+		if (play_beep) {
+			sample = apply_beep(sample, configured_in, new_beep_index, new_beep_timer, new_txprogress_message )<<5; 	// Apply beep -if selected - atom sample by sample.
+		} else {
+			// Update vu-meter bar in the LCD screen.
+	    	power_acc += (sample < 0) ? -sample : sample;	// Power average for UI vu-meter
+	
+			if (new_power_acc_count) {
+				new_power_acc_count--;
+			} else {				// power_acc_count = 0
+				new_power_acc_count = new_divider;
+				new_level_message.value = power_acc / (new_divider *8);	// Why ?orig / (new_divider / 4);	// Why ?
+				shared_memory.application_queue.push(new_level_message);
+				power_acc = 0;
+			}	
+		}	
+
 		q = sample / 32768.0f;
 		q *= 256.0f;										 // Original 64.0f,now x4 (+12 dB's BB_modulation in AM & DSB)	
 		switch (mode) {
diff --git a/firmware/baseband/dsp_modulate.hpp b/firmware/baseband/dsp_modulate.hpp
index fdc6e95f..de964a68 100644
--- a/firmware/baseband/dsp_modulate.hpp
+++ b/firmware/baseband/dsp_modulate.hpp
@@ -24,6 +24,8 @@
 
 #include "dsp_types.hpp"
 #include "dsp_hilbert.hpp"
+#include "tone_gen.hpp"
+#include "baseband_processor.hpp"
 
 namespace dsp {
 namespace modulate {
@@ -41,13 +43,28 @@ enum class Mode {
 
 class Modulator {
 public:
-	virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) = 0;
+	virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer,bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message, uint32_t& new_power_acc_count, uint32_t& new_divider   ) = 0;
 	virtual ~Modulator();
 
 	Mode get_mode();
 	void set_mode(Mode new_mode);
 
     void set_over(uint32_t new_over);
+	void set_gain_vumeter_beep(float new_audio_gain , bool new_play_beep );
+	int32_t apply_beep(int32_t sample_in, bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message );
+	float audio_gain { };
+	bool play_beep { false };
+	uint32_t power_acc_count { 0 };		// this var it is initialized from Proc_mictx.cpp
+	uint32_t divider { };				// this var it is initialized from Proc_mictx.cpp
+    uint64_t power_acc { 0 };	// it is aux Accumulated sum (Absolute sample signal) , initialitzed to zero. 
+	AudioLevelReportMessage level_message { };	
+
+private: 
+	static constexpr size_t baseband_fs = 1536000U;
+	TXProgressMessage txprogress_message { };
+   	ToneGen  beep_gen { };
+	uint32_t beep_index { }, beep_timer { };
+
 
 protected:
     uint32_t    over = 1;
@@ -60,7 +77,7 @@ class SSB : public Modulator {
 public:
 	SSB();
 
-	virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer);
+	virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message, uint32_t& new_power_acc_count, uint32_t& new_divider  );
 
 private:
 	dsp::HilbertTransform	hilbert;
@@ -72,8 +89,9 @@ class FM : public Modulator {
 public:
 	FM();
 
-	virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer);
+	virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& configured_in,  uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message, uint32_t& new_power_acc_count, uint32_t& new_divider  ) ;
 	void set_fm_delta(uint32_t new_delta);
+	void set_tone_gen_configure(const uint32_t delta, const float tone_mix_weight); 
 
 ///
 
@@ -81,13 +99,16 @@ private:
 	uint32_t	fm_delta { 0 };
 	uint32_t	phase { 0 }, sphase { 0 };
 	int32_t		sample { 0 }, delta { };
+	ToneGen 	tone_gen { };
+	
+
 };
 
 class AM : public Modulator {
 public:
         AM();
 
-        virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer);
+        virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& configured_in, uint32_t& new_beep_index, uint32_t& new_beep_timer, TXProgressMessage& new_txprogress_message, AudioLevelReportMessage& new_level_message, uint32_t& new_power_acc_count, uint32_t& new_divider  );
 };
 
 } /* namespace modulate */
diff --git a/firmware/baseband/proc_mictx.cpp b/firmware/baseband/proc_mictx.cpp
index d5c0040c..6c9c9b97 100644
--- a/firmware/baseband/proc_mictx.cpp
+++ b/firmware/baseband/proc_mictx.cpp
@@ -35,8 +35,10 @@ void MicTXProcessor::execute(const buffer_c8_t& buffer){
 	if (!configured) return;
 	
 	audio_input.read_audio_buffer(audio_buffer);
-	modulator->execute(audio_buffer, buffer);
-
+	modulator->set_gain_vumeter_beep(audio_gain, play_beep ) ;	
+	modulator->execute(audio_buffer, buffer, configured, beep_index, beep_timer, txprogress_message, level_message, power_acc_count, divider );	// Now "Key Tones & CTCSS" baseband additon inside FM mod. dsp_modulate.cpp"
+    
+   /* Original fw 1.3.1  good reference, beep and vu-meter
 	for (size_t i = 0; i < buffer.count; i++) {
 		
 		if (!play_beep) {
@@ -67,13 +69,13 @@ void MicTXProcessor::execute(const buffer_c8_t& buffer){
 					beep_index++;
 				}
 			}
-			
-			sample = beep_gen.process(0);
-		}
-		
-		/*
+			sample = beep_gen.process(0);    // TODO : Pending how to move inside modulate.cpp
+		} 
+	 */ 	
+				
+       /* Original fw 1.3.1  good reference FM moulation version, including "key tones CTCSS"  fw 1.3.1 
 		sample = tone_gen.process(sample);
-		
+				
 		// FM
 		if (configured) {
 			delta = sample * fm_delta;
@@ -89,8 +91,8 @@ void MicTXProcessor::execute(const buffer_c8_t& buffer){
 		}
 		
 		buffer.p[i] = { re, im };
-		*/
-	}
+		
+	}  */
 }
 
 void MicTXProcessor::on_message(const Message* const msg) {
@@ -100,12 +102,16 @@ void MicTXProcessor::on_message(const Message* const msg) {
 	switch(msg->id) {
 		case Message::ID::AudioTXConfig:
 			if (fm_enabled) {
-				dsp::modulate::FM *fm = new dsp::modulate::FM();
-				
-				fm->set_fm_delta(config_message.deviation_hz * (0xFFFFFFUL / baseband_fs));
+				 dsp::modulate::FM *fm = new dsp::modulate::FM();
+
+				// Config fm_delta private var inside DSP modulate.cpp
+				 fm->set_fm_delta(config_message.deviation_hz * (0xFFFFFFUL / baseband_fs));
+
+				 // Config properly the private tone_gen function parameters inside DSP modulate.cpp  	
+				 fm->set_tone_gen_configure(config_message.tone_key_delta, config_message.tone_key_mix_weight);		
 				modulator = fm;
 			}
-			
+
 			if (usb_enabled) {
 				modulator = new dsp::modulate::SSB();
 				modulator->set_mode(dsp::modulate::Mode::USB);
@@ -124,7 +130,7 @@ void MicTXProcessor::on_message(const Message* const msg) {
 				modulator->set_mode(dsp::modulate::Mode::DSB);
 			}
 
-            modulator->set_over(baseband_fs / 24000);
+			modulator->set_over(baseband_fs / 24000); // Keep no change.
 			
 			am_enabled = config_message.am_enabled;
 			usb_enabled = config_message.usb_enabled;
@@ -137,8 +143,9 @@ void MicTXProcessor::on_message(const Message* const msg) {
 			audio_gain = config_message.audio_gain;
 			divider = config_message.divider;
 			power_acc_count = 0;
-			
-			tone_gen.configure(config_message.tone_key_delta, config_message.tone_key_mix_weight);
+
+			// now this config  moved, in the case Message::ID::AudioTXConfig , only FM case.
+			// tone_gen.configure(config_message.tone_key_delta, config_message.tone_key_mix_weight);	
 			
 			txprogress_message.done = true;
 
diff --git a/firmware/baseband/proc_mictx.hpp b/firmware/baseband/proc_mictx.hpp
index b26cae38..175e53c7 100644
--- a/firmware/baseband/proc_mictx.hpp
+++ b/firmware/baseband/proc_mictx.hpp
@@ -49,8 +49,8 @@ private:
 	};
 	
 	AudioInput audio_input { };
-	ToneGen tone_gen { };
-	ToneGen beep_gen { };
+	// ToneGen tone_gen { };   moved to  dsp_modulate.cpp
+	// ToneGen beep_gen { };   moved to  dsp_modulate.cpp	
 	dsp::modulate::Modulator *modulator = NULL ;
 
 	bool am_enabled { false };
diff --git a/firmware/baseband/tone_gen.cpp b/firmware/baseband/tone_gen.cpp
index dd74579d..d015f94f 100644
--- a/firmware/baseband/tone_gen.cpp
+++ b/firmware/baseband/tone_gen.cpp
@@ -24,14 +24,20 @@
 #include "sine_table_int8.hpp"
 
 
+/*
 int32_t ToneGen::tone_sine() {
+	//  TODO :  Added for Sonde App. We keep it by now , but it needs to be reviewed in Sonde
+	//  Hoepfully we can manage without it , same as previous fw  1.3.1  
 	int32_t tone_sample = sine_table_i8[tone_phase_] * 0x1000000;
 	tone_phase_ += delta_;
 
 	return tone_sample;
 }
+*/ 
+
 
 int32_t ToneGen::tone_square() {
+	//  TODO :  Added for Sonde App. We keep it by now , but it needs to be reviewed in Sonde
 	int32_t tone_sample = 0;
 
 	if(tone_phase_  < (UINT32_MAX / 2)) {
@@ -46,33 +52,66 @@ int32_t ToneGen::tone_square() {
 	return tone_sample;
 }
 
+/*   
 void ToneGen::configure(const uint32_t delta, const float tone_mix_weight) {
+	//  Confirmed !  It is not working well in the fw 1.4.4  Mic App , CTCSS generation, (but added for Sonde App)
+	//  I Think it should be deleted or modified but not use it as it is in Mic App . 	
+
 	delta_ = (uint8_t) ((delta & 0xFF000000U) >> 24);
+	delta_ = delta;
 	tone_mix_weight_ = tone_mix_weight;
 	input_mix_weight_ = 1.0 - tone_mix_weight;
-
 	current_tone_type_ = sine;
 }
+*/ 
+
+
 
 void ToneGen::configure(const uint32_t freq, const float tone_mix_weight, const tone_type tone_type, const uint32_t sample_rate) {
+	//  TODO :  Added for Sonde App. We keep it by now to avoid compile errors, but it needs to be reviewed in Sonde
 	delta_ = (uint8_t) ((freq * sizeof(sine_table_i8)) / sample_rate);
 	tone_mix_weight_ = tone_mix_weight;
 	input_mix_weight_ = 1.0 - tone_mix_weight;
 	current_tone_type_ = tone_type;
+}  
+
+
+
+
+//  ----Original available core SW code from fw 1.3.1 ,  Working also well in Mic App CTCSS Gen from fw 1.4.0 onwards
+
+// Original direct-look-up synthesis algorithm with Fractional delta phase. It is OK
+// Delta  and Accumulator fase are stored in 32 bits (4 bytes), 1st top byte used as Modulo-256 Sine look-up table [index]
+// the lower 3 bytes (24 bits)  are used as a Fractional Detla and Accumulator phase, to have very finer Fstep control.
+
+void ToneGen::configure(const uint32_t delta, const float tone_mix_weight) {
+	delta_ = delta;
+	tone_mix_weight_ = tone_mix_weight;
+	input_mix_weight_ = 1.0 - tone_mix_weight;
 }
 
 int32_t ToneGen::process(const int32_t sample_in) {
 	if (!delta_)
 		return sample_in;
 	
-	int32_t tone_sample = 0;
-	
-	if(current_tone_type_ == sine) {
-		tone_sample = tone_sine();
-	}
-	else if(current_tone_type_ == square) {
-		tone_sample = tone_square();
-	}
+	int32_t tone_sample = sine_table_i8[(tone_phase_ & 0xFF000000U) >> 24];
+	tone_phase_ += delta_;
+	
+	return (sample_in * input_mix_weight_) + (tone_sample * tone_mix_weight_);
+}
+// -------------------------------------------------------------
+
+
+
+int32_t ToneGen::process_square(const int32_t sample_in) {
+	//  TODO :  Added for Sonde App. We keep it by now , but it needs to be reviewed in Sonde
+	if (!delta_)
+		return sample_in;
+	
+	int32_t tone_sample = 0;
+	
+		tone_sample = tone_square();
+	
 	
 	return (sample_in * input_mix_weight_) + (tone_sample * tone_mix_weight_);
 }
diff --git a/firmware/baseband/tone_gen.hpp b/firmware/baseband/tone_gen.hpp
index c1eee222..eca69e2f 100644
--- a/firmware/baseband/tone_gen.hpp
+++ b/firmware/baseband/tone_gen.hpp
@@ -28,7 +28,7 @@
 
 class ToneGen {
 public:
-	enum tone_type { sine, square };
+	enum tone_type { sine, square };    // TODO:  Added for Radio Sonde.cpp PR 376, 381 , we need to check if keep or not.
 
 	/*ToneGen(const size_t sample_rate
 	) : sample_rate_ { sample_rate }
@@ -38,6 +38,7 @@ public:
 	void configure(const uint32_t freq, const float tone_mix_weight, const tone_type tone_type, const uint32_t sample_rate);
 
 	int32_t process(const int32_t sample_in);
+	int32_t process_square(const int32_t sample_in);
 
 private:
 	tone_type current_tone_type_ { sine };
@@ -45,19 +46,23 @@ private:
 	float input_mix_weight_ { 1 };
 	float tone_mix_weight_ { 0 };
 
-	uint8_t delta_ { 0 };
-	uint8_t tone_phase_ { 0 };
+	uint32_t delta_ { 0 };
+	uint32_t tone_phase_ { 0 };
+
+//	uint8_t delta_ { 0 };		// TODO:  Added for Radio Sonde.cpp PR 376, 381 , we need to check if keep or not.
+//	uint8_t tone_phase_ { 0 };  // TODO:  Added for Radio Sonde.cpp PR 376, 381 , we need to check if keep or not.
 
 	/**
 	 * Generator function which selects every other sample from the reference sine waveform to the output sample:
 	 */
-	int32_t tone_sine();
+	int32_t tone_sine();// TODO:  Added for Radio Sonde.cpp PR 376, 381 , we need to check if keep or not.
 
 
 	/**
 	 * Generator function for square waves:
 	 */	
-	int32_t tone_square();
+	int32_t tone_square();		// TODO:  Added for Radio Sonde.cpp PR 376, 381 , we need to check if keep or not.
+
 };
 
 #endif /* __TONE_GEN_H__ */
diff --git a/firmware/common/ak4951.cpp b/firmware/common/ak4951.cpp
index 1f88a3d6..74301ecb 100644
--- a/firmware/common/ak4951.cpp
+++ b/firmware/common/ak4951.cpp
@@ -216,347 +216,90 @@ void AK4951::speaker_disable() {
 	set_speaker_power(false);
 }
 
-void AK4951::microphone_enable(int8_t alc_mode) {
-// alc_mode =0 = (OFF =same as original code = NOT using AK4951 Programmable digital filter block), 
-// alc_mode >1 (with  DIGITAL FILTER BLOCK , example :  1:(+12dB) , 2:(+9dB)", 3:(+6dB), ...) 
-
-// map.r.digital_mic.DMIC = 0;		// originally commented code 
-// update(Register::DigitalMic);	// originally commented code 
-
-uint_fast8_t mgain =0b0111;   			// Pre-amp mic (Original code,  =0b0111 (+21dB's=7x3dBs),(Max is NOT 0b1111!, it is 0b1010=+30dBs=10x3dBs) 
-
-map.r.signal_select_2.INL = 0b01;		// Lch input signal = LIN2 , our ext. MONO MIC is connected here LIN2 in Portapack.
-map.r.signal_select_2.INR = 0b01;		// Rch input signal = RIN2 , Not used ,not connected ,but no problem. 
-map.r.signal_select_2.MICL = 0;			// MPWR = 2.4V  (it has two possible settings , 2.4V or 2.0V) , (majority smarthphones around 2V , range 1V-5V)
-update(Register::SignalSelect2);
-
-// ------Common code part, =  original setting  conditions, it is fine for all user-GUI alc_modes: OFF , and ALC modes .*/
-map.r.digital_filter_select_1.HPFAD = 1;	// HPF1 ON (after ADC);page 40 datasheet, HPFAD bit controls the ON/OFF of the HPF1 (HPF ON is recommended).
-map.r.digital_filter_select_1.HPFC = 0b11;	// HPF Cut off frequency of high pass filter from 236.8 Hz @fs=48k  ("00":3.7Hz, "01":14,8Hz, "10":118,4Hz) 
-update(Register::DigitalFilterSelect1);
-
-// map.r.r_ch_mic_gain_setting.MGR = 0x80;	// Microphone sensitivity correction = 0dB.,  (not used by now , original code cond.)
-// update(Register::RchMicGainSetting);		// (those two lines , not activated, same as original)	
-
-// pre-load 4 byes LPF coefficicients (.lpf_coefficient_0,1,2,3), FSA 14..0, FSB 14..0 , (fcut initial 6kHz, fs 48Khz). 	
-// it will be default pre-loading coeff. for al  ALC modes,  LPF bit is activated down, for all ALC digital modes.
-map.r.lpf_coefficient_0.l = 0x5F;		// Pre-loading here LPF 6kHz, 1st Order from digital Block , Fc=6000 Hz,  fs = 48khz
-map.r.lpf_coefficient_1.h = 0x09;		// LPF bit is activated down, for all ALC digital modes.
-map.r.lpf_coefficient_2.l = 0xBF;		// Writting reg to AK4951,  with "update", following instructions.
-map.r.lpf_coefficient_3.h = 0x32;
-
-update(Register::LPFCoefficient0);	// Writing  pre-loaded  4 bytes LPF  CoefFiecients 14 bits (FSA13..0,  FSB13..0
-update(Register::LPFCoefficient1);	// In this case , LPF 6KHz , when we activate the LPF block.
-update(Register::LPFCoefficient2);
-update(Register::LPFCoefficient3);
-
-// Reset , setting OFF all 5 x Digital Equalizer filters 
-map.r.digital_filter_select_3.EQ1 = 0;	  // EQ1 Coeffic Setting , (0: Disable-default, audio data passes  EQ1 block by 0dB gain). When EQ1="1”, the settings of E1A15-0, E1B15-0 and E1C15-0 bits are enabled
-map.r.digital_filter_select_3.EQ2 = 0;	  // EQ2 Coeffic Setting , (0: Disable-default, audio data passes  EQ2 block by 0dB gain). When EQ2="1”, the settings of E2A15-0, E2B15-0 and E2C15-0 bits are enabled
-map.r.digital_filter_select_3.EQ3 = 0;    // EQ3 Coeffic Setting , (0: Disable-default, audio data passes  EQ3 block by 0dB gain). When EQ3="1”, the settings of E3A15-0, E3B15-0 and E3C15-0 bits are enabled
-map.r.digital_filter_select_3.EQ4 = 0;    // EQ4 Coeffic Setting , (0: Disable-default, audio data passes  EQ4 block by 0dB gain). When EQ4="1”, the settings of E4A15-0, E4B15-0 and E4C15-0 bits are enabled
-map.r.digital_filter_select_3.EQ5 = 0;    // EQ5 Coeffic Setting , (0: Disable-default, audio data passes  EQ5 block by 0dB gain). When EQ5="1”, the settings of E5A15-0, E5B15-0 and E5C15-0 bits are enabled
-update(Register::DigitalFilterSelect3);		// A,B,C EQ1 Coefficients are already pre-loaded in ak4951.hpp
-
-
-  if (alc_mode==0) { 		// Programmable Digital Filter OFF,  same as original condition., no Digital ALC, nor Wind Noise Filter, LPF , EQ 
-
-		map.r.digital_filter_select_2.LPF = 0;	  // LPF-Block, Coeffic Setting Enable (OFF-Default), When LPF bit is “0”, audio data passes the LPF block by 0dB gain. 
-		update(Register::DigitalFilterSelect2);
-
-	  	// Pre-loading AUDIO PATH with all DIGITAL BLOCK by pased, see, audio path block diagramm AK4951 datasheet + Table  Playback mode  -Recording mode. 
-		// Digital filter block PATH is BY PASSED (we can swith off DIG. BLOCK power , PMPFIL=0) .The Path in Recording Mode 2 & Playback Mode 2  (NO DIG FILTER BLOCK AT ALL, not for MIC recording, nor for Playback)
-		map.r.digital_filter_mode.ADCPF = 1;	// ADCPF bit swith ("0" Mic  after ADC Output connected (recording mode) to the DIGITAL FILTER  BLOCK. ("1" Playback mode)  
-		map.r.digital_filter_mode.PFSDO = 0;	// ADC bit switch ("0" : 1st order HPF) connectedto the  Output. By bass DIGITAL block .
-    	map.r.digital_filter_mode.PFDAC = 0b00;	// (Input selector for DAC (not used in MIC), SDTI= Audio Serial Data Input Pin)
-		update(Register::DigitalFilterMode);		// Writing the Audio Path :  NO DIGITAL BLOCK or DIG BLOCK FOR  MIC ,   Audio mode path : Playback mode /-Recording mode.
-
-		map.r.power_management_1.PMADL  = 1;	// ADC Lch = Lch input signal. Mic Amp Lch and ADC Lch Power Management
-		map.r.power_management_1.PMADR  = 1;	// ADC Rch = Rch input signal. Mic Amp Rch and ADC Rch Power Management
-		map.r.power_management_1.PMPFIL = 0;	// Pre-loading , Programmable Dig. filter OFF ,filter unused, routed around.(original value = 0 ) 
-		update(Register::PowerManagement1);		// Activating the Power management of the used blocks . (Mic ADC always + Dig Block filter , when used )
-
-		// 1059/fs, 22ms @ 48kHz
-		chThdSleepMilliseconds(22);
-
-  } else {    // ( alc_mode !=0) 
-
-	switch(alc_mode) { 	// Pre-loading register values depending on user-GUI selection (they will be sended below, with "update(Register_name::xxx )".
-	 
-		case 1: 	// ALC-> on, (+12dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0xC0;    	// REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, C0H=+12dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xC0; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xC0; 	// Right Input Dig Vol Setting, same comment as above , The value of IVOL should be <= than REF’s
-
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-		case 2: 	// ALC-> on, (+09dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0xB8;    	// REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, B8H= +9dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xB8; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xB8; 	// Right Input Dig Vol Setting, same comment as above , The value of IVOL should be <= than REF’s
-
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-	    case 3: 	// ALC-> on, (+06dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0xB0;    	// 0xB8 , REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, B0H= +6dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xB0; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xB0; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
-		
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-	    case 4: 	// ALC-> on, (+03dB's) Auto Vol max + Wind Noise cancel + Pre-amp Mic (+21dB=original)
-					// + EQ boosting ~<2kHz (f0:1,1k, fb:1,7K, k=1,8)   && + LPF 3,5k 
-			map.r.alc_mode_control_2.REF = 0xA8;    	// 0xA8 , REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A8H= +3dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xA8; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xA8; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
-
-			//The EQn (n=1, 2, 3, 4 or 5) coefficient must be set when EQn bit = “0” or PMPFIL bit = “0”.
-			map.r.digital_filter_select_3.EQ1 = 1;	    // EQ1 Coeffic Setting , (0: Disable-default, audio data passes  EQ1 block by 0dB gain). When EQ1="1”, the settings of E1A15-0, E1B15-0 and E1C15-0 bits are enabled
-			update(Register::DigitalFilterSelect3);		// A,B,C EQ1 Coefficients are already pre-loaded in ak4951.hpp
-
-			map.r.lpf_coefficient_0.l = 0x0D;		// Pre-loading here LPF 3,5k  ,  1st Order from digital Block , Fc=3.500 Hz,  fs = 48khz
-		   	map.r.lpf_coefficient_1.h = 0x06;		// LPF bit is activated down, for all ALC digital modes.
-       		map.r.lpf_coefficient_2.l = 0x1A;		// Writting reg to AK4951 , down with update....
-	   		map.r.lpf_coefficient_3.h = 0x2C;
-   			// LPF bit is activated down, for all ALC digital modes.			
- 			break;
-
-	  	case 5: 	// ALC-> on, (+03dB's) Auto Vol max + Wind Noise cancel + Pre-amp Mic (+21dB=original)
-		  			// + EQ boosting ~<3kHz (f0~1k4,fb~2,4k,k=1,8)   && LPF 4kHz 
-			map.r.alc_mode_control_2.REF = 0xA8;    	// 0xA0 , REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A8H= +3dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xA8; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xA8; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
-		
-			map.r.digital_filter_select_3.EQ2 = 1;	  // EQ2 Coeffic Setting , (0: Disable-default, audio data passes  EQ2 block by 0dB gain). When EQ2="1”, the settings of E2A15-0, E2B15-0 and E2C15-0 bits are enabled
-			update(Register::DigitalFilterSelect3);
-
-			map.r.lpf_coefficient_0.l = 0xC3;		// Pre-loading here LPF 4k  ,  1st Order from digital Block , Fc=4000 Hz,  fs = 48khz
-	   		map.r.lpf_coefficient_1.h = 0x06;		// LPF bit is activated down, for all ALC digital modes.
-       		map.r.lpf_coefficient_2.l = 0x86;		// Writting reg to AK4951 , down with update....
-	   		map.r.lpf_coefficient_3.h = 0x2D;
-   			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-		 case 6: 	// ALC-> on, (+03dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0xA8;    	// REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A0H= 0dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xA8; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xA8; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
-
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-		 case 7: 	// ALC-> on, (+00dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0xA0;    	// REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A0H= 0dBs) 
-			map.r.l_ch_input_volume_control.IV = 0xA0; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0xA0; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
-
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-		case 8:		// ALC-> on, (-03dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0x98;    	//REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 98H=-03dBs) 
-			map.r.l_ch_input_volume_control.IV = 0x98; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0x98; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
+void AK4951::microphone_enable() {
+// map.r.digital_mic.DMIC = 0;
+// update(Register::DigitalMic);
 	
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
+	const uint_fast8_t mgain = 0b0111;
+	map.r.signal_select_1.MGAIN20 = mgain & 7;
+	map.r.signal_select_1.PMMP = 1;
+	map.r.signal_select_1.MPSEL = 1;	// MPWR2 pin
+	map.r.signal_select_1.MGAIN3 = (mgain >> 3) & 1;
+	update(Register::SignalSelect1);
 
-		case 9: 	// ALC-> on, (-06dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
-			map.r.alc_mode_control_2.REF = 0x90;    	// REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 90H=-06dBs) 
-			map.r.l_ch_input_volume_control.IV = 0x90; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0x90; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
+	map.r.signal_select_2.INL = 0b01;	// Lch input signal = LIN2
+	map.r.signal_select_2.INR = 0b01;	// Rch input signal = RIN2
+	map.r.signal_select_2.MICL = 0;		// MPWR = 2.4V
+	update(Register::SignalSelect2);
 
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-		case 10: 	// ALC-> on, (-09dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz - Pre-amp MIC -3dB (18dB's) 
-			// Reduce also Pre-amp Mic -3dB's (+18dB's)		
-			mgain = 0b0110;		// Pre-amp mic Mic Gain Pre-amp (+18dB), Original=0b0111 (+21dB's =7x3dBs),
-	
-			map.r.alc_mode_control_2.REF = 0x88;    	// REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 88H=-09dBs) 
-			map.r.l_ch_input_volume_control.IV = 0x88; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0x88; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
- 	
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-
-		case 11: // ALC-> on, (-12dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz - Pre-amp MIC -6dB (15dB's) 	
-			// Reduce also Pre-amp Mic -6dB's (+15dB's)  	
-			mgain = 0b0101;		// Pre-amp mic Mic Gain Pre-amp (+15dB), (Original=0b0111 (+21dB's= 7x3dBs),
-
-			map.r.alc_mode_control_2.REF = 0x80;    	// REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 80H=-12dBs) 
-			map.r.l_ch_input_volume_control.IV = 0x80; 	// Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-			map.r.r_ch_input_volume_control.IV = 0x80; 	// Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
-
-			// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
-			// LPF bit is activated down, for all ALC digital modes.
-			break;
-	}
-
-	//-------------------------------DIGITAL  ALC  (Automatic Level Control ) --- --------
-	map.r.alc_mode_control_1.ALC = 0;  // LMTH2-0, WTM1-0, RGAIN2-0, REF7-0, RFST1-0, EQFC1-0, FRATT, FRN and ALCEQN bits (needs to be set up with  ALC disable = 0) 
-	update(Register::ALCModeControl1);
-
-	map.r.timer_select.FRN = 0;      	// (FRN= 0  Fast Recovery mode , enable ) 
-	map.r.timer_select.FRATT = 0;    	// Fast Recovery Ref. Volume Atten. Amount -0,00106dB's, timing 4/fs (default)
-	map.r.timer_select.ADRST = 0b00; 	// initial offset ADC cycles , 22ms @fs=48Khz.
+// map.r.r_ch_mic_gain_setting.MGR = 0x80;	// Microphone sensitivity correction = 0dB.
+// update(Register::RchMicGainSetting);
+/*
+	map.r.timer_select.FRN = ?;
+	map.r.timer_select.FRATT = ?;
+	map.r.timer_select.ADRST = 0b??;
 	update(Register::TimerSelect);
 
-	map.r.alc_timer_select.RFST = 0b00;   // RFST1-0: ALC Fast Recovery Speed Default: “00” (0.0032dB)
-	map.r.alc_timer_select.WTM = 0b00;    // ALC Recovery Operation Waiting Period 128/fs = 2,7 mseg (min=default)
-	map.r.alc_timer_select.EQFC = 0b10;   // Selecting default, fs 48Khz , ALCEQ: First order zero pole high pass filter fc2=100Hz, fc1=150Hz
-	map.r.alc_timer_select.IVTM = 0;      // IVTM bit set the vol transition time ,236/fs = 4,9msecs (min) (default was 19,7msegs.)
+	map.r.alc_timer_select. = ?;
 	update(Register::ALCTimerSelect);
-
-	map.r.alc_mode_control_1.LMTH10 = 0b11;  // ALC Limiter Detec Level/ Recovery Counter Reset; lower 2 bits (Ob111=-8,4dbs), (default 0b000=-2,5dBs)
-	map.r.alc_mode_control_1.RGAIN  = 0b000; // ALC Recovery Gain Step, max step , max speed. Default: “000” (0.00424dB)
-	map.r.alc_mode_control_1.ALC = 1;		 // ALC Enable .  (we are now, NOT in MANUAL volume mode, only becomes manual when (ALC=“0” while ADCPF=“1”. )
-	map.r.alc_mode_control_1.LMTH2 = 1;      // ALC Limiter Detection Level/ Recovery Counter Reset Level,Upper bit,default 0b000      
-	map.r.alc_mode_control_1.ALCEQN = 1;     // ALC EQ Off =1 not used by now,   0: ALC EQ On (default) 
+	map.r.alc_mode_control_1. = ?;
+	map.r.alc_mode_control_1.ALC = 1;
 	update(Register::ALCModeControl1);
 
-	//   map.r.alc_mode_control_2.REF = 0x??;   	// Pre-loaded in top part. Maximum gain at ALC recovery operation,.(FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs) 
+	map.r.alc_mode_control_2.REF = ?;
 	update(Register::ALCModeControl2);
-
-	//   map.r.l_ch_input_volume_control.IV = 0x??; // Pre-loaded in top part. Left, Input Digital Volume Setting,  (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
-    update(Register::LchInputVolumeControl);
-	
-	//   map.r.r_ch_input_volume_control.IV = 0x??; // Pre-loaded in top part. Right,Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs) 
-    update(Register::RchInputVolumeControl);
-	
-	
-	//---------------Switch ON, Digital Automatic Wind Noise Filter reduction  -------------------     
-	// Difficult to realise that Dynamic HPF Wind noise filter benefit, maybe because we have another fixed HPF 236.8 Hz .
-	// Anyway ,  we propose to  activate it , with default  setting conditions.
-	map.r.power_management_1.PMPFIL = 0;	// (*1) To programm SENC, STG , we need PMPFIL = 0 . (but this disconnect Digital block power supply.
-	update(Register::PowerManagement1);		//  Updated PMPFIL to 0 . (*1)
-
-	map.r.auto_hpf_control.STG = 0b00;		//  (00=LOW ATTENUATION Level), lets put 11 (HIGH ATTENUATION Level) (default 00)
-	map.r.auto_hpf_control.SENC = 0b011;	//  (000=LOW sensitivity detection)… 111((MAX sensitivity detection) (default 011)
-	map.r.auto_hpf_control.AHPF = 1;    	//  Autom. Wind noise filter ON  (AHPF bit=“1”).It atten. wind noise when detecting ,and adjusts the atten. level dynamically.
-	update(Register::AutoHPFControl);       
-
-	//  We are in Digital Block ON , (Wind Noise Filter+ALC+LPF+EQ),==>  needs at the end , PMPFIL=1 , Program. Dig.filter ON   
-	//  map.r.power_management_1.PMPFIL = 1;	// that instruction is at the end , we can skp pre-loading Programmable Dig. filter ON (*1)
-	//---------------------------------------------------------------------
- 	    
-	// Writing AUDIO PATH diagramm, Changing  Audio mode path : Playback mode1 /-Recording mode2. (Figure 37 AK4951 datasheet, Table 27. Recording Playback Mode)
-	// When changing those modes, PMPFIL bit must be “0”, it is OK (*1)
-	map.r.digital_filter_mode.ADCPF = 1;	// ADCPF bit swith ("0" Mic  after ADC Output connected (recording mode) to the DIGITAL FILTER  BLOCK. ("1" Playback mode)  
-	map.r.digital_filter_mode.PFSDO = 1;	// ADC (+ 1st order HPF) Output
-    map.r.digital_filter_mode.PFDAC = 0b00;	// (Input selector for DAC (not used in MIC), SDTI= Audio Serial Data Input Pin)
-	update(Register::DigitalFilterMode);	// Writing the Audio Path :  NO DIGITAL BLOCK or DIG BLOCK FOR  MIC ,   Audio mode path : Playback mode /-Recording mode.
-
-	// The EQn (n=1, 2, 3, 4 or 5) coefficient must be set when EQn bit = “0” or PMPFIL bit = “0”., but we are already (*1) 
-	// map.r.power_management_1.PMPFIL = 0;		// In the previous Wind Noise Filter , we already set up PPFIL = 0
-	// update(Register::PowerManagement1);		// Activating the Power management of the used blocks . (Mic ADC always + Dig Block filter , when used )
-
-	// ... Set EQ & LPF  coefficients ---------------------------------
-
-	// writting to the IC ak4951 reg. settings defined in Ak4951.hpp ,  the 30 bytes ,  EQ coefficient =  5 (EQ1,2,3,4,5) x 3 (A,B,C coefficients) x 2 bytes (16 bits)
-    update(Register::E1Coefficient0);	// we could pre-load here,ex ,"map.r.e1_coefficient_0.l = 0x50;" , EQ1 Coefficient A  : A7...A0, but already done in ak4951.hpp
-    update(Register::E1Coefficient1);	// we could pre-load here,ex ,"map.r.e1_coefficient_1.h = 0xFE;" , EQ1 Coefficient A  : A15..A8,  "   "
-    update(Register::E1Coefficient2);	// we could pre-load here,ex ,"map.r.e1_coefficient_2.l = 0x29;" , EQ1 Coefficient B  : B7...B0,  "   "   
-    update(Register::E1Coefficient3);	// we could pre-load here,ex ,"map.r.e1_coefficient_3.h = 0xC5;" , EQ1 Coefficient B  : B15..B8,  "   "
-    update(Register::E1Coefficient4);	// we could pre-load here,ex ,"map.r.e1_coefficient_4.l = 0xA0;" , EQ1 Coefficient C  : C7...C0,  "   "
-    update(Register::E1Coefficient5);	// we could pre-load here,ex ,"map.r.e1_coefficient_5.h = 0x1C;" , EQ1 Coefficient C  : C15..C8,  "   "
-		
-	update(Register::E2Coefficient0);	// writing  pre-loaded EQ2 coefficcients
-    update(Register::E2Coefficient1);	
-    update(Register::E2Coefficient2);
-    update(Register::E2Coefficient3);
-    update(Register::E2Coefficient4);
-    update(Register::E2Coefficient5);
-
-	// Already pre-loaded LPF coefficients to 6k, 3,5k or 4k  ,(LPF 6Khz all digital alc modes top , except when  3k5 , 4k) 
-	update(Register::LPFCoefficient0);	// Writing  pre-loaded  4 bytes  LPF  CoefFiecients 14 bits (FSA13..0,  FSB13..0
-	update(Register::LPFCoefficient1);
-	update(Register::LPFCoefficient2);
-	update(Register::LPFCoefficient3);
-
-    // Activating LPF block , (and re-configuring the rest of bits of the same register)
-	map.r.digital_filter_select_2.HPF = 0;    // HPF2-Block, Coeffic Setting Enable (OFF-Default), When HPF bit is “0”, audio data passes the HPF2 block by is 0dB gain. 
-	map.r.digital_filter_select_2.LPF = 1;	  // LPF-Block, Coeffic Setting Enable (OFF-Default), When LPF bit is “0”, audio data passes the LPF block by 0dB gain. 	
-	map.r.digital_filter_select_2.FIL3 = 0;	  // Stereo_Emphasis_Filter-Block,(OFF-Default)  Coefficient Setting Enable , OFF , Disable.	
-	map.r.digital_filter_select_2.EQ0 = 0;    // Gain Compensation-Block, (OFF-Default) Coeffic Setting Enable, When EQ0 bit = “0” audio data passes the EQ0 block by 0dB gain. 
-	map.r.digital_filter_select_2.GN = 0b00;  // Gain Setting of the Gain Compensation Block Default: “00”-Default (0dB)	
+*/
+// map.r.l_ch_input_volume_control.IV = 0xe1;
+// update(Register::LchInputVolumeControl);
+// map.r.r_ch_input_volume_control.IV = 0xe1;
+// update(Register::RchInputVolumeControl);
+/*
+	map.r.auto_hpf_control.STG = 0b00;
+	map.r.auto_hpf_control.SENC = 0b011;
+	map.r.auto_hpf_control.AHPF = 0;
+	update(Register::AutoHPFControl);
+*/
+	map.r.digital_filter_select_1.HPFAD = 1;	// HPF1 (after ADC) = on
+	map.r.digital_filter_select_1.HPFC = 0b11;	// 2336.8 Hz @ fs=48k
+	update(Register::DigitalFilterSelect1);
+/*
+	map.r.digital_filter_select_2.HPF = 0;
+	map.r.digital_filter_select_2.LPF = 0;
+	map.r.digital_filter_select_2.FIL3 = 0;
+	map.r.digital_filter_select_2.EQ0 = 0;
+	map.r.digital_filter_select_2.GN = 0b00;
 	update(Register::DigitalFilterSelect2);
 
-		// Acitivating digital block , power supply 
-	map.r.power_management_1.PMADL  = 1;	// ADC Lch = Lch input signal. Mic Amp Lch and ADC Lch Power Management
-	map.r.power_management_1.PMADR  = 1;	// ADC Rch = Rch input signal. Mic Amp Rch and ADC Rch Power Management
-	map.r.power_management_1.PMPFIL = 1;	// Pre-loaded in top part.  Orig value=0, Programmable Digital filter unused (not power up), routed around. 
-	update(Register::PowerManagement1);		// Activating the Power management of the used blocks . (Mic ADC always + Dig Block filter , when used )
+	map.r.digital_filter_select_3.EQ1 = 0;
+	map.r.digital_filter_select_3.EQ2 = 0;
+	map.r.digital_filter_select_3.EQ3 = 0;
+	map.r.digital_filter_select_3.EQ4 = 0;
+	map.r.digital_filter_select_3.EQ5 = 0;
+	update(Register::DigitalFilterSelect3);
+*/
+	map.r.digital_filter_mode.PFSDO = 0;	// ADC (+ 1st order HPF) Output
+	map.r.digital_filter_mode.ADCPF = 1;	// ADC Output (default)
+	update(Register::DigitalFilterMode);
+
+	// ... Set coefficients ...
+
+	map.r.power_management_1.PMADL = 1;		// ADC Lch = Lch input signal
+	map.r.power_management_1.PMADR = 1;		// ADC Rch = Rch input signal
+	map.r.power_management_1.PMPFIL = 0;	// Programmable filter unused, routed around.
+	update(Register::PowerManagement1);
 
 	// 1059/fs, 22ms @ 48kHz
 	chThdSleepMilliseconds(22);
-
-  } 
-
-  // Common  part  for all alc_mode , --------------------------
-  // const uint_fast8_t mgain = 0b0111;		// Already pre-loaded , in above switch case . 
-  map.r.signal_select_1.MGAIN20 = mgain & 7;  // writing 3 lower bits of mgain , (pre-amp mic gain).  
-  map.r.signal_select_1.PMMP = 1;				// Activating DC Mic Power supply through 2kohms res., similar majority  smartphones  headphone+mic jack, "plug-in-power"
-  map.r.signal_select_1.MPSEL = 1;			// MPWR2 pin ,selecting output voltage to  MPWR2 pin, that we are using in portapack ext. MIC)
-  map.r.signal_select_1.MGAIN3 = (mgain >> 3) & 1; // writing 4th upper bit of mgain  (pre-amp mic gain).
-  update(Register::SignalSelect1);
-	
 }
 
-
-
 void AK4951::microphone_disable() {
-	map.r.power_management_1.PMADL = 0;		// original code  , disable Power managem.Mic  ADC L
-	map.r.power_management_1.PMADR = 0;		// original code  , disable Power managem.Mic  ADC R
-	map.r.power_management_1.PMPFIL = 0;	// original code  , disable Power managem. all Programmable Dig. block  
+	map.r.power_management_1.PMADL = 0;
+	map.r.power_management_1.PMADR = 0;
+	map.r.power_management_1.PMPFIL = 0;
 	update(Register::PowerManagement1);
 
-	map.r.alc_mode_control_1.ALC = 0;		// original code  , Restore , disable ALC block.
+	map.r.alc_mode_control_1.ALC = 0;
 	update(Register::ALCModeControl1);
-
-	map.r.auto_hpf_control.AHPF = 0;    	//----------- new code addition , Restore disable Wind noise filter OFF (AHPF bit=“0”). 
-	update(Register::AutoHPFControl); 
-
-	//Restore original AUDIO PATH , condition,  (Digital filter block PATH is BY PASSED)  (we can also swith off DIG. BLOCK power , PMPFIL=0)
-	// The Path in Recording Mode 2 & Playback Mode 2 , (NO DIG FILTER BLOCK AT ALL, not for MIC recording, nor for Playback)
-	map.r.digital_filter_mode.ADCPF = 1;	// new code addition , ADCPF bit swith ("0" Mic  after ADC Output connected (recording mode) to the DIGITAL FILTER  BLOCK. ("1" Playback mode)  
-	map.r.digital_filter_mode.PFSDO = 0;	// new code addition , ADC bit switch ("0" : 1st order HPF) connectedto the  Output. By bass DIGITAL block .
-   	map.r.digital_filter_mode.PFDAC = 0b00;	// new code addition , (Input selector for DAC (not used in MIC), SDTI= Audio Serial Data Input Pin)
-	update(Register::DigitalFilterMode);	// Writing the Audio Path :  NO DIGITAL BLOCK or DIG BLOCK FOR  MIC ,   Audio mode path : Playback mode /-Recording mode.
-	
-	// Restore original condition , LPF , OFF . same as when not using DIGITAL Programmable block
-	map.r.digital_filter_select_2.LPF = 0;	  // LPF-Block, Coeffic Setting Enable (OFF-Default), When LPF bit is “0”, audio data passes the LPF block by 0dB gain. 
-	update(Register::DigitalFilterSelect2);
-
-	map.r.lpf_coefficient_0.l = 0x00;		// Pre-loading here LPF 6k  ,  1st Order from digital Block , Fc=6000 Hz,  fs = 48khz
-	map.r.lpf_coefficient_1.h = 0x00;		// LPF bit is activated down, for all ALC digital modes.
-	map.r.lpf_coefficient_2.l = 0x00;		// Writting reg to AK4951 , down with update....
-	map.r.lpf_coefficient_3.h = 0x00;
-
-	update(Register::LPFCoefficient0);		// Writing  pre-loaded  4 bytes  LPF  CoefFiecients 14 bits (FSA13..0,  FSB13..0
-	update(Register::LPFCoefficient1);
-	update(Register::LPFCoefficient2);
-	update(Register::LPFCoefficient3);
-
-	// Switch off all EQ 1,2,3,4,5
-	map.r.digital_filter_select_3.EQ1 = 0;	  // EQ1 Coeffic Setting , (0: Disable-default, audio data passes  EQ1 block by 0dB gain). When EQ1="1”, the settings of E1A15-0, E1B15-0 and E1C15-0 bits are enabled
-	map.r.digital_filter_select_3.EQ2 = 0;	  // EQ2 Coeffic Setting , (0: Disable-default, audio data passes  EQ2 block by 0dB gain). When EQ2="1”, the settings of E2A15-0, E2B15-0 and E2C15-0 bits are enabled
-	map.r.digital_filter_select_3.EQ3 = 0;    // EQ3 Coeffic Setting , (0: Disable-default, audio data passes  EQ3 block by 0dB gain). When EQ3="1”, the settings of E3A15-0, E3B15-0 and E3C15-0 bits are enabled
-	map.r.digital_filter_select_3.EQ4 = 0;    // EQ4 Coeffic Setting , (0: Disable-default, audio data passes  EQ4 block by 0dB gain). When EQ4="1”, the settings of E4A15-0, E4B15-0 and E4C15-0 bits are enabled
-	map.r.digital_filter_select_3.EQ5 = 0;    // EQ5 Coeffic Setting , (0: Disable-default, audio data passes  EQ5 block by 0dB gain). When EQ5="1”, the settings of E5A15-0, E5B15-0 and E5C15-0 bits are enabled
-	update(Register::DigitalFilterSelect3);
-
 }
 
 reg_t AK4951::read(const address_t reg_address) {
diff --git a/firmware/common/ak4951.hpp b/firmware/common/ak4951.hpp
index 3b1f9e1a..df168a09 100644
--- a/firmware/common/ak4951.hpp
+++ b/firmware/common/ak4951.hpp
@@ -773,41 +773,40 @@ constexpr RegisterMap default_after_reset { Register_Type {
 		.REV = 0b1100,
 	},
 	
-	// just pre-loading into memory, 30 bytes =  EQ 1,2,3,4,5  x A,B,C (2 x bytes) coefficients, but it will be written from ak4951.cpp 
-	.e1_coefficient_0 = { .l = 0xCA },		//EQ1 Coefficient A  : A7...A0,		BW : 300Hz - 1700Hz  (fo = 1150Hz , fb= 1700Hz) , k=1,8 peaking 
-	.e1_coefficient_1 = { .h = 0x05 },		//EQ1 Coefficient A  : A15..A8
-	.e1_coefficient_2 = { .l = 0xEB },		//EQ1 Coefficient B  : B7...B0
-	.e1_coefficient_3 = { .h = 0x38 },		//EQ1 Coefficient B  : B15...B8
-	.e1_coefficient_4 = { .l = 0x6F },		//EQ1 Coefficient C  : C7...C0
-	.e1_coefficient_5 = { .h = 0xE6 },		//EQ1 Coefficient C  : C15..C8
+	.e1_coefficient_0 = { .l = 0x00 },
+	.e1_coefficient_1 = { .h = 0x00 },
+	.e1_coefficient_2 = { .l = 0x00 },
+	.e1_coefficient_3 = { .h = 0x00 },
+	.e1_coefficient_4 = { .l = 0x00 },
+	.e1_coefficient_5 = { .h = 0x00 },
 	
-	.e2_coefficient_0 = { .l = 0x05 },		//EQ2 Coefficient A  : A7...A0,		BW : 250Hz - 2700Hz  (fo = 1475Hz , fb= 2450Hz) , k=1,8 peaking  
-	.e2_coefficient_1 = { .h = 0x08 },		//EQ2 Coefficient A  : A15..A8
-	.e2_coefficient_2 = { .l = 0x11 },		//EQ2 Coefficient B  : B7...B0
-	.e2_coefficient_3 = { .h = 0x36 },		//EQ2 Coefficient B  : B15...B8
-	.e2_coefficient_4 = { .l = 0xE9 },		//EQ2 Coefficient C  : C7...C0
-	.e2_coefficient_5 = { .h = 0xE8 },		//EQ2 Coefficient C  : C15..C8
+	.e2_coefficient_0 = { .l = 0x00 },
+	.e2_coefficient_1 = { .h = 0x00 },
+	.e2_coefficient_2 = { .l = 0x00 },
+	.e2_coefficient_3 = { .h = 0x00 },
+	.e2_coefficient_4 = { .l = 0x00 },
+	.e2_coefficient_5 = { .h = 0x00 },
 	
-	.e3_coefficient_0 = { .l = 0x00 },		//EQ3 Coefficient A  : A7...A0,		not used  currently 
-	.e3_coefficient_1 = { .h = 0x00 },		//EQ3 Coefficient A  : A15..A8
-	.e3_coefficient_2 = { .l = 0x00 },		//EQ3 Coefficient B  : B7...B0
-	.e3_coefficient_3 = { .h = 0x00 },		//EQ3 Coefficient B  : B15...B8
-	.e3_coefficient_4 = { .l = 0x00 },		//EQ3 Coefficient C  : C7...C0
-	.e3_coefficient_5 = { .h = 0x00 },		//EQ3 Coefficient C  : C15..C8
+	.e3_coefficient_0 = { .l = 0x00 },
+	.e3_coefficient_1 = { .h = 0x00 },
+	.e3_coefficient_2 = { .l = 0x00 },
+	.e3_coefficient_3 = { .h = 0x00 },
+	.e3_coefficient_4 = { .l = 0x00 },
+	.e3_coefficient_5 = { .h = 0x00 },
 	
-	.e4_coefficient_0 = { .l = 0x00 },		//EQ4 Coefficient A  : A7...A0,		not used  currently
-	.e4_coefficient_1 = { .h = 0x00 },		//EQ4 Coefficient A  : A15..A8
-	.e4_coefficient_2 = { .l = 0x00 },		//EQ4 Coefficient B  : B7...B0
-	.e4_coefficient_3 = { .h = 0x00 },		//EQ4 Coefficient B  : B15...B8
-	.e4_coefficient_4 = { .l = 0x00 },		//EQ4 Coefficient C  : C7...C0
-	.e4_coefficient_5 = { .h = 0x00 },		//EQ4 Coefficient C  : C15..C8
+	.e4_coefficient_0 = { .l = 0x00 },
+	.e4_coefficient_1 = { .h = 0x00 },
+	.e4_coefficient_2 = { .l = 0x00 },
+	.e4_coefficient_3 = { .h = 0x00 },
+	.e4_coefficient_4 = { .l = 0x00 },
+	.e4_coefficient_5 = { .h = 0x00 },
 
-	.e5_coefficient_0 = { .l = 0x00 },		//EQ5 Coefficient A  : A7...A0,		not used currently
-	.e5_coefficient_1 = { .h = 0x00 },		//EQ5 Coefficient A  : A15..A8
-	.e5_coefficient_2 = { .l = 0x00 },		//EQ5 Coefficient B  : B7...B0
-	.e5_coefficient_3 = { .h = 0x00 },		//EQ5 Coefficient B  : B15...B8
-	.e5_coefficient_4 = { .l = 0x00 },		//EQ5 Coefficient C  : C7...C0
-	.e5_coefficient_5 = { .h = 0x00 },		//EQ5 Coefficient C  : C15..C8
+	.e5_coefficient_0 = { .l = 0x00 },
+	.e5_coefficient_1 = { .h = 0x00 },
+	.e5_coefficient_2 = { .l = 0x00 },
+	.e5_coefficient_3 = { .h = 0x00 },
+	.e5_coefficient_4 = { .l = 0x00 },
+	.e5_coefficient_5 = { .h = 0x00 },
 } };
 
 class AK4951 : public audio::Codec  {
@@ -842,7 +841,7 @@ public:
 	void set_headphone_volume(const volume_t volume) override;
 	void headphone_mute();
 
-	void microphone_enable(int8_t alc_mode); 	// added user GUI parameter , to set up AK4951 ALC mode.
+	void microphone_enable();
 	void microphone_disable();
 
 	size_t reg_count() const override {
diff --git a/firmware/common/wm8731.hpp b/firmware/common/wm8731.hpp
index ac164a04..a5c8908b 100644
--- a/firmware/common/wm8731.hpp
+++ b/firmware/common/wm8731.hpp
@@ -345,9 +345,8 @@ public:
  	void speaker_disable() {};
 
 
-	void microphone_enable(int8_t alc_mode) override {
-		(void)alc_mode; 		// to avoid "unused warning" when compiling. (@WM8731 we do not use that parameter) 
-		// TODO: Implement,
+	void microphone_enable() override {
+		// TODO: Implement
 	}
 
 	void microphone_disable() override {