diff --git a/firmware/application/apps/soundboard_app.cpp b/firmware/application/apps/soundboard_app.cpp
index 02039ae3..344afd35 100644
--- a/firmware/application/apps/soundboard_app.cpp
+++ b/firmware/application/apps/soundboard_app.cpp
@@ -111,6 +111,7 @@ void SoundBoardView::start_tx(const uint32_t id) {
 		1536000 / 20,		// Update vu-meter at 20Hz
 		transmitter_model.channel_bandwidth(),
 		0,	// Gain is unused
+		8,	// shift_bits_s16, default 8 bits, but  also unused
 		TONES_F2D(tone_key_frequency(tone_key_index), 1536000),
 		0, //AM
 		0, //DSB
diff --git a/firmware/application/apps/ui_mictx.cpp b/firmware/application/apps/ui_mictx.cpp
index 397b595b..ef8a4cf6 100644
--- a/firmware/application/apps/ui_mictx.cpp
+++ b/firmware/application/apps/ui_mictx.cpp
@@ -74,6 +74,7 @@ void MicTXView::configure_baseband() {
 		sampling_rate / 20,		// Update vu-meter at 20Hz
 		transmitting ? transmitter_model.channel_bandwidth() : 0,
 		mic_gain,
+		shift_bits_s16,			// to be used in dsp_modulate
 		TONES_F2D(tone_key_frequency(tone_key_index), sampling_rate),
 		enable_am,
 		enable_dsb,
@@ -188,7 +189,7 @@ void MicTXView::rxaudio(bool is_on) {
 
 		baseband::run_image(portapack::spi_flash::image_tag_mic_tx);
 		audio::output::stop();
-		audio::input::start();	// set up audio input =  mic config of any audio coded AK4951/WM8731, (in WM8731 parameter will be ignored)
+		audio::input::start(ak4951_alc_and_wm8731_boost_GUI);	// When detected AK4951 => set up ALC mode; when detected WM8731 => set up mic_boost ON/OFF.
 		portapack::pin_i2s0_rx_sda.mode(3);
 		configure_baseband();
 	}
@@ -213,12 +214,12 @@ MicTXView::MicTXView(
 	
 	baseband::run_image(portapack::spi_flash::image_tag_mic_tx);
 	
-	
-	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() ) {
+	if (audio::debug::codec_name() =="WM8731" ) {
 	add_children({		
 		&labels_WM8731,		// we have audio codec WM8731, same MIC menu  as original.
 		&vumeter,
 		&options_gain,		// MIC GAIN float factor on the GUI.
+		&options_wm8731_boost_mode,
 //		&check_va,
 		&field_va,
 		&field_va_level,
@@ -283,13 +284,43 @@ MicTXView::MicTXView(
 		mic_gain = v / 10.0;
 		configure_baseband();
 	};
-	options_gain.set_selected_index(1);		// x1.0
+	options_gain.set_selected_index(1);		// x1.0  preselected default.
+
+ 	if (audio::debug::codec_name() =="WM8731") {
+		options_wm8731_boost_mode.on_change = [this](size_t, int8_t v) {
+
+			switch(v) {
+			case 0: 	                // +12 dB’s respect reference level orig fw 1.5.x fw FM : when +20dB's boost ON) and shift bits (>>8), 
+				shift_bits_s16 = 6; 	//  now mic-boost on (+20dBs) and shift bits (>>6), +20+12=32 dB’s (orig fw +20 dBs+ 0dBs)=> +12dB's respect ref.
+				break;					
+			case 1: 	                // +06 dB’s reference level , (when +20dB's boost ON) 
+				shift_bits_s16 = 7; 	// now mic-boost on (+20dBs) and shift bits (>>7), +20+06=26 dB’s (orig fw +20 dBs+ 0dBs) => +06dB's respect ref.	 
+				break;					
+			case 2: 	                
+				shift_bits_s16 = 4; 	// +04 dB’s respect ref level , (when +20dB's boost OFF) 
+				break;					// now mic-boost off (+00dBs) shift bits (4) (+0+24dB's)=24 dBs => +04dB's respect ref.	 
+			case 3:
+				shift_bits_s16 = 5;		// -02 dB’s respect ref level , (when +20dB's boost OFF) 
+				break;					// now mic-boost off (+00dBs) shift bits (5) (+0+18dB's)=18 dBs => -02dB's respect ref.	 
+			case 4:
+				shift_bits_s16 = 6;     // -08 dB’s respect ref level , (when +20dB's boost OFF) 
+				break;					// now mic-boost off (+00dBs) shift bits (6) (+0+12dB's)=12 dBs => -08dB's respect ref.	
+			}
+			ak4951_alc_and_wm8731_boost_GUI   = v;	//  0,..4 WM8731_boost dB's options, (combination boost on/off , and effective gain in captured data >>x) 
+			audio::input::start(ak4951_alc_and_wm8731_boost_GUI);  // Detected (WM8731) , set up the proper wm_boost on/off , 0..4 (0,1) boost_on , (2,3,4) boost_0ff
+			configure_baseband(); 					// to update in real timme,sending msg ,  var-parameters >>shift_bits FM msg ,to audio_tx from M0 to M4 Proc -
+		};
+		options_wm8731_boost_mode.set_selected_index(3);	// preset GUI index 3 as default WM -> -02 dB's .	
+	} else {
+	   	shift_bits_s16 = 8;         	// Initialized default fixed >>8_FM for FM tx mod ,  shift audio data for AK4951  ,using top 8 bits s16 data (>>8)
+	  	options_ak4951_alc_mode.on_change = [this](size_t, int8_t v) {
+	    	ak4951_alc_and_wm8731_boost_GUI   = v;   				// 0,..11,  AK4951 Mic -Automatic volume Level Control  options, 
+			audio::input::start(ak4951_alc_and_wm8731_boost_GUI);  	// Detected (AK4951) ==> Set up proper ALC mode from 0..11 options
+			configure_baseband(); 									// sending fixed >>8_FM , var-parameters msg ,  to audiotx from this M0 to M4 process.
+		};
+	}
 	
-	options_ak4951_alc_mode.on_change = [this](size_t, int8_t v) {
-	      ak4951_alc_GUI_selected  = v;
-		  audio::input::start();
-	};
-	// options_ak4951_alc_mode.set_selected_index(0);
+   	// options_ak4951_alc_mode.set_selected_index(0);
 	 
 	tx_frequency = transmitter_model.tuning_frequency();
 	field_frequency.set_value(transmitter_model.tuning_frequency());
@@ -539,9 +570,9 @@ MicTXView::MicTXView(
 	transmitter_model.set_baseband_bandwidth(1750000);
 	
 	set_tx(false);
-	
-	audio::set_rate(audio::Rate::Hz_24000);
-	audio::input::start();		// originally , audio::input::start();  (we added parameter)
+
+  	audio::set_rate(audio::Rate::Hz_24000);
+	audio::input::start(ak4951_alc_and_wm8731_boost_GUI);	// When detected AK4951 => set up ALC mode; when detected WM8731 => set up mic_boost ON/OFF.	
 }
 
 MicTXView::~MicTXView() {
diff --git a/firmware/application/apps/ui_mictx.hpp b/firmware/application/apps/ui_mictx.hpp
index 9b091e53..be3e397f 100644
--- a/firmware/application/apps/ui_mictx.hpp
+++ b/firmware/application/apps/ui_mictx.hpp
@@ -83,7 +83,7 @@ private:
 	bool rx_enabled { false };
 	uint32_t tone_key_index { };
 	float mic_gain { 1.0 };
-    uint8_t  ak4951_alc_GUI_selected { 0 }; 
+    uint8_t  ak4951_alc_and_wm8731_boost_GUI  { 0 }; 
 	uint32_t audio_level { 0 };
 	uint32_t va_level { };
 	uint32_t attack_ms { };
@@ -99,6 +99,7 @@ private:
 	rf::Frequency rx_frequency { 0 };
 	int32_t focused_ui { 2 };
 	bool button_touch { false };
+	uint8_t shift_bits_s16 {4} ; // shift bits factor to the captured ADC S16 audio sample.
 
 	//AM TX Stuff
 	bool enable_am { false };
@@ -109,6 +110,7 @@ private:
 	
 	Labels labels_WM8731 {		
 		{ { 3 * 8, 1 * 8 }, "MIC-GAIN:", Color::light_grey() },
+		{ { 17 * 8, 1 * 8 }, "Boost", Color::light_grey() },
 		{ { 3 * 8, 3 * 8 }, "F:", Color::light_grey() },
 		{ { 15 * 8, 3 * 8 }, "BW:   FM kHz", Color::light_grey() },
 		{ { 3 * 8, 5 * 8 }, "GAIN:", Color::light_grey() },
@@ -171,7 +173,7 @@ private:
 		{ 20 * 8, 1 * 8 },				// Coordinates are: int:x (px), int:y (px)				
 		11,
 		{
-			{ " OFF-20kHz", 0 },	// Nothing changed from ORIGINAL,keeping ALL programm. AK4951 Dig. block->OFF)
+			{ " OFF-12kHz", 0 },	// Nothing changed from ORIGINAL,keeping ALL programmable  AK4951 Digital Block->OFF, sampling 24Khz)
 			{ "+12dB-6kHz", 1 },	// ALC-> on, (+12dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
 			{ "+09dB-6kHz", 2 },	// ALC-> on, (+09dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
 			{ "+06dB-6kHz", 3 },	// ALC-> on, (+06dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
@@ -186,6 +188,18 @@ private:
 		}
 	};
 
+OptionsField options_wm8731_boost_mode {
+		{ 22 * 8, 1 * 8 },				// Coordinates are: int:x (px), int:y (px)				
+		5,
+		{
+			{ "ON +12dB", 0 },	// WM8731 Mic Boost ON ,original+12dBs condition, easy to saturate ADC sat in high voice  ,relative G = +12 dB's respect ref level
+			{ "ON +06dB", 1 },	// WM8731 Mic Boost ON ,original+6 dBs condition, easy to saturate ADC sat in high voice  ,relative G = +06 dB's respect ref level
+			{ "OFF+04dB", 2 },	// WM8731 Mic Boost OFF to avoid ADC sat in high voice  ,relative  G =  +04 dB's (respect ref level) , always effective sampling 24khz 
+			{ "OFF-02dB", 3 },	// WM8731 Mic Boost OFF to avoid ADC sat in high voice  ,relative  G =  -02 dB's (respect ref level) 
+			{ "OFF-08dB", 4 },	// WM8731 Mic Boost OFF to avoid ADC sat in high voice  ,relative  G =  -12 dB's (respect ref level) 
+		}
+	};
+
 	FrequencyField field_frequency {
 		{ 5 * 8, 3 * 8 },
 	};
diff --git a/firmware/application/audio.cpp b/firmware/application/audio.cpp
index 4ba23acc..ce32076c 100644
--- a/firmware/application/audio.cpp
+++ b/firmware/application/audio.cpp
@@ -168,8 +168,8 @@ void speaker_mute() {
 
 namespace input {
 
-void start() {
-	audio_codec->microphone_enable();
+void start(int8_t alc_mode) {
+	audio_codec->microphone_enable(alc_mode);   // added user-GUI selection for AK4951, ALC mode parameter.
 	i2s::i2s0::rx_start();
 }
 
diff --git a/firmware/application/audio.hpp b/firmware/application/audio.hpp
index b02e42aa..56301ee8 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() = 0;
+	virtual void microphone_enable(int8_t alc_mode) = 0;	// added user-GUI  AK4951 ,selected ALC mode.
 	virtual void microphone_disable() = 0;
 
 	virtual size_t reg_count() const = 0;
@@ -59,7 +59,7 @@ public:
 
 namespace output {
 
-void start();
+void start();		// this  other start(),no changed. ,in namespace output , used to config audio playback mode, 
 void stop();
 
 void mute();
@@ -72,7 +72,7 @@ void speaker_unmute();
 
 namespace input {
 
-void start();
+void start(int8_t alc_mode);  // added parameter user-GUI select AK4951-ALC mode for config mic path,(recording mode in datasheet),
 void stop();
 
 } /* namespace input */
diff --git a/firmware/application/baseband_api.cpp b/firmware/application/baseband_api.cpp
index 74f472d8..ec7ba1a6 100644
--- a/firmware/application/baseband_api.cpp
+++ b/firmware/application/baseband_api.cpp
@@ -183,12 +183,13 @@ void kill_afsk() {
 }
 
 void set_audiotx_config(const uint32_t divider, const float deviation_hz, const float audio_gain,
-					const uint32_t tone_key_delta, const bool am_enabled, const bool dsb_enabled,
-					const bool usb_enabled, const bool lsb_enabled) {
+					uint8_t audio_shift_bits_s16, const uint32_t tone_key_delta, const bool am_enabled, 
+					const bool dsb_enabled, const bool usb_enabled, const bool lsb_enabled) {
 	const AudioTXConfigMessage message {
 		divider,
 		deviation_hz,
 		audio_gain,
+		audio_shift_bits_s16,
 		tone_key_delta,
 		(float)persistent_memory::tone_mix() / 100.0f,
 		am_enabled,
diff --git a/firmware/application/baseband_api.hpp b/firmware/application/baseband_api.hpp
index bcc43375..5c9b3c95 100644
--- a/firmware/application/baseband_api.hpp
+++ b/firmware/application/baseband_api.hpp
@@ -61,8 +61,8 @@ void set_tones_config(const uint32_t bw, const uint32_t pre_silence, const uint1
 void kill_tone();
 void set_sstv_data(const uint8_t vis_code, const uint32_t pixel_duration);
 void set_audiotx_config(const uint32_t divider, const float deviation_hz, const float audio_gain,
-			const uint32_t tone_key_delta, const bool am_enabled, const bool dsb_enabled,
-			const bool usb_enabled, const bool lsb_enabled);
+			uint8_t audio_shift_bits_s16, const uint32_t tone_key_delta, const bool am_enabled, 
+			const bool dsb_enabled,	const bool usb_enabled, const bool lsb_enabled);
 void set_fifo_data(const int8_t * data);
 void set_pitch_rssi(int32_t avg, bool enabled);
 void set_afsk_data(const uint32_t afsk_samples_per_bit, const uint32_t afsk_phase_inc_mark, const uint32_t afsk_phase_inc_space,
diff --git a/firmware/baseband/dsp_modulate.cpp b/firmware/baseband/dsp_modulate.cpp
index 04102800..34633715 100644
--- a/firmware/baseband/dsp_modulate.cpp
+++ b/firmware/baseband/dsp_modulate.cpp
@@ -42,10 +42,16 @@ 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 ;    
+void Modulator::set_gain_shiftbits_vumeter_beep(float new_audio_gain ,uint8_t new_audio_shift_bits_s16, bool new_play_beep ) {
+	//new_audio_shift_bits_s16 are the direct shift bits (FM mod >>x) , and it is fixed to >>8_FM (AK) or  4,5,6, (WM boost OFF) or 6,7 (WM boost ON)
+      audio_gain = new_audio_gain ;
+	  audio_shift_bits_s16_FM = new_audio_shift_bits_s16;  //FM :        >>8(AK) fixed ,  >>4,5,6 (WM boost OFF)
+	  if (new_audio_shift_bits_s16==8) {				   //FM : we are in AK codec IC => for AM-SSB-DSB we were using >>2 fixed (wm boost ON) .	
+			audio_shift_bits_s16_AM_DSB_SSB = 2;	       //AM-DSB-SSB: >>2(AK) fixed ,  >>0,1,2 (WM boost OFF)	  
+	  } else {
+		    audio_shift_bits_s16_AM_DSB_SSB = (new_audio_shift_bits_s16-4) ; //AM-DSB-SSB: >>0,1,2 (WM boost OFF), >>2,3 (WM boost ON)	  
+	  }
 	  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 ) {
@@ -85,7 +91,7 @@ void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& co
 		if (counter % 128 == 0) {
 			float	i = 0.0, q = 0.0;
 
-		    sample = audio.p[counter / over] >> 2;
+		    sample = audio.p[counter / over] >> audio_shift_bits_s16_AM_DSB_SSB;			// originally fixed   >> 2, now >>2 for AK,   0,1,2,3 for WM (boost off)
 			sample *= audio_gain;      //  Apply GAIN  Scale factor to the audio TX modulation.
 	
 				//switch (mode) {
@@ -145,7 +151,7 @@ void FM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& con
 
 	for (size_t counter = 0; counter < buffer.count; counter++) {
 	 
-	    sample = audio.p[counter>>6] >> 8;			// 	sample = audio.p[counter / over] >> 8;   (not enough efficient running code, over = 1536000/240000= 64 )
+	    sample = audio.p[counter>>6] >> audio_shift_bits_s16_FM ;			// Orig. >>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) {
@@ -190,7 +196,7 @@ void AM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& con
 	
 	for (size_t counter = 0; counter < buffer.count; counter++) {
 		if (counter % 128 == 0) {
-			sample = audio.p[counter / over] >> 2;
+			sample = audio.p[counter / over] >> audio_shift_bits_s16_AM_DSB_SSB; // originally fixed >> 2, now >>2 for AK, 0,1,2,3 for WM (boost off)
 			sample *= audio_gain;      					 // Apply GAIN  Scale factor to the audio TX modulation.
 		}
 
diff --git a/firmware/baseband/dsp_modulate.hpp b/firmware/baseband/dsp_modulate.hpp
index de964a68..e8a2c432 100644
--- a/firmware/baseband/dsp_modulate.hpp
+++ b/firmware/baseband/dsp_modulate.hpp
@@ -50,9 +50,11 @@ public:
 	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 );
+	void set_gain_shiftbits_vumeter_beep(float new_audio_gain ,uint8_t new_audio_shift_bits_s16, 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 { };
+	uint8_t audio_shift_bits_s16_FM { }; 		// shift bits factor to the captured ADC S16 audio sample.
+	uint8_t	audio_shift_bits_s16_AM_DSB_SSB { };
 	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
diff --git a/firmware/baseband/proc_mictx.cpp b/firmware/baseband/proc_mictx.cpp
index 6c9c9b97..1ee5b94d 100644
--- a/firmware/baseband/proc_mictx.cpp
+++ b/firmware/baseband/proc_mictx.cpp
@@ -35,7 +35,7 @@ void MicTXProcessor::execute(const buffer_c8_t& buffer){
 	if (!configured) return;
 	
 	audio_input.read_audio_buffer(audio_buffer);
-	modulator->set_gain_vumeter_beep(audio_gain, play_beep ) ;	
+	modulator->set_gain_shiftbits_vumeter_beep(audio_gain, audio_shift_bits_s16, 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
@@ -141,6 +141,7 @@ void MicTXProcessor::on_message(const Message* const msg) {
 			}
 			
 			audio_gain = config_message.audio_gain;
+			audio_shift_bits_s16 = config_message.audio_shift_bits_s16;
 			divider = config_message.divider;
 			power_acc_count = 0;
 
diff --git a/firmware/baseband/proc_mictx.hpp b/firmware/baseband/proc_mictx.hpp
index 175e53c7..59a07cb0 100644
--- a/firmware/baseband/proc_mictx.hpp
+++ b/firmware/baseband/proc_mictx.hpp
@@ -61,6 +61,8 @@ private:
 
 	uint32_t divider { };
 	float audio_gain { };
+	uint8_t audio_shift_bits_s16 { } ; // shift bits factor to the captured ADC S16 audio sample.
+
 	uint64_t power_acc { 0 };
 	uint32_t power_acc_count { 0 };
 	bool play_beep { false };
diff --git a/firmware/common/ak4951.cpp b/firmware/common/ak4951.cpp
index 74301ecb..1f88a3d6 100644
--- a/firmware/common/ak4951.cpp
+++ b/firmware/common/ak4951.cpp
@@ -216,90 +216,347 @@ void AK4951::speaker_disable() {
 	set_speaker_power(false);
 }
 
-void AK4951::microphone_enable() {
-// map.r.digital_mic.DMIC = 0;
-// update(Register::DigitalMic);
+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
 	
-	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);
+			// 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;
 
-	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);
+		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.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);
+			// 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;
 
-	map.r.alc_timer_select. = ?;
-	update(Register::ALCTimerSelect);
-	map.r.alc_mode_control_1. = ?;
-	map.r.alc_mode_control_1.ALC = 1;
+		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.alc_mode_control_2.REF = ?;
+	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.
+	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.)
+	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) 
+	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) 
 	update(Register::ALCModeControl2);
-*/
-// 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;
+
+	//   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)	
 	update(Register::DigitalFilterSelect2);
 
-	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);
+		// 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 )
 
 	// 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;
-	map.r.power_management_1.PMADR = 0;
-	map.r.power_management_1.PMPFIL = 0;
+	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  
 	update(Register::PowerManagement1);
 
-	map.r.alc_mode_control_1.ALC = 0;
+	map.r.alc_mode_control_1.ALC = 0;		// original code  , Restore , disable ALC block.
 	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 df168a09..3b1f9e1a 100644
--- a/firmware/common/ak4951.hpp
+++ b/firmware/common/ak4951.hpp
@@ -773,40 +773,41 @@ constexpr RegisterMap default_after_reset { Register_Type {
 		.REV = 0b1100,
 	},
 	
-	.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 },
+	// 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
 	
-	.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 },
+	.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
 	
-	.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 },
+	.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
 	
-	.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 },
+	.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
 
-	.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 },
+	.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
 } };
 
 class AK4951 : public audio::Codec  {
@@ -841,7 +842,7 @@ public:
 	void set_headphone_volume(const volume_t volume) override;
 	void headphone_mute();
 
-	void microphone_enable();
+	void microphone_enable(int8_t alc_mode); 	// added user GUI parameter , to set up AK4951 ALC mode.
 	void microphone_disable();
 
 	size_t reg_count() const override {
diff --git a/firmware/common/message.hpp b/firmware/common/message.hpp
index 172aff88..38bc1217 100644
--- a/firmware/common/message.hpp
+++ b/firmware/common/message.hpp
@@ -868,6 +868,7 @@ public:
 		const uint32_t divider,
 		const float deviation_hz,
 		const float audio_gain,
+		const uint8_t audio_shift_bits_s16,
 		const uint32_t tone_key_delta,
 		const float tone_key_mix_weight,
 		const bool am_enabled,
@@ -878,6 +879,7 @@ public:
 		divider(divider),
 		deviation_hz(deviation_hz),
 		audio_gain(audio_gain),
+		audio_shift_bits_s16(audio_shift_bits_s16),
 		tone_key_delta(tone_key_delta),
 		tone_key_mix_weight(tone_key_mix_weight),
 		am_enabled(am_enabled),
@@ -890,6 +892,7 @@ public:
 	const uint32_t divider;
 	const float deviation_hz;
 	const float audio_gain;
+	const uint8_t audio_shift_bits_s16;
 	const uint32_t tone_key_delta;
 	const float tone_key_mix_weight;
 	const bool am_enabled;
diff --git a/firmware/common/wm8731.cpp b/firmware/common/wm8731.cpp
index 25686c1c..82aefdc8 100644
--- a/firmware/common/wm8731.cpp
+++ b/firmware/common/wm8731.cpp
@@ -84,7 +84,7 @@ void WM8731::init() {
 	});
 
 	write(AnalogAudioPathControl {
-		.micboost = 1,	// Enable 20dB boost
+		.micboost = 0,	// Disable 20dB boost by default
 		.mutemic = 0,	// Disable mute (unmute)
 		.insel = 1,		// Microphone input to ADC
 		.bypass = 0,
diff --git a/firmware/common/wm8731.hpp b/firmware/common/wm8731.hpp
index a5c8908b..bd3d5cb0 100644
--- a/firmware/common/wm8731.hpp
+++ b/firmware/common/wm8731.hpp
@@ -345,18 +345,28 @@ public:
  	void speaker_disable() {};
 
 
-	void microphone_enable() override {
-		// TODO: Implement
-	}
+void microphone_enable(int8_t wm8731_boost_GUI) override {
+		microphone_mute(true); 								// c/m to reduce "plop noise" when changing wm8731_boost_GUI.
+		// chThdSleepMilliseconds(20);  					// does not help to reduce the "plop noise"
+		microphone_boost((wm8731_boost_GUI<2) ? 1 : 0 ); 	// 1 = Enable Boost (+20 dBs) .  0 = Disable Boost (0dBs).
+		chThdSleepMilliseconds(120);						// >50 msegs, very effective , >100 msegs minor improvement ,120 msegs trade off speed .
+		microphone_mute(false); 
+		//	(void)alc_mode; 		In prev. fw version ,  when we did not use at all param., to avoid "unused warning" when compiling.) 
+}
 
 	void microphone_disable() override {
 		// TODO: Implement
 	}
 
-	// void microphone_mute(const bool mute) {
-	// 	map.r.analog_audio_path_control.mutemic = (mute ? 0 : 1);
-	// 	write(Register::AnalogAudioPathControl);
-	// }
+	void microphone_boost(const bool boost) {
+		map.r.analog_audio_path_control.micboost = (boost ? 1 : 0);
+	 	write(Register::AnalogAudioPathControl);
+	}
+
+	void microphone_mute(const bool mute) {
+		map.r.analog_audio_path_control.mutemic = (mute ? 1 : 0);	//1 = Enable Mute ,  0 = Disable Mute
+	 	write(Register::AnalogAudioPathControl);
+	}
 
 	// void set_adc_source(const ADCSource adc_source) {
 	// 	map.r.analog_audio_path_control.insel = toUType(adc_source);