Merge branch 'eried:next' into next

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Brumi-2021 2022-05-08 12:39:27 +02:00 committed by GitHub
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13 changed files with 341 additions and 416 deletions

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@ -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({

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@ -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() {

View File

@ -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();
}

View File

@ -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 */

View File

@ -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) {

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@ -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 */

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@ -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;

View File

@ -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 };

View File

@ -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_);
}

View File

@ -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__ */

View File

@ -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 REFs
// 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 REFs
// 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 REFs
// 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 REFs
//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 REFs
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 REFs
// 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 REFs
// 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 REFs
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 REFs
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 REFs
// 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 REFs
// 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) {

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@ -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 {

View File

@ -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 {