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https://github.com/eried/portapack-mayhem.git
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Recovered lost ctcss/roger beep/correct mic gain in mic app from 1.5.1 without ALC (Auto mic Limit Control-AK) (#633)
* Update spectrum_collector.cpp lower case correction * Update spectrum_collector.cpp Description changed , better explanation. * Revert "Update spectrum_collector.cpp" This reverts commit4a6fc35384. * Revert "Update spectrum_collector.cpp" This reverts commit35cece1cb0. * Revert "Solving Compile error on gcc10 . Keeping same safety protection about the size of the array ,but with slightly different sintax." This reverts commitf4db4e2b53. * Recovered CTCSS-Roger_beep-MIC-GAIN from 1.5.1 * Temporary removing ALC-( for AK4951 platorm)
This commit is contained in:
Brumi-2021
GitHub
parent
c9db1aab30
commit
1027e80d53
@ -186,7 +186,7 @@ void MicTXView::rxaudio(bool is_on) {
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baseband::run_image(portapack::spi_flash::image_tag_mic_tx);
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audio::output::stop();
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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)
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audio::input::start(); // set up audio input = mic config of any audio coded AK4951/WM8731, (in WM8731 parameter will be ignored)
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portapack::pin_i2s0_rx_sda.mode(3);
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configure_baseband();
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}
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@ -212,7 +212,7 @@ MicTXView::MicTXView(
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baseband::run_image(portapack::spi_flash::image_tag_mic_tx);
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if ( audio_codec_wm8731.detected() ) {
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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() ) {
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add_children({
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&labels_WM8731, // we have audio codec WM8731, same MIC menu as original.
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&vumeter,
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@ -285,7 +285,7 @@ MicTXView::MicTXView(
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options_ak4951_alc_mode.on_change = [this](size_t, int8_t v) {
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ak4951_alc_GUI_selected = v;
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audio::input::start(ak4951_alc_GUI_selected);
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audio::input::start();
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};
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// options_ak4951_alc_mode.set_selected_index(0);
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@ -341,23 +341,37 @@ MicTXView::MicTXView(
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enable_dsb = false;
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field_bw.set_value(transmitter_model.channel_bandwidth() / 1000);
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//if (rx_enabled)
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rxaudio(rx_enabled); //Update now if we have RX audio on
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rxaudio(rx_enabled); //Update now if we have RX audio on
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options_tone_key.hidden(0); // we are in FM mode , we should have active the Key-tones & CTCSS option.
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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)
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break;
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case 1:
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enable_am = true;
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rxaudio(rx_enabled); //Update now if we have RX audio on
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rxaudio(rx_enabled); //Update now if we have RX audio on
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options_tone_key.set_selected_index(0); // we are NOT in FM mode , we reset the possible previous key-tones &CTCSS selection.
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set_dirty(); // Refresh display
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options_tone_key.hidden(1); // we hide that Key-tones & CTCSS input selecction, (no meaning in AM/DSB/SSB).
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field_rxbw.hidden(1); // we hide the NFM BW selection in other modes non_FM (no meaning in AM/DSB/SSB)
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check_rogerbeep.hidden(0); // make visible again the "rogerbeep" selection.
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break;
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case 2:
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enable_usb = true;
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rxaudio(rx_enabled); //Update now if we have RX audio on
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rxaudio(rx_enabled); //Update now if we have RX audio on
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check_rogerbeep.set_value(false); // reset the possible activation of roger beep, because it is not compatible with SSB , by now.
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check_rogerbeep.hidden(1); // hide that roger beep selection.
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set_dirty(); // Refresh display
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break;
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case 3:
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enable_lsb = true;
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rxaudio(rx_enabled); //Update now if we have RX audio on
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rxaudio(rx_enabled); //Update now if we have RX audio on
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check_rogerbeep.set_value(false); // reset the possible activation of roger beep, because it is not compatible with SSB , by now.
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check_rogerbeep.hidden(1); // hide that roger beep selection.
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set_dirty(); // Refresh display
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break;
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case 4:
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enable_dsb = true;
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rxaudio(rx_enabled); //Update now if we have RX audio on
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rxaudio(rx_enabled); //Update now if we have RX audio on
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check_rogerbeep.hidden(0); // make visible again the "rogerbeep" selection.
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break;
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}
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//configure_baseband();
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@ -525,7 +539,7 @@ MicTXView::MicTXView(
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set_tx(false);
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audio::set_rate(audio::Rate::Hz_24000);
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audio::input::start(ak4951_alc_GUI_selected); // originally , audio::input::start(); (we added parameter)
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audio::input::start(); // originally , audio::input::start(); (we added parameter)
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}
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MicTXView::~MicTXView() {
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@ -168,8 +168,8 @@ void speaker_mute() {
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namespace input {
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void start(int8_t alc_mode) {
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audio_codec->microphone_enable(alc_mode); // added user-GUI selection for AK4951, ALC mode parameter.
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void start() {
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audio_codec->microphone_enable();
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i2s::i2s0::rx_start();
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}
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@ -49,7 +49,7 @@ public:
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virtual volume_range_t headphone_gain_range() const = 0;
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virtual void set_headphone_volume(const volume_t volume) = 0;
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virtual void microphone_enable(int8_t alc_mode) = 0; // added user-GUI AK4951 ,selected ALC mode.
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virtual void microphone_enable() = 0;
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virtual void microphone_disable() = 0;
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virtual size_t reg_count() const = 0;
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@ -59,7 +59,7 @@ public:
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namespace output {
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void start(); // this other start(),no changed. ,in namespace output , used to config audio playback mode,
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void start();
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void stop();
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void mute();
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@ -72,7 +72,7 @@ void speaker_unmute();
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namespace input {
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void start(int8_t alc_mode); // added parameter user-GUI select AK4951-ALC mode for config mic path,(recording mode in datasheet),
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void start();
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void stop();
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} /* namespace input */
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@ -21,6 +21,8 @@
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#include "dsp_modulate.hpp"
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#include "sine_table_int8.hpp"
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#include "portapack_shared_memory.hpp"
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#include "tonesets.hpp"
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namespace dsp {
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namespace modulate {
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@ -40,13 +42,42 @@ void Modulator::set_over(uint32_t new_over) {
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over = new_over;
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}
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void Modulator::set_gain_vumeter_beep(float new_audio_gain , bool new_play_beep ) {
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audio_gain = new_audio_gain ;
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play_beep = new_play_beep;
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}
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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 ) {
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if (play_beep) { // We need to add audio beep sample.
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if (new_beep_timer) {
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new_beep_timer--;
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} else {
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new_beep_timer = baseband_fs * 0.05; // 50ms
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if (new_beep_index == BEEP_TONES_NB) {
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configured_in = false;
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shared_memory.application_queue.push(new_txprogress_message);
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} else {
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beep_gen.configure(beep_deltas[new_beep_index], 1.0); // config sequentially the audio beep tone.
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new_beep_index++;
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}
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}
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sample_in = beep_gen.process(0); // Get sample of the selected sequence of 6 beep tones , and overwrite audio sample. Mix 0%.
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}
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return sample_in; // Return audio mic scaled with gain , 8 bit sample or audio beep sample.
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}
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///
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SSB::SSB() : hilbert() {
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mode = Mode::LSB;
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}
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void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
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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 ) {
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// 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 .
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int32_t sample = 0;
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int8_t re = 0, im = 0;
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@ -55,7 +86,9 @@ void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
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float i = 0.0, q = 0.0;
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sample = audio.p[counter / over] >> 2;
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//switch (mode) {
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sample *= audio_gain; // Apply GAIN Scale factor to the audio TX modulation.
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//switch (mode) {
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//case Mode::LSB:
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hilbert.execute(sample / 32768.0f, i, q);
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//case Mode::USB: hilbert.execute(sample / 32768.0f, q, i);
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@ -72,9 +105,23 @@ void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
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//re = q;
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//im = i;
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//break;
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}
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buffer.p[counter] = { re, im };
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// Update vu-meter bar in the LCD screen.
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power_acc += (sample < 0) ? -sample : sample; // Power average for UI vu-meter
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if (new_power_acc_count) {
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new_power_acc_count--;
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} else { // power_acc_count = 0
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new_power_acc_count = new_divider;
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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 .
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shared_memory.application_queue.push(new_level_message);
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power_acc = 0;
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}
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}
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}
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@ -88,15 +135,39 @@ void FM::set_fm_delta(uint32_t new_delta) {
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fm_delta = new_delta;
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}
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void FM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
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void FM::set_tone_gen_configure(const uint32_t set_delta, const float set_tone_mix_weight) {
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tone_gen.configure(set_delta, set_tone_mix_weight);
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}
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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 ) {
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int32_t sample = 0;
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int8_t re, im;
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for (size_t counter = 0; counter < buffer.count; counter++) {
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if (counter % over == 0) {
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sample = audio.p[counter / over] >> 8;
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delta = sample * fm_delta;
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}
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sample = audio.p[counter>>6] >> 8; // sample = audio.p[counter / over] >> 8; (not enough efficient running code, over = 1536000/240000= 64 )
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sample *= audio_gain; // Apply GAIN Scale factor to the audio TX modulation.
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if (play_beep) {
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sample = apply_beep(sample, configured_in, new_beep_index, new_beep_timer, new_txprogress_message ); // Apply beep -if selected - atom ,sample by sample.
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} else {
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// Update vu-meter bar in the LCD screen.
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power_acc += (sample < 0) ? -sample : sample; // Power average for UI vu-meter
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if (new_power_acc_count) {
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new_power_acc_count--;
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} else { // power_acc_count = 0
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new_power_acc_count = new_divider;
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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 .
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shared_memory.application_queue.push(new_level_message);
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power_acc = 0;
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}
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// TODO: pending to optimize CPU running code.
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// So far , we can not handle all 3 issues at the same time (vu-meter , CTCSS, beep).
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sample = tone_gen.process(sample); // Add selected Key_Tone or CTCSS subtone , atom function() , sample by sample.
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}
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delta = sample * fm_delta; // Modulate FM
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phase += delta;
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sphase = phase >> 24;
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@ -112,16 +183,33 @@ AM::AM() {
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mode = Mode::AM;
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}
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void AM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
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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 ) {
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int32_t sample = 0;
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int8_t re = 0, im = 0;
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float q = 0.0;
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float q = 0.0;
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for (size_t counter = 0; counter < buffer.count; counter++) {
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if (counter % 128 == 0) {
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sample = audio.p[counter / over] >> 2;
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sample *= audio_gain; // Apply GAIN Scale factor to the audio TX modulation.
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}
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if (play_beep) {
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sample = apply_beep(sample, configured_in, new_beep_index, new_beep_timer, new_txprogress_message )<<5; // Apply beep -if selected - atom sample by sample.
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} else {
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// Update vu-meter bar in the LCD screen.
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power_acc += (sample < 0) ? -sample : sample; // Power average for UI vu-meter
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if (new_power_acc_count) {
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new_power_acc_count--;
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} else { // power_acc_count = 0
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new_power_acc_count = new_divider;
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new_level_message.value = power_acc / (new_divider *8); // Why ?orig / (new_divider / 4); // Why ?
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shared_memory.application_queue.push(new_level_message);
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power_acc = 0;
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}
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}
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q = sample / 32768.0f;
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q *= 256.0f; // Original 64.0f,now x4 (+12 dB's BB_modulation in AM & DSB)
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switch (mode) {
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@ -24,6 +24,8 @@
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#include "dsp_types.hpp"
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#include "dsp_hilbert.hpp"
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#include "tone_gen.hpp"
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#include "baseband_processor.hpp"
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namespace dsp {
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namespace modulate {
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@ -41,13 +43,28 @@ enum class Mode {
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class Modulator {
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public:
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virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) = 0;
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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;
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virtual ~Modulator();
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Mode get_mode();
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void set_mode(Mode new_mode);
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void set_over(uint32_t new_over);
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void set_gain_vumeter_beep(float new_audio_gain , bool new_play_beep );
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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 );
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float audio_gain { };
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bool play_beep { false };
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uint32_t power_acc_count { 0 }; // this var it is initialized from Proc_mictx.cpp
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uint32_t divider { }; // this var it is initialized from Proc_mictx.cpp
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uint64_t power_acc { 0 }; // it is aux Accumulated sum (Absolute sample signal) , initialitzed to zero.
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AudioLevelReportMessage level_message { };
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private:
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static constexpr size_t baseband_fs = 1536000U;
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TXProgressMessage txprogress_message { };
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ToneGen beep_gen { };
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uint32_t beep_index { }, beep_timer { };
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protected:
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uint32_t over = 1;
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@ -60,7 +77,7 @@ class SSB : public Modulator {
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public:
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SSB();
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virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer);
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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 );
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private:
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dsp::HilbertTransform hilbert;
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@ -72,8 +89,9 @@ class FM : public Modulator {
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public:
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FM();
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virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer);
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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 ) ;
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void set_fm_delta(uint32_t new_delta);
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void set_tone_gen_configure(const uint32_t delta, const float tone_mix_weight);
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///
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@ -81,13 +99,16 @@ private:
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uint32_t fm_delta { 0 };
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uint32_t phase { 0 }, sphase { 0 };
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int32_t sample { 0 }, delta { };
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ToneGen tone_gen { };
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};
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class AM : public Modulator {
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public:
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AM();
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virtual void execute(const buffer_s16_t& audio, const buffer_c8_t& buffer);
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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 );
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};
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} /* namespace modulate */
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@ -35,8 +35,10 @@ void MicTXProcessor::execute(const buffer_c8_t& buffer){
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if (!configured) return;
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audio_input.read_audio_buffer(audio_buffer);
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modulator->execute(audio_buffer, buffer);
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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;
|
||||
|
||||
|
||||
@ -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 };
|
||||
|
||||
@ -103,27 +103,13 @@ void SpectrumCollector::post_message(const buffer_c16_t& data) {
|
||||
}
|
||||
}
|
||||
|
||||
/* 3 types of Windowing time domain shapes declaration , but only used Hamming , shapes for FFT
|
||||
GCC10 compile sintax error c/m (1/2),
|
||||
The primary diff. between const and constexpr variables is that
|
||||
the initialization of a const var can be deferred until run time.
|
||||
A constexpr var. must be initialized at compile time. ...
|
||||
A var. can be declared with constexpr , when it has a literal type and is initialized.
|
||||
GCC compile sintax error c/m (2/2)
|
||||
Static assert --> Tests a software assertion at compile time for debugging.
|
||||
we keep the same safety compile protection , just changing slightly the sintax checking that the size of the called array is power of 2.
|
||||
if the bool "constant expression" is TRUE (normal case) , the declaration has no effect.
|
||||
if the bool "constant expression" is FALSE (abnormal array size) , it is aborted the compile with a msg error.
|
||||
*/
|
||||
|
||||
|
||||
template<typename T> // Although currently we are not using that Windowing shape, we apply the same GCC10 compile error c/m
|
||||
template<typename T>
|
||||
static typename T::value_type spectrum_window_none(const T& s, const size_t i) {
|
||||
static_assert(power_of_two(ARRAY_ELEMENTS(s)), "Array number of elements must be power of 2"); // c/m compile error GCC10 , OK for all GCC versions.
|
||||
return s[i];
|
||||
};
|
||||
|
||||
template<typename T> // Currently we are calling and using that Window shape.
|
||||
template<typename T>
|
||||
static typename T::value_type spectrum_window_hamming_3(const T& s, const size_t i) {
|
||||
static_assert(power_of_two(ARRAY_ELEMENTS(s)), "Array number of elements must be power of 2"); // c/m compile error GCC10 , OK for all GCC versions.
|
||||
const size_t mask = s.size() - 1; // c/m compile error GCC10 , constexpr->const
|
||||
@ -131,7 +117,7 @@ static typename T::value_type spectrum_window_hamming_3(const T& s, const size_t
|
||||
return s[i] * 0.54f + (s[(i-1) & mask] + s[(i+1) & mask]) * -0.23f;
|
||||
};
|
||||
|
||||
template<typename T> // Although currently we are not using that Windowing shape, we apply the same GCC10 compile error c/m
|
||||
template<typename T>
|
||||
static typename T::value_type spectrum_window_blackman_3(const T& s, const size_t i) {
|
||||
static_assert(power_of_two(ARRAY_ELEMENTS(s)), "Array number of elements must be power of 2"); // c/m compile error GCC10 , OK for all GCC versions.
|
||||
const size_t mask = s.size() - 1; // c/m compile error GCC10 , constexpr->const
|
||||
|
||||
@ -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_);
|
||||
}
|
||||
|
||||
@ -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__ */
|
||||
|
||||
@ -216,347 +216,90 @@ void AK4951::speaker_disable() {
|
||||
set_speaker_power(false);
|
||||
}
|
||||
|
||||
void AK4951::microphone_enable(int8_t alc_mode) {
|
||||
// alc_mode =0 = (OFF =same as original code = NOT using AK4951 Programmable digital filter block),
|
||||
// alc_mode >1 (with DIGITAL FILTER BLOCK , example : 1:(+12dB) , 2:(+9dB)", 3:(+6dB), ...)
|
||||
|
||||
// map.r.digital_mic.DMIC = 0; // originally commented code
|
||||
// update(Register::DigitalMic); // originally commented code
|
||||
|
||||
uint_fast8_t mgain =0b0111; // Pre-amp mic (Original code, =0b0111 (+21dB's=7x3dBs),(Max is NOT 0b1111!, it is 0b1010=+30dBs=10x3dBs)
|
||||
|
||||
map.r.signal_select_2.INL = 0b01; // Lch input signal = LIN2 , our ext. MONO MIC is connected here LIN2 in Portapack.
|
||||
map.r.signal_select_2.INR = 0b01; // Rch input signal = RIN2 , Not used ,not connected ,but no problem.
|
||||
map.r.signal_select_2.MICL = 0; // MPWR = 2.4V (it has two possible settings , 2.4V or 2.0V) , (majority smarthphones around 2V , range 1V-5V)
|
||||
update(Register::SignalSelect2);
|
||||
|
||||
// ------Common code part, = original setting conditions, it is fine for all user-GUI alc_modes: OFF , and ALC modes .*/
|
||||
map.r.digital_filter_select_1.HPFAD = 1; // HPF1 ON (after ADC);page 40 datasheet, HPFAD bit controls the ON/OFF of the HPF1 (HPF ON is recommended).
|
||||
map.r.digital_filter_select_1.HPFC = 0b11; // HPF Cut off frequency of high pass filter from 236.8 Hz @fs=48k ("00":3.7Hz, "01":14,8Hz, "10":118,4Hz)
|
||||
update(Register::DigitalFilterSelect1);
|
||||
|
||||
// map.r.r_ch_mic_gain_setting.MGR = 0x80; // Microphone sensitivity correction = 0dB., (not used by now , original code cond.)
|
||||
// update(Register::RchMicGainSetting); // (those two lines , not activated, same as original)
|
||||
|
||||
// pre-load 4 byes LPF coefficicients (.lpf_coefficient_0,1,2,3), FSA 14..0, FSB 14..0 , (fcut initial 6kHz, fs 48Khz).
|
||||
// it will be default pre-loading coeff. for al ALC modes, LPF bit is activated down, for all ALC digital modes.
|
||||
map.r.lpf_coefficient_0.l = 0x5F; // Pre-loading here LPF 6kHz, 1st Order from digital Block , Fc=6000 Hz, fs = 48khz
|
||||
map.r.lpf_coefficient_1.h = 0x09; // LPF bit is activated down, for all ALC digital modes.
|
||||
map.r.lpf_coefficient_2.l = 0xBF; // Writting reg to AK4951, with "update", following instructions.
|
||||
map.r.lpf_coefficient_3.h = 0x32;
|
||||
|
||||
update(Register::LPFCoefficient0); // Writing pre-loaded 4 bytes LPF CoefFiecients 14 bits (FSA13..0, FSB13..0
|
||||
update(Register::LPFCoefficient1); // In this case , LPF 6KHz , when we activate the LPF block.
|
||||
update(Register::LPFCoefficient2);
|
||||
update(Register::LPFCoefficient3);
|
||||
|
||||
// Reset , setting OFF all 5 x Digital Equalizer filters
|
||||
map.r.digital_filter_select_3.EQ1 = 0; // EQ1 Coeffic Setting , (0: Disable-default, audio data passes EQ1 block by 0dB gain). When EQ1="1”, the settings of E1A15-0, E1B15-0 and E1C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ2 = 0; // EQ2 Coeffic Setting , (0: Disable-default, audio data passes EQ2 block by 0dB gain). When EQ2="1”, the settings of E2A15-0, E2B15-0 and E2C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ3 = 0; // EQ3 Coeffic Setting , (0: Disable-default, audio data passes EQ3 block by 0dB gain). When EQ3="1”, the settings of E3A15-0, E3B15-0 and E3C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ4 = 0; // EQ4 Coeffic Setting , (0: Disable-default, audio data passes EQ4 block by 0dB gain). When EQ4="1”, the settings of E4A15-0, E4B15-0 and E4C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ5 = 0; // EQ5 Coeffic Setting , (0: Disable-default, audio data passes EQ5 block by 0dB gain). When EQ5="1”, the settings of E5A15-0, E5B15-0 and E5C15-0 bits are enabled
|
||||
update(Register::DigitalFilterSelect3); // A,B,C EQ1 Coefficients are already pre-loaded in ak4951.hpp
|
||||
|
||||
|
||||
if (alc_mode==0) { // Programmable Digital Filter OFF, same as original condition., no Digital ALC, nor Wind Noise Filter, LPF , EQ
|
||||
|
||||
map.r.digital_filter_select_2.LPF = 0; // LPF-Block, Coeffic Setting Enable (OFF-Default), When LPF bit is “0”, audio data passes the LPF block by 0dB gain.
|
||||
update(Register::DigitalFilterSelect2);
|
||||
|
||||
// Pre-loading AUDIO PATH with all DIGITAL BLOCK by pased, see, audio path block diagramm AK4951 datasheet + Table Playback mode -Recording mode.
|
||||
// Digital filter block PATH is BY PASSED (we can swith off DIG. BLOCK power , PMPFIL=0) .The Path in Recording Mode 2 & Playback Mode 2 (NO DIG FILTER BLOCK AT ALL, not for MIC recording, nor for Playback)
|
||||
map.r.digital_filter_mode.ADCPF = 1; // ADCPF bit swith ("0" Mic after ADC Output connected (recording mode) to the DIGITAL FILTER BLOCK. ("1" Playback mode)
|
||||
map.r.digital_filter_mode.PFSDO = 0; // ADC bit switch ("0" : 1st order HPF) connectedto the Output. By bass DIGITAL block .
|
||||
map.r.digital_filter_mode.PFDAC = 0b00; // (Input selector for DAC (not used in MIC), SDTI= Audio Serial Data Input Pin)
|
||||
update(Register::DigitalFilterMode); // Writing the Audio Path : NO DIGITAL BLOCK or DIG BLOCK FOR MIC , Audio mode path : Playback mode /-Recording mode.
|
||||
|
||||
map.r.power_management_1.PMADL = 1; // ADC Lch = Lch input signal. Mic Amp Lch and ADC Lch Power Management
|
||||
map.r.power_management_1.PMADR = 1; // ADC Rch = Rch input signal. Mic Amp Rch and ADC Rch Power Management
|
||||
map.r.power_management_1.PMPFIL = 0; // Pre-loading , Programmable Dig. filter OFF ,filter unused, routed around.(original value = 0 )
|
||||
update(Register::PowerManagement1); // Activating the Power management of the used blocks . (Mic ADC always + Dig Block filter , when used )
|
||||
|
||||
// 1059/fs, 22ms @ 48kHz
|
||||
chThdSleepMilliseconds(22);
|
||||
|
||||
} else { // ( alc_mode !=0)
|
||||
|
||||
switch(alc_mode) { // Pre-loading register values depending on user-GUI selection (they will be sended below, with "update(Register_name::xxx )".
|
||||
|
||||
case 1: // ALC-> on, (+12dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0xC0; // REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, C0H=+12dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xC0; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xC0; // Right Input Dig Vol Setting, same comment as above , The value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 2: // ALC-> on, (+09dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0xB8; // REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, B8H= +9dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xB8; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xB8; // Right Input Dig Vol Setting, same comment as above , The value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 3: // ALC-> on, (+06dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0xB0; // 0xB8 , REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, B0H= +6dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xB0; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xB0; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 4: // ALC-> on, (+03dB's) Auto Vol max + Wind Noise cancel + Pre-amp Mic (+21dB=original)
|
||||
// + EQ boosting ~<2kHz (f0:1,1k, fb:1,7K, k=1,8) && + LPF 3,5k
|
||||
map.r.alc_mode_control_2.REF = 0xA8; // 0xA8 , REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A8H= +3dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xA8; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xA8; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
//The EQn (n=1, 2, 3, 4 or 5) coefficient must be set when EQn bit = “0” or PMPFIL bit = “0”.
|
||||
map.r.digital_filter_select_3.EQ1 = 1; // EQ1 Coeffic Setting , (0: Disable-default, audio data passes EQ1 block by 0dB gain). When EQ1="1”, the settings of E1A15-0, E1B15-0 and E1C15-0 bits are enabled
|
||||
update(Register::DigitalFilterSelect3); // A,B,C EQ1 Coefficients are already pre-loaded in ak4951.hpp
|
||||
|
||||
map.r.lpf_coefficient_0.l = 0x0D; // Pre-loading here LPF 3,5k , 1st Order from digital Block , Fc=3.500 Hz, fs = 48khz
|
||||
map.r.lpf_coefficient_1.h = 0x06; // LPF bit is activated down, for all ALC digital modes.
|
||||
map.r.lpf_coefficient_2.l = 0x1A; // Writting reg to AK4951 , down with update....
|
||||
map.r.lpf_coefficient_3.h = 0x2C;
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 5: // ALC-> on, (+03dB's) Auto Vol max + Wind Noise cancel + Pre-amp Mic (+21dB=original)
|
||||
// + EQ boosting ~<3kHz (f0~1k4,fb~2,4k,k=1,8) && LPF 4kHz
|
||||
map.r.alc_mode_control_2.REF = 0xA8; // 0xA0 , REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A8H= +3dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xA8; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xA8; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
map.r.digital_filter_select_3.EQ2 = 1; // EQ2 Coeffic Setting , (0: Disable-default, audio data passes EQ2 block by 0dB gain). When EQ2="1”, the settings of E2A15-0, E2B15-0 and E2C15-0 bits are enabled
|
||||
update(Register::DigitalFilterSelect3);
|
||||
|
||||
map.r.lpf_coefficient_0.l = 0xC3; // Pre-loading here LPF 4k , 1st Order from digital Block , Fc=4000 Hz, fs = 48khz
|
||||
map.r.lpf_coefficient_1.h = 0x06; // LPF bit is activated down, for all ALC digital modes.
|
||||
map.r.lpf_coefficient_2.l = 0x86; // Writting reg to AK4951 , down with update....
|
||||
map.r.lpf_coefficient_3.h = 0x2D;
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 6: // ALC-> on, (+03dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0xA8; // REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A0H= 0dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xA8; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xA8; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 7: // ALC-> on, (+00dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0xA0; // REF7-0 bits,max gain at ALC recoveryoperation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, A0H= 0dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0xA0; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0xA0; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 8: // ALC-> on, (-03dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0x98; //REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 98H=-03dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0x98; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0x98; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
void AK4951::microphone_enable() {
|
||||
// map.r.digital_mic.DMIC = 0;
|
||||
// update(Register::DigitalMic);
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
const uint_fast8_t mgain = 0b0111;
|
||||
map.r.signal_select_1.MGAIN20 = mgain & 7;
|
||||
map.r.signal_select_1.PMMP = 1;
|
||||
map.r.signal_select_1.MPSEL = 1; // MPWR2 pin
|
||||
map.r.signal_select_1.MGAIN3 = (mgain >> 3) & 1;
|
||||
update(Register::SignalSelect1);
|
||||
|
||||
case 9: // ALC-> on, (-06dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz + Pre-amp Mic (+21dB=original)
|
||||
map.r.alc_mode_control_2.REF = 0x90; // REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 90H=-06dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0x90; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0x90; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
map.r.signal_select_2.INL = 0b01; // Lch input signal = LIN2
|
||||
map.r.signal_select_2.INR = 0b01; // Rch input signal = RIN2
|
||||
map.r.signal_select_2.MICL = 0; // MPWR = 2.4V
|
||||
update(Register::SignalSelect2);
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 10: // ALC-> on, (-09dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz - Pre-amp MIC -3dB (18dB's)
|
||||
// Reduce also Pre-amp Mic -3dB's (+18dB's)
|
||||
mgain = 0b0110; // Pre-amp mic Mic Gain Pre-amp (+18dB), Original=0b0111 (+21dB's =7x3dBs),
|
||||
|
||||
map.r.alc_mode_control_2.REF = 0x88; // REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 88H=-09dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0x88; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0x88; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
|
||||
case 11: // ALC-> on, (-12dB's) Auto Vol max + Wind Noise cancel + LPF 6kHz - Pre-amp MIC -6dB (15dB's)
|
||||
// Reduce also Pre-amp Mic -6dB's (+15dB's)
|
||||
mgain = 0b0101; // Pre-amp mic Mic Gain Pre-amp (+15dB), (Original=0b0111 (+21dB's= 7x3dBs),
|
||||
|
||||
map.r.alc_mode_control_2.REF = 0x80; // REF7-0 bits,max gain at ALC recovery operation,(FFH +36dBs , D0H +18dBs, A0H 0dBs, 80H=-12dBs)
|
||||
map.r.l_ch_input_volume_control.IV = 0x80; // Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.r_ch_input_volume_control.IV = 0x80; // Right Input Dig Vol Setting, same comment as above , Then value of IVOL should be <= than REF’s
|
||||
|
||||
// Already Pre-loaded, "map.r.lpf_coefficient", 6Khz - LPF 1st Order from digital Block,Fc=6000Hz,fs = 48khz
|
||||
// LPF bit is activated down, for all ALC digital modes.
|
||||
break;
|
||||
}
|
||||
|
||||
//-------------------------------DIGITAL ALC (Automatic Level Control ) --- --------
|
||||
map.r.alc_mode_control_1.ALC = 0; // LMTH2-0, WTM1-0, RGAIN2-0, REF7-0, RFST1-0, EQFC1-0, FRATT, FRN and ALCEQN bits (needs to be set up with ALC disable = 0)
|
||||
update(Register::ALCModeControl1);
|
||||
|
||||
map.r.timer_select.FRN = 0; // (FRN= 0 Fast Recovery mode , enable )
|
||||
map.r.timer_select.FRATT = 0; // Fast Recovery Ref. Volume Atten. Amount -0,00106dB's, timing 4/fs (default)
|
||||
map.r.timer_select.ADRST = 0b00; // initial offset ADC cycles , 22ms @fs=48Khz.
|
||||
// map.r.r_ch_mic_gain_setting.MGR = 0x80; // Microphone sensitivity correction = 0dB.
|
||||
// update(Register::RchMicGainSetting);
|
||||
/*
|
||||
map.r.timer_select.FRN = ?;
|
||||
map.r.timer_select.FRATT = ?;
|
||||
map.r.timer_select.ADRST = 0b??;
|
||||
update(Register::TimerSelect);
|
||||
|
||||
map.r.alc_timer_select.RFST = 0b00; // RFST1-0: ALC Fast Recovery Speed Default: “00” (0.0032dB)
|
||||
map.r.alc_timer_select.WTM = 0b00; // ALC Recovery Operation Waiting Period 128/fs = 2,7 mseg (min=default)
|
||||
map.r.alc_timer_select.EQFC = 0b10; // Selecting default, fs 48Khz , ALCEQ: First order zero pole high pass filter fc2=100Hz, fc1=150Hz
|
||||
map.r.alc_timer_select.IVTM = 0; // IVTM bit set the vol transition time ,236/fs = 4,9msecs (min) (default was 19,7msegs.)
|
||||
map.r.alc_timer_select. = ?;
|
||||
update(Register::ALCTimerSelect);
|
||||
|
||||
map.r.alc_mode_control_1.LMTH10 = 0b11; // ALC Limiter Detec Level/ Recovery Counter Reset; lower 2 bits (Ob111=-8,4dbs), (default 0b000=-2,5dBs)
|
||||
map.r.alc_mode_control_1.RGAIN = 0b000; // ALC Recovery Gain Step, max step , max speed. Default: “000” (0.00424dB)
|
||||
map.r.alc_mode_control_1.ALC = 1; // ALC Enable . (we are now, NOT in MANUAL volume mode, only becomes manual when (ALC=“0” while ADCPF=“1”. )
|
||||
map.r.alc_mode_control_1.LMTH2 = 1; // ALC Limiter Detection Level/ Recovery Counter Reset Level,Upper bit,default 0b000
|
||||
map.r.alc_mode_control_1.ALCEQN = 1; // ALC EQ Off =1 not used by now, 0: ALC EQ On (default)
|
||||
map.r.alc_mode_control_1. = ?;
|
||||
map.r.alc_mode_control_1.ALC = 1;
|
||||
update(Register::ALCModeControl1);
|
||||
|
||||
// map.r.alc_mode_control_2.REF = 0x??; // Pre-loaded in top part. Maximum gain at ALC recovery operation,.(FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
map.r.alc_mode_control_2.REF = ?;
|
||||
update(Register::ALCModeControl2);
|
||||
|
||||
// map.r.l_ch_input_volume_control.IV = 0x??; // Pre-loaded in top part. Left, Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
update(Register::LchInputVolumeControl);
|
||||
|
||||
// map.r.r_ch_input_volume_control.IV = 0x??; // Pre-loaded in top part. Right,Input Digital Volume Setting, (FFH +36dBs , D0H +18dBs, A0H 0dBs, 70H=-18dBs)
|
||||
update(Register::RchInputVolumeControl);
|
||||
|
||||
|
||||
//---------------Switch ON, Digital Automatic Wind Noise Filter reduction -------------------
|
||||
// Difficult to realise that Dynamic HPF Wind noise filter benefit, maybe because we have another fixed HPF 236.8 Hz .
|
||||
// Anyway , we propose to activate it , with default setting conditions.
|
||||
map.r.power_management_1.PMPFIL = 0; // (*1) To programm SENC, STG , we need PMPFIL = 0 . (but this disconnect Digital block power supply.
|
||||
update(Register::PowerManagement1); // Updated PMPFIL to 0 . (*1)
|
||||
|
||||
map.r.auto_hpf_control.STG = 0b00; // (00=LOW ATTENUATION Level), lets put 11 (HIGH ATTENUATION Level) (default 00)
|
||||
map.r.auto_hpf_control.SENC = 0b011; // (000=LOW sensitivity detection)… 111((MAX sensitivity detection) (default 011)
|
||||
map.r.auto_hpf_control.AHPF = 1; // Autom. Wind noise filter ON (AHPF bit=“1”).It atten. wind noise when detecting ,and adjusts the atten. level dynamically.
|
||||
update(Register::AutoHPFControl);
|
||||
|
||||
// We are in Digital Block ON , (Wind Noise Filter+ALC+LPF+EQ),==> needs at the end , PMPFIL=1 , Program. Dig.filter ON
|
||||
// map.r.power_management_1.PMPFIL = 1; // that instruction is at the end , we can skp pre-loading Programmable Dig. filter ON (*1)
|
||||
//---------------------------------------------------------------------
|
||||
|
||||
// Writing AUDIO PATH diagramm, Changing Audio mode path : Playback mode1 /-Recording mode2. (Figure 37 AK4951 datasheet, Table 27. Recording Playback Mode)
|
||||
// When changing those modes, PMPFIL bit must be “0”, it is OK (*1)
|
||||
map.r.digital_filter_mode.ADCPF = 1; // ADCPF bit swith ("0" Mic after ADC Output connected (recording mode) to the DIGITAL FILTER BLOCK. ("1" Playback mode)
|
||||
map.r.digital_filter_mode.PFSDO = 1; // ADC (+ 1st order HPF) Output
|
||||
map.r.digital_filter_mode.PFDAC = 0b00; // (Input selector for DAC (not used in MIC), SDTI= Audio Serial Data Input Pin)
|
||||
update(Register::DigitalFilterMode); // Writing the Audio Path : NO DIGITAL BLOCK or DIG BLOCK FOR MIC , Audio mode path : Playback mode /-Recording mode.
|
||||
|
||||
// The EQn (n=1, 2, 3, 4 or 5) coefficient must be set when EQn bit = “0” or PMPFIL bit = “0”., but we are already (*1)
|
||||
// map.r.power_management_1.PMPFIL = 0; // In the previous Wind Noise Filter , we already set up PPFIL = 0
|
||||
// update(Register::PowerManagement1); // Activating the Power management of the used blocks . (Mic ADC always + Dig Block filter , when used )
|
||||
|
||||
// ... Set EQ & LPF coefficients ---------------------------------
|
||||
|
||||
// writting to the IC ak4951 reg. settings defined in Ak4951.hpp , the 30 bytes , EQ coefficient = 5 (EQ1,2,3,4,5) x 3 (A,B,C coefficients) x 2 bytes (16 bits)
|
||||
update(Register::E1Coefficient0); // we could pre-load here,ex ,"map.r.e1_coefficient_0.l = 0x50;" , EQ1 Coefficient A : A7...A0, but already done in ak4951.hpp
|
||||
update(Register::E1Coefficient1); // we could pre-load here,ex ,"map.r.e1_coefficient_1.h = 0xFE;" , EQ1 Coefficient A : A15..A8, " "
|
||||
update(Register::E1Coefficient2); // we could pre-load here,ex ,"map.r.e1_coefficient_2.l = 0x29;" , EQ1 Coefficient B : B7...B0, " "
|
||||
update(Register::E1Coefficient3); // we could pre-load here,ex ,"map.r.e1_coefficient_3.h = 0xC5;" , EQ1 Coefficient B : B15..B8, " "
|
||||
update(Register::E1Coefficient4); // we could pre-load here,ex ,"map.r.e1_coefficient_4.l = 0xA0;" , EQ1 Coefficient C : C7...C0, " "
|
||||
update(Register::E1Coefficient5); // we could pre-load here,ex ,"map.r.e1_coefficient_5.h = 0x1C;" , EQ1 Coefficient C : C15..C8, " "
|
||||
|
||||
update(Register::E2Coefficient0); // writing pre-loaded EQ2 coefficcients
|
||||
update(Register::E2Coefficient1);
|
||||
update(Register::E2Coefficient2);
|
||||
update(Register::E2Coefficient3);
|
||||
update(Register::E2Coefficient4);
|
||||
update(Register::E2Coefficient5);
|
||||
|
||||
// Already pre-loaded LPF coefficients to 6k, 3,5k or 4k ,(LPF 6Khz all digital alc modes top , except when 3k5 , 4k)
|
||||
update(Register::LPFCoefficient0); // Writing pre-loaded 4 bytes LPF CoefFiecients 14 bits (FSA13..0, FSB13..0
|
||||
update(Register::LPFCoefficient1);
|
||||
update(Register::LPFCoefficient2);
|
||||
update(Register::LPFCoefficient3);
|
||||
|
||||
// Activating LPF block , (and re-configuring the rest of bits of the same register)
|
||||
map.r.digital_filter_select_2.HPF = 0; // HPF2-Block, Coeffic Setting Enable (OFF-Default), When HPF bit is “0”, audio data passes the HPF2 block by is 0dB gain.
|
||||
map.r.digital_filter_select_2.LPF = 1; // LPF-Block, Coeffic Setting Enable (OFF-Default), When LPF bit is “0”, audio data passes the LPF block by 0dB gain.
|
||||
map.r.digital_filter_select_2.FIL3 = 0; // Stereo_Emphasis_Filter-Block,(OFF-Default) Coefficient Setting Enable , OFF , Disable.
|
||||
map.r.digital_filter_select_2.EQ0 = 0; // Gain Compensation-Block, (OFF-Default) Coeffic Setting Enable, When EQ0 bit = “0” audio data passes the EQ0 block by 0dB gain.
|
||||
map.r.digital_filter_select_2.GN = 0b00; // Gain Setting of the Gain Compensation Block Default: “00”-Default (0dB)
|
||||
*/
|
||||
// map.r.l_ch_input_volume_control.IV = 0xe1;
|
||||
// update(Register::LchInputVolumeControl);
|
||||
// map.r.r_ch_input_volume_control.IV = 0xe1;
|
||||
// update(Register::RchInputVolumeControl);
|
||||
/*
|
||||
map.r.auto_hpf_control.STG = 0b00;
|
||||
map.r.auto_hpf_control.SENC = 0b011;
|
||||
map.r.auto_hpf_control.AHPF = 0;
|
||||
update(Register::AutoHPFControl);
|
||||
*/
|
||||
map.r.digital_filter_select_1.HPFAD = 1; // HPF1 (after ADC) = on
|
||||
map.r.digital_filter_select_1.HPFC = 0b11; // 2336.8 Hz @ fs=48k
|
||||
update(Register::DigitalFilterSelect1);
|
||||
/*
|
||||
map.r.digital_filter_select_2.HPF = 0;
|
||||
map.r.digital_filter_select_2.LPF = 0;
|
||||
map.r.digital_filter_select_2.FIL3 = 0;
|
||||
map.r.digital_filter_select_2.EQ0 = 0;
|
||||
map.r.digital_filter_select_2.GN = 0b00;
|
||||
update(Register::DigitalFilterSelect2);
|
||||
|
||||
// Acitivating digital block , power supply
|
||||
map.r.power_management_1.PMADL = 1; // ADC Lch = Lch input signal. Mic Amp Lch and ADC Lch Power Management
|
||||
map.r.power_management_1.PMADR = 1; // ADC Rch = Rch input signal. Mic Amp Rch and ADC Rch Power Management
|
||||
map.r.power_management_1.PMPFIL = 1; // Pre-loaded in top part. Orig value=0, Programmable Digital filter unused (not power up), routed around.
|
||||
update(Register::PowerManagement1); // Activating the Power management of the used blocks . (Mic ADC always + Dig Block filter , when used )
|
||||
map.r.digital_filter_select_3.EQ1 = 0;
|
||||
map.r.digital_filter_select_3.EQ2 = 0;
|
||||
map.r.digital_filter_select_3.EQ3 = 0;
|
||||
map.r.digital_filter_select_3.EQ4 = 0;
|
||||
map.r.digital_filter_select_3.EQ5 = 0;
|
||||
update(Register::DigitalFilterSelect3);
|
||||
*/
|
||||
map.r.digital_filter_mode.PFSDO = 0; // ADC (+ 1st order HPF) Output
|
||||
map.r.digital_filter_mode.ADCPF = 1; // ADC Output (default)
|
||||
update(Register::DigitalFilterMode);
|
||||
|
||||
// ... Set coefficients ...
|
||||
|
||||
map.r.power_management_1.PMADL = 1; // ADC Lch = Lch input signal
|
||||
map.r.power_management_1.PMADR = 1; // ADC Rch = Rch input signal
|
||||
map.r.power_management_1.PMPFIL = 0; // Programmable filter unused, routed around.
|
||||
update(Register::PowerManagement1);
|
||||
|
||||
// 1059/fs, 22ms @ 48kHz
|
||||
chThdSleepMilliseconds(22);
|
||||
|
||||
}
|
||||
|
||||
// Common part for all alc_mode , --------------------------
|
||||
// const uint_fast8_t mgain = 0b0111; // Already pre-loaded , in above switch case .
|
||||
map.r.signal_select_1.MGAIN20 = mgain & 7; // writing 3 lower bits of mgain , (pre-amp mic gain).
|
||||
map.r.signal_select_1.PMMP = 1; // Activating DC Mic Power supply through 2kohms res., similar majority smartphones headphone+mic jack, "plug-in-power"
|
||||
map.r.signal_select_1.MPSEL = 1; // MPWR2 pin ,selecting output voltage to MPWR2 pin, that we are using in portapack ext. MIC)
|
||||
map.r.signal_select_1.MGAIN3 = (mgain >> 3) & 1; // writing 4th upper bit of mgain (pre-amp mic gain).
|
||||
update(Register::SignalSelect1);
|
||||
|
||||
}
|
||||
|
||||
|
||||
|
||||
void AK4951::microphone_disable() {
|
||||
map.r.power_management_1.PMADL = 0; // original code , disable Power managem.Mic ADC L
|
||||
map.r.power_management_1.PMADR = 0; // original code , disable Power managem.Mic ADC R
|
||||
map.r.power_management_1.PMPFIL = 0; // original code , disable Power managem. all Programmable Dig. block
|
||||
map.r.power_management_1.PMADL = 0;
|
||||
map.r.power_management_1.PMADR = 0;
|
||||
map.r.power_management_1.PMPFIL = 0;
|
||||
update(Register::PowerManagement1);
|
||||
|
||||
map.r.alc_mode_control_1.ALC = 0; // original code , Restore , disable ALC block.
|
||||
map.r.alc_mode_control_1.ALC = 0;
|
||||
update(Register::ALCModeControl1);
|
||||
|
||||
map.r.auto_hpf_control.AHPF = 0; //----------- new code addition , Restore disable Wind noise filter OFF (AHPF bit=“0”).
|
||||
update(Register::AutoHPFControl);
|
||||
|
||||
//Restore original AUDIO PATH , condition, (Digital filter block PATH is BY PASSED) (we can also swith off DIG. BLOCK power , PMPFIL=0)
|
||||
// The Path in Recording Mode 2 & Playback Mode 2 , (NO DIG FILTER BLOCK AT ALL, not for MIC recording, nor for Playback)
|
||||
map.r.digital_filter_mode.ADCPF = 1; // new code addition , ADCPF bit swith ("0" Mic after ADC Output connected (recording mode) to the DIGITAL FILTER BLOCK. ("1" Playback mode)
|
||||
map.r.digital_filter_mode.PFSDO = 0; // new code addition , ADC bit switch ("0" : 1st order HPF) connectedto the Output. By bass DIGITAL block .
|
||||
map.r.digital_filter_mode.PFDAC = 0b00; // new code addition , (Input selector for DAC (not used in MIC), SDTI= Audio Serial Data Input Pin)
|
||||
update(Register::DigitalFilterMode); // Writing the Audio Path : NO DIGITAL BLOCK or DIG BLOCK FOR MIC , Audio mode path : Playback mode /-Recording mode.
|
||||
|
||||
// Restore original condition , LPF , OFF . same as when not using DIGITAL Programmable block
|
||||
map.r.digital_filter_select_2.LPF = 0; // LPF-Block, Coeffic Setting Enable (OFF-Default), When LPF bit is “0”, audio data passes the LPF block by 0dB gain.
|
||||
update(Register::DigitalFilterSelect2);
|
||||
|
||||
map.r.lpf_coefficient_0.l = 0x00; // Pre-loading here LPF 6k , 1st Order from digital Block , Fc=6000 Hz, fs = 48khz
|
||||
map.r.lpf_coefficient_1.h = 0x00; // LPF bit is activated down, for all ALC digital modes.
|
||||
map.r.lpf_coefficient_2.l = 0x00; // Writting reg to AK4951 , down with update....
|
||||
map.r.lpf_coefficient_3.h = 0x00;
|
||||
|
||||
update(Register::LPFCoefficient0); // Writing pre-loaded 4 bytes LPF CoefFiecients 14 bits (FSA13..0, FSB13..0
|
||||
update(Register::LPFCoefficient1);
|
||||
update(Register::LPFCoefficient2);
|
||||
update(Register::LPFCoefficient3);
|
||||
|
||||
// Switch off all EQ 1,2,3,4,5
|
||||
map.r.digital_filter_select_3.EQ1 = 0; // EQ1 Coeffic Setting , (0: Disable-default, audio data passes EQ1 block by 0dB gain). When EQ1="1”, the settings of E1A15-0, E1B15-0 and E1C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ2 = 0; // EQ2 Coeffic Setting , (0: Disable-default, audio data passes EQ2 block by 0dB gain). When EQ2="1”, the settings of E2A15-0, E2B15-0 and E2C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ3 = 0; // EQ3 Coeffic Setting , (0: Disable-default, audio data passes EQ3 block by 0dB gain). When EQ3="1”, the settings of E3A15-0, E3B15-0 and E3C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ4 = 0; // EQ4 Coeffic Setting , (0: Disable-default, audio data passes EQ4 block by 0dB gain). When EQ4="1”, the settings of E4A15-0, E4B15-0 and E4C15-0 bits are enabled
|
||||
map.r.digital_filter_select_3.EQ5 = 0; // EQ5 Coeffic Setting , (0: Disable-default, audio data passes EQ5 block by 0dB gain). When EQ5="1”, the settings of E5A15-0, E5B15-0 and E5C15-0 bits are enabled
|
||||
update(Register::DigitalFilterSelect3);
|
||||
|
||||
}
|
||||
|
||||
reg_t AK4951::read(const address_t reg_address) {
|
||||
|
||||
@ -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 {
|
||||
|
||||
@ -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 {
|
||||
|
||||
Loading…
Reference in New Issue
Block a user