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Invest limit bw ssb to 3 k (#2188)

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* Adding_TX_BW_user_selection_to_the_Mic_SSB_mode

* TX_SSB_BW_adjustm_2k_3k_4k

* init_value_correction

* Comment_correction

* Leaving_only_OK_txbw_SSB_2K_3K

* Limti_SSB_TX_BW_2K_3K
This commit is contained in:
Brumi-2021
2024-07-06 13:34:09 +02:00
committed by GitHub
parent b53b5c70d3
commit 032168234a
7 changed files with 73 additions and 18 deletions
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13
firmware/baseband/dsp_hilbert.cpp
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@@ -27,6 +27,7 @@ namespace dsp {
HilbertTransform::HilbertTransform() {
n = 0;
sos_input.configure(half_band_lpf_config);
sos_i.configure(half_band_lpf_config);
sos_q.configure(half_band_lpf_config);
}
@@ -34,22 +35,26 @@ HilbertTransform::HilbertTransform() {
void HilbertTransform::execute(float in, float& out_i, float& out_q) {
float a = 0, b = 0;
float in_filtered = sos_input.execute(in) * 1.0f; // Anti-aliasing LPF at fs/4 mic audio filter front-end.
// it is exactly matching with the usefull mic BW_cut_off =fs/4 of the following Hilbert Transf. implementation
// Ex. to TX SSB 3khz BW (3KHZ =fs/4) ,we need to run Hiblert_Trf fs = 12khz . And our anti-aliasing filter is also fcut =fs/4 when we use a half_band_LPF .
switch (n) {
case 0:
a = in;
a = in_filtered;
b = 0;
break;
case 1:
a = 0;
b = -in;
b = -in_filtered;
break;
case 2:
a = -in;
a = -in_filtered;
b = 0;
break;
case 3:
a = 0;
b = in;
b = in_filtered;
break;
}

1
firmware/baseband/dsp_hilbert.hpp
View File

@@ -34,6 +34,7 @@ class HilbertTransform {
private:
uint8_t n = 0;
SOSFilter<5> sos_input = {};
SOSFilter<5> sos_i = {};
SOSFilter<5> sos_q = {};
};

22
firmware/baseband/dsp_modulate.cpp
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@@ -81,6 +81,25 @@ SSB::SSB()
mode = Mode::LSB;
}
void SSB::set_fs_div_factor(float new_bw_ssb) {
switch ((int)new_bw_ssb / 1000) {
case 2:
fs_div_factor = 192; // TXBW_ssb = 2khz = fs/4 ; fs of Hilbert Transform => 8khz fs Hilbert = 1536.000/8000= 192
break;
case 3:
fs_div_factor = 128; // TXBW_ssb = 3khz = fs/4 ; fs of Hilbert Transform => 12khz fs Hilbert = 1536.000/12000= 128
break;
/* TODO pending to investigate 4khz, now is making aliasing folding with audio sweep .
case 4:
fs_div_factor = 96; // TXBW_ssb = 4khz = fs/4 ; fs of Hilbert Transform => 16khz fs Hilbert = 1536.000/16000= 96
break;
*/
default:
fs_div_factor = 128; // TXBW_ssb = 3khz = fs/4 ; fs of Hilbert Transform => 12khz fs Hilbert = 1536.000/12000= 128
break;
}
}
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) {
// unused
(void)configured_in;
@@ -93,9 +112,10 @@ void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer, bool& co
int8_t re = 0, im = 0;
for (size_t counter = 0; counter < buffer.count; counter++) {
if (counter % 128 == 0) {
if (counter % fs_div_factor == 0) { // Ex. TX bw_ssb 3khz =fs/4 Hilbert Transform sample rate, 128 = 1.536.000 hz / 12.000 hz (fs H.T.)
float i = 0.0, q = 0.0;
// over = 1.536.000/24khz = 64 . (Mic audio has fixed SR in audio_p buffer[] = 24khz), but in tx mode , we are running Transceiver fs @tx = 1.536.000 Hz.
sample = audio.p[counter / over] >> audio_shift_bits_s16_AM_DSB_SSB; // originally fixed >> 2, now >>2 for AK, 0,1,2,3 for WM (boost off)
sample *= audio_gain; // Apply GAIN Scale factor to the audio TX modulation.

3
firmware/baseband/dsp_modulate.hpp
View File

@@ -79,9 +79,12 @@ class SSB : public Modulator {
SSB();
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_fs_div_factor(float new_bw_ssb);
private:
dsp::HilbertTransform hilbert;
float new_bw_ssb{3.0};
int fs_div_factor{128};
};
///

22
firmware/baseband/proc_mictx.cpp
View File

@@ -113,13 +113,27 @@ void MicTXProcessor::on_message(const Message* const msg) {
}
if (usb_enabled) {
modulator = new dsp::modulate::SSB();
modulator->set_mode(dsp::modulate::Mode::USB);
dsp::modulate::SSB* ssb = new dsp::modulate::SSB();
// Config fs_div_factor private var inside DSP modulate.cpp
ssb->set_fs_div_factor(config_message.deviation_hz); // we use same FM var deviation_hz , to set up also SSB BW
ssb->set_mode(dsp::modulate::Mode::USB);
modulator = ssb;
// modulator = new dsp::modulate::SSB(); // Keeping previous code as ref., when not passing deviation_hz parameter.
// modulator->set_mode(dsp::modulate::Mode::USB);
}
if (lsb_enabled) {
modulator = new dsp::modulate::SSB();
modulator->set_mode(dsp::modulate::Mode::LSB);
dsp::modulate::SSB* ssb = new dsp::modulate::SSB();
// Config fs_div_factor private var inside DSP modulate.cpp
ssb->set_fs_div_factor(config_message.deviation_hz); // we use same FM var deviation_hz , to set up also SSB BW
ssb->set_mode(dsp::modulate::Mode::LSB);
modulator = ssb;
// modulator = new dsp::modulate::SSB(); // Keeping previous code as ref., when not passing deviation_hz parameter.
// modulator->set_mode(dsp::modulate::Mode::LSB);
}
if (am_enabled) {
modulator = new dsp::modulate::AM();