Adding Wefax demodulation mode inside Audio App (#2539)

* Adding_new_WFAX_GUI_mode_Audio_App

* Wefax_APT_demodulation_structure

* Solving REC Apt signal.wav from WFAX

* clang format issues

* correcting comments

firmware/baseband/audio_output.cpp

@@ -72,7 +72,7 @@ void AudioOutput::on_block(const buffer_f32_t& audio) {
     if (do_processing) {
         const auto audio_present_now = squelch.execute(audio);
 
-        hpf.execute_in_place(audio);
+        hpf.execute_in_place(audio);  // IIRBiquadFilter name is "hpf", but we will call with "hpf-coef" for all  except WFAX with "lpf-coef".
         deemph.execute_in_place(audio);
 
         audio_present_history = (audio_present_history << 1) | (audio_present_now ? 1 : 0);

firmware/baseband/dsp_demodulate.cpp

@@ -24,6 +24,8 @@
 #include "complex.hpp"
 #include "fxpt_atan2.hpp"
 #include "utility_m4.hpp"
+#include "dsp_hilbert.hpp"
+#include "dsp_modulate.hpp"
 
 #include <hal.h>
 
@@ -63,12 +65,7 @@ buffer_f32_t SSB::execute(
 
     return {dst.p, src.count, src.sampling_rate};
 }
-/*
-static inline float angle_approx_4deg0(const complex32_t t) {
-        const auto x = static_cast<float>(t.imag()) / static_cast<float>(t.real());
-        return 16384.0f * x;
-}
-*/
+
 static inline float angle_approx_0deg27(const complex32_t t) {
     if (t.real()) {
         const auto x = static_cast<float>(t.imag()) / static_cast<float>(t.real());
@@ -82,6 +79,32 @@ static inline float angle_precise(const complex32_t t) {
     return atan2f(t.imag(), t.real());
 }
 
+buffer_f32_t SSB_FM::execute(    // Added to handle WFAX (HF weather map )-
+    const buffer_c16_t& src,     // input arg , pointer Complex c16 i,q buffer.
+    const buffer_f32_t& dst) {   // input arg , pointer f32 buffer audio demodulated
+    complex16_t* src_p = src.p;  // removed const ; init src_p pointer with the mem address pointed by src.p.
+    const auto src_end = &src.p[src.count];
+    auto dst_p = dst.p;
+    float mag_sq_lpf_norm;
+
+    while (src_p < src_end) {
+        // FM APT audio tone demod: real part (USB-differentiator)  and AM tone demodulation + lpf (to remove the subcarrier.)
+        real_to_complex.execute((src_p++)->real(), mag_sq_lpf_norm);
+        *(dst_p++) = mag_sq_lpf_norm;  // already normalized/32.768f and clipped to +1.0f for the wav file.
+
+        real_to_complex.execute((src_p++)->real(), mag_sq_lpf_norm);
+        *(dst_p++) = mag_sq_lpf_norm;
+
+        real_to_complex.execute((src_p++)->real(), mag_sq_lpf_norm);
+        *(dst_p++) = mag_sq_lpf_norm;
+
+        real_to_complex.execute((src_p++)->real(), mag_sq_lpf_norm);
+        *(dst_p++) = mag_sq_lpf_norm;
+    }
+
+    return {dst.p, src.count, src.sampling_rate};
+}
+
 buffer_f32_t FM::execute(
     const buffer_c16_t& src,
     const buffer_f32_t& dst) {

firmware/baseband/dsp_demodulate.hpp

@@ -23,6 +23,7 @@
 #define __DSP_DEMODULATE_H__
 
 #include "dsp_types.hpp"
+#include "dsp_hilbert.hpp"
 
 namespace dsp {
 namespace demodulate {
@@ -47,6 +48,17 @@ class SSB {
     static constexpr float k = 1.0f / 32768.0f;
 };
 
+class SSB_FM {  // Added to handle WFAX-
+   public:
+    buffer_f32_t execute(
+        const buffer_c16_t& src,
+        const buffer_f32_t& dst);
+
+   private:
+    static constexpr float k = 1.0f / 32768.0f;
+    dsp::Real_to_Complex real_to_complex;  // It is a member variable of SSB_FM.
+};
+
 class FM {
    public:
     buffer_f32_t execute(

firmware/baseband/dsp_hilbert.cpp

@@ -21,6 +21,7 @@
 
 #include "dsp_hilbert.hpp"
 #include "dsp_sos_config.hpp"
+#include "utility_m4.hpp"
 
 namespace dsp {
 
@@ -83,4 +84,88 @@ void HilbertTransform::execute(float in, float& out_i, float& out_q) {
     n = (n + 1) % 4;
 }
 
+Real_to_Complex::Real_to_Complex() {
+    // No need to call a separate configuration method like "Real_to_Complex()" externally before using the execute() method
+    // This is the constructor for the Real_to_Complex class.
+    // It initializes the member variables and calls the configure function for the sos_input, sos_i, and sos_q filters.
+    // to ensure the object is ready to use right after instantiation.
+
+    n = 0;
+
+    sos_input.configure(full_band_lpf_config);
+    sos_i.configure(full_band_lpf_config);
+    sos_q.configure(full_band_lpf_config);
+    sos_mag_sq.configure(quarter_band_lpf_config);  // for APT LPF subcarrier filter. (1/4 Nyquist fs/2 = 1/4 * 12Khz/2 = 1.5khz)
+}
+
+void Real_to_Complex::execute(float in, float& out_mag_sq_lpf) {
+    // Full_band LPF  means a LP filter with f_cut_off = fs/2; Full band = Full max band = 1/2 * fs_max =  1.0 x f_Nyquist = 1 * fs/2 = fs/2
+    float a = 0, b = 0;
+    float out_i = 0, out_q = 0, out_mag_sq = 0;
+    // int32_t packed;
+
+    float in_filtered = sos_input.execute(in) * 1.0f;  // Anti-aliasing full band LPF,  fc = fs/2= 6k, audio filter front-end.
+
+    switch (n) {
+        case 0:
+            a = in_filtered;
+            b = 0;
+            break;
+        case 1:
+            a = 0;
+            b = -in_filtered;
+            break;
+        case 2:
+            a = -in_filtered;
+            b = 0;
+            break;
+        case 3:
+            a = 0;
+            b = in_filtered;
+            break;
+    }
+
+    float i = sos_i.execute(a) * 1.0f;  // better keep <1.0f to minimize recorded APT(t) black level artifacts.-
+    float q = sos_q.execute(b) * 1.0f;
+
+    switch (n) {  // shifting down -fs4 (fs = 12khz , fs/4 = 3khz)
+        case 0:
+            out_i = i;
+            out_q = q;
+            break;
+        case 1:
+            out_i = -q;
+            out_q = i;
+            break;
+        case 2:
+            out_i = -i;
+            out_q = -q;
+            break;
+        case 3:
+            out_i = q;
+            out_q = -i;
+            break;
+    }
+
+    n = (n + 1) % 4;
+
+    /*   res = __smuad(val1,val2);  	p1 = val1[15:0] × val2[15:0]
+                                                p2 = val1[31:16] × val2[31:16]
+                                                res[31:0] = p1 + p2
+                                        return res; */
+
+    // Not strict Magnitude complex calculation, it is a cross multiplication (lower 16 bit real x lower 16 imag) + 0 (higher 16 bits comp),
+    // but better visual results comparing real magnitude calculation, (better map diagonal lines reproduction, and less artifacts in APT signal(t)
+    out_mag_sq = __SMUAD(out_i, out_q);                        // "cross-magnitude" of the complex (out_i + j out_q)
+    out_mag_sq_lpf = sos_mag_sq.execute((out_mag_sq)) * 2.0f;  // LPF quater band = 1.5khz APT signal
+
+    out_mag_sq_lpf /= 32768.0f;  // normalize ;
+    // Compress clipping positive APT signal [-1.5 ..1.5] input , converted to [-1.0 ...1.0] with "S" compressor gain shape.
+    if (out_mag_sq_lpf > 1.0f) {
+        out_mag_sq_lpf = 1.0f;  // clipped signal  at +1.0f,  APT signal is positive, no need to clip -1.0
+    } else {
+        out_mag_sq_lpf = out_mag_sq_lpf * (1.5f - ((out_mag_sq_lpf * out_mag_sq_lpf) / 2.0f));
+    }
+}
+
 } /* namespace dsp */

firmware/baseband/dsp_hilbert.hpp

@@ -39,6 +39,19 @@ class HilbertTransform {
     SOSFilter<5> sos_q = {};
 };
 
+class Real_to_Complex {
+   public:
+    Real_to_Complex();  // Additional initialization
+    void execute(float in, float& out_mag_sq_lpf);
+
+   private:
+    uint8_t n = 0;
+    SOSFilter<5> sos_input = {};
+    SOSFilter<5> sos_i = {};
+    SOSFilter<5> sos_q = {};
+    SOSFilter<5> sos_mag_sq = {};
+};
+
 } /* namespace dsp */
 
 #endif /*__DSP_HILBERT_H__*/

firmware/baseband/proc_am_audio.cpp

@@ -27,6 +27,7 @@
 #include "event_m4.hpp"
 
 #include <array>
+#include "dsp_hilbert.hpp"
 
 void NarrowbandAMAudio::execute(const buffer_c8_t& buffer) {
     if (!configured) {
@@ -44,16 +45,28 @@ void NarrowbandAMAudio::execute(const buffer_c8_t& buffer) {
     // TODO: Feed channel_stats post-decimation data?
     feed_channel_stats(channel_out);
 
-    auto audio = demodulate(channel_out);
+    auto audio = demodulate(channel_out);  // now 3 AM demodulation types : demod_am, demod_ssb, demod_ssb_fm (for Wefax)
     audio_compressor.execute_in_place(audio);
     audio_output.write(audio);
 }
 
 buffer_f32_t NarrowbandAMAudio::demodulate(const buffer_c16_t& channel) {
-    if (modulation_ssb) {
-        return demod_ssb.execute(channel, audio_buffer);
-    } else {
-        return demod_am.execute(channel, audio_buffer);
+    switch (modulation_ssb) {  // enum class Modulation : int32_t {DSB = 0, SSB = 1, SSB_FM = 2}
+        case (int)(AMConfigureMessage::Modulation::DSB):
+            return demod_am.execute(channel, audio_buffer);
+            break;
+
+        case (int)(AMConfigureMessage::Modulation::SSB):
+            return demod_ssb.execute(channel, audio_buffer);
+            break;
+
+        case (int)(AMConfigureMessage::Modulation::SSB_FM):  // Added to handle Weather Fax mode.
+            // chDbgPanic("case SSB_FM demodulation");                   // Debug.
+            return demod_ssb_fm.execute(channel, audio_buffer);  // Calling a derivative of demod_ssb (USB) , but with different FIR taps + FM audio tones demod.
+            break;
+
+        default:
+            break;
     }
 }
 
@@ -98,8 +111,9 @@ void NarrowbandAMAudio::configure(const AMConfigureMessage& message) {
     channel_filter_high_f = message.channel_filter.high_frequency_normalized * channel_filter_input_fs;
     channel_filter_transition = message.channel_filter.transition_normalized * channel_filter_input_fs;
     channel_spectrum.set_decimation_factor(1.0f);
-    modulation_ssb = (message.modulation == AMConfigureMessage::Modulation::SSB);
-    audio_output.configure(message.audio_hpf_config);
+    // modulation_ssb = (message.modulation == AMConfigureMessage::Modulation::SSB);  // originally we had just 2 AM types of demod. (DSB , SSB)
+    modulation_ssb = (int)message.modulation;              // now sending by message , 3 types of AM demod :   enum class Modulation : int32_t {DSB = 0, SSB = 1, SSB_FM = 2}
+    audio_output.configure(message.audio_hpf_lpf_config);  // hpf in all AM demod modes (AM-6K/9K, USB/LSB,DSB), except Wefax (lpf there).
 
     configured = true;
 }

firmware/baseband/proc_am_audio.hpp

@@ -63,9 +63,11 @@ class NarrowbandAMAudio : public BasebandProcessor {
     int32_t channel_filter_transition = 0;
     bool configured{false};
 
-    bool modulation_ssb = false;
+    // bool modulation_ssb = false;  // Origianlly we only had 2 AM demod types {DSB = 0, SSB = 1} , and we could handle it with bool var , 1 bit.
+    int8_t modulation_ssb = 0;  // Now we have 3 AM demod types we will send now index integer  {DSB = 0, SSB = 1, SSB_FM = 2}
     dsp::demodulate::AM demod_am{};
     dsp::demodulate::SSB demod_ssb{};
+    dsp::demodulate::SSB_FM demod_ssb_fm{};  // added for Wfax mode.
     FeedForwardCompressor audio_compressor{};
     AudioOutput audio_output{};