make lpf a IIR

lib/filters.c

@@ -12,32 +12,15 @@ float apply_pre_emphasis(Emphasis *pe, float sample) {
 }
 
 void init_low_pass_filter(LowPassFilter *lp, float cutoff_frequency, float sample_rate) {
-    for (int i = 0; i < FIR_TAPS; i++) {
+    float rc = 1/(M_2PI*cutoff_frequency);
-        for (int j = 0; j < FIR_PHASES; j++) {
+    lp->alpha = sample_rate/(sample_rate+rc);
-            int mi = i * FIR_PHASES + j + 1;
+    lp->prev_sample = 0.0f;
-            float sincpos = mi - (((FIR_TAPS * FIR_PHASES) + 1.0f) / 2.0f);
-            float firlowpass = (sincpos == 0.0f) ? 1.0f : sinf(M_2PI * cutoff_frequency * sincpos / sample_rate) / (PI * sincpos);
-            float window = 0.54f - 0.46f * cosf(M_2PI * mi / (FIR_TAPS * FIR_PHASES)); // Hamming window
-            lp->low_pass_fir[j][i] = firlowpass * window;
-        }
-    }
-    memset(lp->sample_buffer, 0, sizeof(lp->sample_buffer));
-    lp->buffer_index = 0;
 }
 
 float apply_low_pass_filter(LowPassFilter *lp, float sample) {
-    // Update the sample buffer
+    float output = lp->alpha*sample+(1-lp->alpha)*lp->prev_sample;
-    lp->sample_buffer[lp->buffer_index] = sample;
+    lp->prev_sample = output;
-    lp->buffer_index = (lp->buffer_index + 1) % FIR_TAPS;
+    return output;
-
-    // Apply the filter
-    float result = 0.0f;
-    int index = lp->buffer_index;
-    for (int i = 0; i < FIR_TAPS; i++) {
-        result += lp->low_pass_fir[0][i] * lp->sample_buffer[index];
-        index = (index + 1) % FIR_TAPS;
-    }
-    return result*6;
 }
 
 void init_delay_line(DelayLine *delay_line, int max_delay) {

lib/filters.h

@@ -15,9 +15,8 @@ void init_emphasis(Emphasis *pe, float tau, float sample_rate);
 float apply_pre_emphasis(Emphasis *pe, float sample);
 
 typedef struct {
-    float low_pass_fir[FIR_PHASES][FIR_TAPS];
+    float alpha;
-    float sample_buffer[FIR_TAPS];
+    float prev_sample;
-    int buffer_index;
 } LowPassFilter;
 
 void init_low_pass_filter(LowPassFilter *lp, float cutoff_frequency, float sample_rate);

src/crosby_stereo_coder.c

@@ -63,12 +63,12 @@ int main() {
     printf("CrosbySTCode : Stereo encoder (using the crosby system) made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
     // Define formats and buffer atributes
     pa_sample_spec stereo_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
         .channels = 2,
         .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
     };
     pa_sample_spec mono_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE,
         .channels = 1,
         .rate = SAMPLE_RATE
     };

src/polar_stereo_coder.c

@@ -62,12 +62,12 @@ int main() {
     printf("PSTCode : (Polar) Stereo encoder made by radio95 (with help of ChatGPT and Claude, thanks!). Note that this version is for the OIRT band which is in use in Russia, Belarus and other CIS countries\n");
     // Define formats and buffer atributes
     pa_sample_spec stereo_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
         .channels = 2,
         .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
     };
     pa_sample_spec mono_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE,
         .channels = 1,
         .rate = SAMPLE_RATE
     };

src/sca_mod.c

@@ -57,7 +57,7 @@ int main() {
 
     // Define formats and buffer atributes
     pa_sample_spec audio_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
         .channels = 1,
         .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
     };

src/ssb_stereo_coder.c

@@ -65,12 +65,12 @@ int main() {
     printf("SSB-STCode : Stereo encoder made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
     // Define formats and buffer atributes
     pa_sample_spec stereo_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
         .channels = 2,
         .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
     };
     pa_sample_spec mono_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE,
         .channels = 1,
         .rate = SAMPLE_RATE
     };

src/stereo_coder.c

@@ -63,12 +63,12 @@ int main() {
     printf("STCode : Stereo encoder made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
     // Define formats and buffer atributes
     pa_sample_spec stereo_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
         .channels = 2,
         .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
     };
     pa_sample_spec mono_format = {
-        .format = PA_SAMPLE_FLOAT32LE,
+        .format = PA_SAMPLE_FLOAT32NE,
         .channels = 1,
         .rate = SAMPLE_RATE
     };