nug fixes and small tweaks

.vscode/.server-controller-port.log

@@ -1,5 +1,5 @@
 {
   "port": 13452,
-  "time": 1737972017853,
+  "time": 1738158496504,
   "version": "0.0.3"
 }

lib/constants.h

@@ -1,6 +1,6 @@
-#ifndef PI
+#ifndef M_PI
-#define PI 3.14159265358979323846
+#define M_PI 3.14159265358979323846
 #endif
 #ifndef M_2PI
-#define M_2PI (3.14159265358979323846 * 2.0)
+#define M_2PI (M_PI * 2.0)
 #endif

lib/filters.c

@@ -25,11 +25,11 @@ void init_lpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate) {
         float m = n - (FILTER_TAPS - 1.0f) / 2.0f;
         
         // Sinc function
-        float sinc = (m == 0) ? 1.0f : sinf(PI * m * fc) / (PI * m);
+        float sinc = (m == 0) ? 1.0f : sinf(M_PI * m * fc) / (M_PI * m);
         
         // Blackman window
-        float window = 0.42f - 0.5f * cosf(2.0f * PI * n / (FILTER_TAPS - 1)) 
+        float window = 0.42f - 0.5f * cosf(2.0f * M_PI * n / (FILTER_TAPS - 1)) 
-                     + 0.08f * cosf(4.0f * PI * n / (FILTER_TAPS - 1));
+                     + 0.08f * cosf(4.0f * M_PI * n / (FILTER_TAPS - 1));
         
         filter->coeffs[n] = sinc * window;
     }
@@ -53,11 +53,11 @@ void init_hpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate) {
         float m = n - (FILTER_TAPS - 1.0f) / 2.0f;
         
         // Sinc function
-        float sinc = (m == 0) ? -1.0f : sinf(PI * m * fc) / (PI * m);
+        float sinc = (m == 0) ? 1.0f : -sinf(M_PI * m * fc) / (M_PI * m);
         
         // Blackman window
-        float window = 0.42f - 0.5f * cosf(2.0f * PI * n / (FILTER_TAPS - 1)) 
+        float window = 0.42f - 0.5f * cosf(2.0f * M_PI * n / (FILTER_TAPS - 1)) 
-                     + 0.08f * cosf(4.0f * PI * n / (FILTER_TAPS - 1));
+                     + 0.08f * cosf(4.0f * M_PI * n / (FILTER_TAPS - 1));
         
         filter->coeffs[n] = sinc * window;
     }
@@ -148,4 +148,5 @@ float delay_line(DelayLine *delay_line, float in) {
 
 void exit_delay_line(DelayLine *delay_line) {
     free(delay_line->buffer);
+    delay_line->buffer = NULL;
 }

lib/fm_modulator.c

@@ -8,6 +8,7 @@ void init_fm_modulator(FMModulator *fm, float frequency, float deviation, float
 
 float modulate_fm(FMModulator *fm, float sample) {
     float inst_freq = fm->frequency+(sample*fm->deviation);
+    if (inst_freq < 0.0f) inst_freq = 0.0f;
     change_oscillator_frequency(&fm->osc, inst_freq);
     return get_oscillator_sin_sample(&fm->osc);
 }

lib/hilbert.c

@@ -6,7 +6,7 @@ void compute_hilbert_coeffs(float* coeffs, int taps) {
         if ((i - mid) % 2 == 0) {
             coeffs[i] = 0.0f;
         } else {
-            coeffs[i] = 2.0f / (PI * (i - mid));
+            coeffs[i] = 2.0f / (M_PI * (i - mid));
         }
     }
 }

lib/oscillator.c

@@ -12,25 +12,23 @@ void change_oscillator_frequency(Oscillator *osc, float frequency) {
 
 float get_oscillator_sin_sample(Oscillator *osc) {
     float sample = sinf(osc->phase);
-    osc->phase += osc->phase_increment;
+    advance_oscillator(osc);
-    if (osc->phase >= M_2PI) {
-        osc->phase -= M_2PI;
-    }
     return sample;
 }
 
 float get_oscillator_cos_sample(Oscillator *osc) {
     float sample = cosf(osc->phase);
-    osc->phase += osc->phase_increment;
+    advance_oscillator(osc);
-    if (osc->phase >= M_2PI) {
-        osc->phase -= M_2PI;
-    }
     return sample;
 }
 
-float get_oscillator_sin_multiplier_ni(Oscillator *osc, float multiplier) {
+float get_oscillator_sin_multiplier_ni(Oscillator *osc, float multiplier) { // ni = No Increment
     return sinf(osc->phase*multiplier);
 }
 float get_oscillator_cos_multiplier_ni(Oscillator *osc, float multiplier) {
     return cosf(osc->phase*multiplier);
+}
+
+void advance_oscillator(Oscillator *osc) {
+    osc->phase = fmodf(osc->phase + osc->phase_increment, M_2PI);
 }

lib/oscillator.h

@@ -14,4 +14,5 @@ void change_oscillator_frequency(Oscillator *osc, float frequency);
 float get_oscillator_sin_sample(Oscillator *osc);
 float get_oscillator_cos_sample(Oscillator *osc);
 float get_oscillator_sin_multiplier_ni(Oscillator *osc, float multiplier);
-float get_oscillator_cos_multiplier_ni(Oscillator *osc, float multiplier);
+float get_oscillator_cos_multiplier_ni(Oscillator *osc, float multiplier);
+void advance_oscillator(Oscillator *osc);

src/fm95.c

@@ -4,7 +4,7 @@
 
 #define buffer_maxlength 12288
 #define buffer_tlength_fragsize 12288
-#define buffer_prebuf 16
+#define buffer_prebuf 32
 
 #define DEFAULT_STEREO 1
 #define DEFAULT_STEREO_POLAR 0
@@ -14,8 +14,8 @@
 #define DEFAULT_ALSA_OUTPUT 1
 #define DEFAULT_SCA_FREQUENCY 67000.0f
 #define DEFAULT_SCA_DEVIATION 7000.0f
-#define DEFAULT_SCA_CLIPPER_THRESHOLD 1.0f // Full deviation
+#define DEFAULT_SCA_CLIPPER_THRESHOLD 1.0f // Full deviation, if you set this to 0.5 then you may as well set the deviation to 3.5k
-#define DEFAULT_PREEMPHASIS_TAU 50e-6 // Europe
+#define DEFAULT_PREEMPHASIS_TAU 50e-6 // Europe, the freedomers use 75µs
 
 //#define USB
 
@@ -25,7 +25,7 @@
 #include "../lib/hilbert.h"
 #include "../lib/fm_modulator.h"
 
-#define SAMPLE_RATE 192000 // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000)
+#define SAMPLE_RATE 192000
 
 #define INPUT_DEVICE "FM_Audio.monitor"
 #define OUTPUT_DEVICE "plughw:1,0"
@@ -46,6 +46,7 @@
 
 #define LPF_CUTOFF 15000 // Should't need to be changed
 #define HPF_CUTOFF 30 // Unless you wanna have SOME bass then leave this alone
+#define CENTER_BASS 50 // Bass upto this will be mono
 
 volatile sig_atomic_t to_run = 1;
 
@@ -398,7 +399,7 @@ int main(int argc, char **argv) {
     // #endregion
     // #region Setup Filters/Modulaltors/Oscillators
     Oscillator osc;
-    if(polar_stereo == 1) {
+    if(polar_stereo) {
         init_oscillator(&osc, 31250.0, SAMPLE_RATE); // The stereo carrier itself, the stereo carrier in polar is modulated directly on 31.25 khz with a bit of a carrier
     } else {
         init_oscillator(&osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
@@ -423,6 +424,9 @@ int main(int argc, char **argv) {
     FrequencyFilter hpf_l, hpf_r;
     init_hpf(&hpf_l, HPF_CUTOFF, SAMPLE_RATE);
     init_hpf(&hpf_r, HPF_CUTOFF, SAMPLE_RATE);
+
+    FrequencyFilter bass_hpf;
+    init_hpf(&bass_hpf, CENTER_BASS, SAMPLE_RATE);
     // #endregion
 
     signal(SIGINT, stop);
@@ -477,54 +481,47 @@ int main(int argc, char **argv) {
             float mono = (ready_l + ready_r) / 2.0f; // Stereo to Mono
             if(stereo == 1) {
                 float stereo = (ready_l - ready_r) / 2.0f; // Also Stereo to Mono but a bit diffrent
-                if(polar_stereo == 1) {
+                if(CENTER_BASS != 0) stereo = apply_frequency_filter(&bass_hpf, stereo);
-                    if(ssb) {
-                        float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 1); // Get stereo carrier via multiplication
-                        float stereo_carrier_cos = get_oscillator_cos_sample(&osc); // Get Carrier Q of I/Q
 
-                        float stereo_i, stereo_q;
+                float stereo_i, stereo_q;
-                        stereo += 0.2;
+                if(ssb) {
-                        apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // Compute I/Q
+                    // Compute hilbert here
-                        output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME +
+                    apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q);
-                            (stereo_i*stereo_carrier_cos+stereo_q*(stereo_carrier))*STEREO_VOLUME;
+                    mono = delay_line(&monoDelay, mono); // Delay Mono
-                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+                }
-                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
+                
+                if(polar_stereo) {
+                    float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 1);
+                    if(ssb) {
+                        float stereo_carrier_cos = get_oscillator_cos_multiplier_ni(&osc, 1); // Get Carrier Q of I/Q
+
+                        output[i] = mono*MONO_VOLUME +
+                            ((stereo_i+0.2)*stereo_carrier_cos+(stereo_q+0.2)*stereo_carrier)*STEREO_VOLUME;
                     } else {
-                        float stereo_carrier = get_oscillator_sin_sample(&osc);
+                        float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 1);
                         output[i] = mono*MONO_VOLUME +
                             ((stereo+0.2)*stereo_carrier)*STEREO_VOLUME;
-                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
-                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
                     }
                 } else {
+                    float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 2); // Get stereo carrier via multiplication
+                    float pilot = get_oscillator_sin_multiplier_ni(&osc, 1);
                     if(ssb) {
-                        float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 2); // Get stereo carrier via multiplication
                         float stereo_carrier_cos = get_oscillator_cos_multiplier_ni(&osc, 2); // Get Carrier Q of I/Q
-                        float pilot = get_oscillator_sin_sample(&osc);
+                        output[i] = mono*MONO_VOLUME +
-
-                        float stereo_i, stereo_q;
-                        apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // Compute I/Q
-
-                        output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME +
                             pilot*PILOT_VOLUME +
                             (stereo_i*stereo_carrier_cos+stereo_q*stereo_carrier)*STEREO_VOLUME;
-                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
-                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
                     } else {
-                        float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc,2);
-                        float pilot = get_oscillator_sin_sample(&osc);
                         output[i] = mono*MONO_VOLUME +
                             pilot*PILOT_VOLUME +
                             (stereo*stereo_carrier)*STEREO_VOLUME;
-                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
-                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
                     }
                 }
+                advance_oscillator(&osc);
             } else {
                 output[i] = mono*MONO_VOLUME; // Only Mono
-                if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
-                if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
             }
+            if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+            if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
         }
 
         if(alsa_output == 0) {