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change frequency filters implemententaton

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KubaPro010
2025-01-31 17:40:47 +01:00
parent 2557c19b2e
commit 9b4cec73c9
4 changed files with 66 additions and 89 deletions
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2
.vscode/.server-controller-port.log vendored
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@@ -1,5 +1,5 @@
{
"port": 13452,
"time": 1738336716329,
"time": 1738341533118,
"version": "0.0.3"
}

121
lib/filters.c
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@@ -48,80 +48,63 @@ float apply_preemphasis(BiquadFilter *filter, float input) {
return output;
}
void init_lpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate) {
float nyquist = sampleRate / 2.0f;
float fc = cutoffFreq / nyquist;
// Blackman window for sharp transition
for (int n = 0; n < FILTER_TAPS; n++) {
float m = n - (FILTER_TAPS - 1.0f) / 2.0f;
// Sinc function
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 * M_PI * n / (FILTER_TAPS - 1))
+ 0.08f * cosf(4.0f * M_PI * n / (FILTER_TAPS - 1));
filter->coeffs[n] = sinc * window;
}
// Normalize
float sum = 0;
for (int i = 0; i < FILTER_TAPS; i++) sum += filter->coeffs[i];
for (int i = 0; i < FILTER_TAPS; i++) filter->coeffs[i] /= sum;
// Clear delay line
memset(filter->delay, 0, sizeof(filter->delay));
filter->index = 0;
void init_lpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate) {
float x = (cutoffFreq * M_2PI) / sampleRate;
float sinX = sin(x);
float y = sinX / (qFactor*2.0f);
float cosX = cos(x);
float z = (1.0f-cosX)/2.0f;
float _a0 = y + 1.0f;
float _a1 = cosX * -2.0f;
float _a2 = 1.0f - y;
float _b0 = z;
float _b1 = 1.0f - cosX;
float _b2 = z;
filter->y2 = 0;
filter->y1 = 0;
filter->x2 = 0;
filter->x1 = 0;
filter->b0 = _b0/_a0;
filter->b1 = _b1/_a0;
filter->b2 = _b2/_a0;
filter->a1 = -_a1/_a0;
filter->a2 = -_a2/_a0;
}
void init_hpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate) {
float nyquist = sampleRate / 2.0f;
float fc = cutoffFreq / nyquist;
// Blackman window for sharp transition
for (int n = 0; n < FILTER_TAPS; n++) {
float m = n - (FILTER_TAPS - 1.0f) / 2.0f;
// Sinc function
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 * M_PI * n / (FILTER_TAPS - 1))
+ 0.08f * cosf(4.0f * M_PI * n / (FILTER_TAPS - 1));
filter->coeffs[n] = sinc * window;
}
void init_hpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate) {
float x = (cutoffFreq * M_2PI) / sampleRate;
float sinX = sin(x);
float y = sinX / (qFactor*2.0f);
float cosX = cos(x);
float z = (1.0f-cosX)/2.0f;
filter->coeffs[FILTER_TAPS/2] += 1.0f;
// Normalize
float sum = 0;
for (int i = 0; i < FILTER_TAPS; i++) sum += filter->coeffs[i];
for (int i = 0; i < FILTER_TAPS; i++) filter->coeffs[i] /= sum;
// Clear delay line
memset(filter->delay, 0, sizeof(filter->delay));
filter->index = 0;
float _a0 = y + 1.0f;
float _a1 = cosX * -2.0f;
float _a2 = 1.0f - y;
float _b0 = 1.0f - z;
float _b1 = cosX * -2.0f;
float _b2 = 1.0f - z;
filter->y2 = 0;
filter->y1 = 0;
filter->x2 = 0;
filter->x1 = 0;
filter->b0 = _b0/_a0;
filter->b1 = _b1/_a0;
filter->b2 = _b2/_a0;
filter->a1 = -_a1/_a0;
filter->a2 = -_a2/_a0;
}
float apply_frequency_filter(FrequencyFilter* filter, float input) {
// Shift delay line
filter->delay[filter->index] = input;
// Compute output
float output = 0;
int j = filter->index;
for (int i = 0; i < FILTER_TAPS; i++) {
output += filter->coeffs[i] * filter->delay[j];
j = (j + 1) % FILTER_TAPS;
}
// Update index
filter->index = (filter->index + FILTER_TAPS - 1) % FILTER_TAPS;
return output;
float apply_frequency_filter(BiquadFilter* filter, float input) {
float out = input*filter->b0+filter->x1*filter->b1+filter->x2*filter->b2+filter->y1*filter->a1+filter->y2*filter->a2;
filter->y2 = filter->y1;
filter->y1 = out;
filter->x2 = filter->x1;
filter->x1 = input;
return out;
}
float hard_clip(float sample, float threshold) {

12
lib/filters.h
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@@ -15,15 +15,9 @@ typedef struct {
void init_preemphasis(BiquadFilter *filter, float tau, float sample_rate);
float apply_preemphasis(BiquadFilter *filter, float input);
typedef struct {
float coeffs[FILTER_TAPS];
float delay[FILTER_TAPS];
int index;
} FrequencyFilter;
void init_lpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate);
void init_hpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate);
float apply_frequency_filter(FrequencyFilter* filter, float input);
void init_lpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate);
void init_hpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate);
float apply_frequency_filter(BiquadFilter* filter, float input);
float hard_clip(float sample, float threshold);
float soft_clip(float sample, float threshold);

20
src/fm95.c
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@@ -11,7 +11,7 @@
#define DEFAULT_STEREO_SSB 0
#define DEFAULT_CLIPPER_THRESHOLD 1.0f
#define DEFAULT_SOFT_CLIPPER_THRESHOLD 0.95f
#define DEFAULT_ALSA_OUTPUT 1
#define DEFAULT_ALSA_OUTPUT 0
#define DEFAULT_SCA_FREQUENCY 67000.0f
#define DEFAULT_SCA_DEVIATION 7000.0f
#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
@@ -28,7 +28,7 @@
#define SAMPLE_RATE 192000
#define INPUT_DEVICE "FM_Audio.monitor"
#define OUTPUT_DEVICE "plughw:1,0"
#define OUTPUT_DEVICE "alsa_output.pci-0000_00_14.2.analog-stereo"
#define MPX_DEVICE "FM_MPX.monitor"
// #define SCA_DEVICE ""
@@ -470,16 +470,16 @@ int main(int argc, char **argv) {
init_preemphasis(&preemp_l, preemphasis_tau, SAMPLE_RATE);
init_preemphasis(&preemp_r, preemphasis_tau, SAMPLE_RATE);
FrequencyFilter lpf_l, lpf_r;
init_lpf(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_lpf(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
BiquadFilter lpf_l, lpf_r;
init_lpf(&lpf_l, LPF_CUTOFF, 1.0f, SAMPLE_RATE);
init_lpf(&lpf_r, LPF_CUTOFF, 1.0f, SAMPLE_RATE);
FrequencyFilter hpf_l, hpf_r;
init_hpf(&hpf_l, HPF_CUTOFF, SAMPLE_RATE);
init_hpf(&hpf_r, HPF_CUTOFF, SAMPLE_RATE);
BiquadFilter hpf_l, hpf_r;
init_hpf(&hpf_l, HPF_CUTOFF, 1.0f, SAMPLE_RATE);
init_hpf(&hpf_r, HPF_CUTOFF, 1.0f, SAMPLE_RATE);
FrequencyFilter bass_hpf;
init_hpf(&bass_hpf, CENTER_BASS, SAMPLE_RATE);
BiquadFilter bass_hpf;
init_hpf(&bass_hpf, CENTER_BASS, 1.0f, SAMPLE_RATE);
// #endregion
signal(SIGINT, stop);