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add soft clipping

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This commit is contained in:
kuba
2025-01-27 10:51:51 +01:00
parent 6af57685fd
commit 59ec78948a
4 changed files with 73 additions and 41 deletions
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.vscode/.server-controller-port.log
Vendored
+1 -1
View File
@@ -1,5 +1,5 @@
{ {
"port": 13452, "port": 13452,
"time": 1737920538121, "time": 1737970082488,
"version": "0.0.3" "version": "0.0.3"
} }
lib/filters.c
+19 -1
View File
@@ -74,7 +74,7 @@ void init_hpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate) {
filter->index = 0; filter->index = 0;
} }
float apply_freqeuncy_filter(FrequencyFilter* filter, float input) { float apply_frequency_filter(FrequencyFilter* filter, float input) {
// Shift delay line // Shift delay line
filter->delay[filter->index] = input; filter->delay[filter->index] = input;
@@ -92,6 +92,24 @@ float apply_freqeuncy_filter(FrequencyFilter* filter, float input) {
return output; return output;
} }
float hard_clip(float sample, float threshold) {
if (sample > threshold) {
return threshold; // Clip to the upper threshold
} else if (sample < -threshold) {
return -threshold; // Clip to the lower threshold
} else {
return sample; // No clipping
}
}
float soft_clip(float sample, float threshold) {
if (fabs(sample) <= threshold) {
return sample; // Linear region
} else {
float sign = (sample > 0) ? 1.0f : -1.0f;
return sign * (threshold + (1.0f - threshold) * pow(fabs(sample) - threshold, 0.5f));
}
}
void init_delay_line(DelayLine *delay_line, int max_delay) { void init_delay_line(DelayLine *delay_line, int max_delay) {
delay_line->buffer = (float *)calloc(max_delay, sizeof(float)); delay_line->buffer = (float *)calloc(max_delay, sizeof(float));
delay_line->size = max_delay; delay_line->size = max_delay;
lib/filters.h
+5 -1
View File
@@ -21,7 +21,11 @@ typedef struct {
} FrequencyFilter; } FrequencyFilter;
void init_lpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate); void init_lpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate);
float apply_freqeuncy_filter(FrequencyFilter* filter, float input); void init_hpf(FrequencyFilter* filter, float cutoffFreq, float sampleRate);
float apply_frequency_filter(FrequencyFilter* filter, float input);
float hard_clip(float sample, float threshold);
float soft_clip(float sample, float threshold);
typedef struct { typedef struct {
float *buffer; float *buffer;
src/fm95.c
+48 -38
View File
@@ -10,6 +10,7 @@
#define DEFAULT_STEREO_POLAR 0 #define DEFAULT_STEREO_POLAR 0
#define DEFAULT_STEREO_SSB 0 #define DEFAULT_STEREO_SSB 0
#define DEFAULT_CLIPPER_THRESHOLD 1.0f #define DEFAULT_CLIPPER_THRESHOLD 1.0f
#define DEFAULT_SOFT_CLIPPER_THRESHOLD 0.95f
#define DEFAULT_ALSA_OUTPUT 0 #define DEFAULT_ALSA_OUTPUT 0
#define DEFAULT_SCA_FREQUENCY 67000.0f #define DEFAULT_SCA_FREQUENCY 67000.0f
#define DEFAULT_SCA_DEVIATION 7000.0f #define DEFAULT_SCA_DEVIATION 7000.0f
@@ -43,20 +44,11 @@
#define SCA_VOLUME 0.1f #define SCA_VOLUME 0.1f
#define MPX_VOLUME 1.0f #define MPX_VOLUME 1.0f
#define LPF_CUTOFF 15000 #define LPF_CUTOFF 15000 // Should't need to be changed
#define HPF_CUTOFF 30 // Unless you wanna have SOME bass then leave this alone
volatile sig_atomic_t to_run = 1; volatile sig_atomic_t to_run = 1;
float hard_clip(float sample, float threshold) {
if (sample > threshold) {
return threshold; // Clip to the upper threshold
} else if (sample < -threshold) {
return -threshold; // Clip to the lower threshold
} else {
return sample; // No clipping
}
}
void uninterleave(const float *input, float *left, float *right, size_t num_samples) { void uninterleave(const float *input, float *left, float *right, size_t num_samples) {
// For stereo, usually it is like this: LEFT RIGHT LEFT RIGHT LEFT RIGHT so this is used to get LEFT LEFT LEFT and RIGHT RIGHT RIGHT // For stereo, usually it is like this: LEFT RIGHT LEFT RIGHT LEFT RIGHT so this is used to get LEFT LEFT LEFT and RIGHT RIGHT RIGHT
for (size_t i = 0; i < num_samples/2; i++) { for (size_t i = 0; i < num_samples/2; i++) {
@@ -86,15 +78,16 @@ void show_help(char *name) {
" -p,--pulse_out Force pulse output [default: %d]\n" " -p,--pulse_out Force pulse output [default: %d]\n"
" -M,--mpx Override MPX input device [default: %s]\n" " -M,--mpx Override MPX input device [default: %s]\n"
" -C,--sca Override the SCA input device [default: %s]\n" " -C,--sca Override the SCA input device [default: %s]\n"
" -f,--sca_freq Override the SCA frequency [default: %f]\n" " -f,--sca_freq Override the SCA frequency [default: %.1f]\n"
" -F,--sca_dev Override the SCA deviation [default: %f]\n" " -F,--sca_dev Override the SCA deviation [default: %.2f]\n"
" -L,--sca_clip Override the SCA clipper threshold [default: %f]\n" " -L,--sca_clip Override the SCA clipper threshold [default: %.2f]\n"
" -c,--clipper Override the clipper threshold [default: %f]\n" " -c,--clipper Override the clipper threshold [default: %.2f]\n"
" -l,--soft_clip Override the soft clipper threshold [default: %.2f]\n"
" -P,--polar Force Polar Stereo (does not take effect with -m%s)\n" " -P,--polar Force Polar Stereo (does not take effect with -m%s)\n"
" -g,--ge Force Zenith/GE stereo (does not take effect with -m%s)\n" " -g,--ge Force Zenith/GE stereo (does not take effect with -m%s)\n"
" -S,--ssb Force SSB [default: %d]\n" " -S,--ssb Force SSB [default: %d]\n"
" -D,--dsb Force DSB [default: %d]\n" " -D,--dsb Force DSB [default: %d]\n"
" -R,--preemp Override preemphasis [default: %f]\n" " -R,--preemp Override preemphasis [default: %.2f µs]\n"
,name ,name
,DEFAULT_STEREO^1 ,DEFAULT_STEREO^1
,DEFAULT_STEREO ,DEFAULT_STEREO
@@ -116,11 +109,12 @@ void show_help(char *name) {
,DEFAULT_SCA_DEVIATION ,DEFAULT_SCA_DEVIATION
,DEFAULT_SCA_CLIPPER_THRESHOLD ,DEFAULT_SCA_CLIPPER_THRESHOLD
,DEFAULT_CLIPPER_THRESHOLD ,DEFAULT_CLIPPER_THRESHOLD
,DEFAULT_SOFT_CLIPPER_THRESHOLD
,(DEFAULT_STEREO_POLAR == 1) ? ", default" : "" ,(DEFAULT_STEREO_POLAR == 1) ? ", default" : ""
,(DEFAULT_STEREO_POLAR == 1) ? "" : ", default" ,(DEFAULT_STEREO_POLAR == 1) ? "" : ", default"
,DEFAULT_STEREO_SSB ,DEFAULT_STEREO_SSB
,DEFAULT_STEREO_SSB^1 ,DEFAULT_STEREO_SSB^1
,DEFAULT_PREEMPHASIS_TAU ,DEFAULT_PREEMPHASIS_TAU/0.000001
); );
} }
@@ -134,6 +128,7 @@ int main(int argc, char **argv) {
snd_pcm_t *output_handle; snd_pcm_t *output_handle;
float clipper_threshold = DEFAULT_CLIPPER_THRESHOLD; float clipper_threshold = DEFAULT_CLIPPER_THRESHOLD;
float soft_clipper_threshold = DEFAULT_SOFT_CLIPPER_THRESHOLD;
int stereo = DEFAULT_STEREO; int stereo = DEFAULT_STEREO;
int polar_stereo = DEFAULT_STEREO_POLAR; int polar_stereo = DEFAULT_STEREO_POLAR;
int ssb = DEFAULT_STEREO_SSB; int ssb = DEFAULT_STEREO_SSB;
@@ -159,7 +154,7 @@ int main(int argc, char **argv) {
// #region Parse Arguments // #region Parse Arguments
int opt; int opt;
const char *short_opt = "msi:o:apM:C:f:F:L:c:PgSDR:hv"; const char *short_opt = "msi:o:apM:C:f:F:L:c:l:PgSDR:hv";
struct option long_opt[] = struct option long_opt[] =
{ {
{"mono", no_argument, NULL, 'm'}, {"mono", no_argument, NULL, 'm'},
@@ -174,6 +169,7 @@ int main(int argc, char **argv) {
{"sca_dev", required_argument, NULL, 'F'}, {"sca_dev", required_argument, NULL, 'F'},
{"sca_clip", required_argument, NULL, 'L'}, {"sca_clip", required_argument, NULL, 'L'},
{"clipper", required_argument, NULL, 'c'}, {"clipper", required_argument, NULL, 'c'},
{"soft_clip", required_argument, NULL, 'l'},
{"polar", no_argument, NULL, 'P'}, {"polar", no_argument, NULL, 'P'},
{"ge", no_argument, NULL, 'g'}, {"ge", no_argument, NULL, 'g'},
{"ssb", no_argument, NULL, 'S'}, {"ssb", no_argument, NULL, 'S'},
@@ -182,7 +178,7 @@ int main(int argc, char **argv) {
{"help", no_argument, NULL, 'h'}, {"help", no_argument, NULL, 'h'},
{"version", no_argument, NULL, 'v'}, {"version", no_argument, NULL, 'v'},
{0, 0, 0, 0} // No trailing comma here {0, 0, 0, 0}
}; };
while((opt = getopt_long(argc, argv, short_opt, long_opt, NULL)) != -1) { while((opt = getopt_long(argc, argv, short_opt, long_opt, NULL)) != -1) {
@@ -217,19 +213,23 @@ int main(int argc, char **argv) {
break; break;
case 'f': //SCA freq case 'f': //SCA freq
sca_frequency = strtof(optarg, NULL); sca_frequency = strtof(optarg, NULL);
printf("Running with a SCA frequency of %f\n", sca_frequency); printf("Running with a SCA frequency of %.2f\n", sca_frequency);
break; break;
case 'F': //SCA deviation case 'F': //SCA deviation
sca_deviation = strtof(optarg, NULL); sca_deviation = strtof(optarg, NULL);
printf("Running with a SCA deviation of %f\n", sca_deviation); printf("Running with a SCA deviation of %.1f\n", sca_deviation);
break; break;
case 'L': //SCA clip case 'L': //SCA clip
sca_clipper_threshold = strtof(optarg, NULL); sca_clipper_threshold = strtof(optarg, NULL);
printf("Running with a SCA clipper threshold of %f\n", sca_clipper_threshold); printf("Running with a SCA clipper threshold of %.3f\n", sca_clipper_threshold);
break; break;
case 'c': //Clipper case 'c': //Clipper
clipper_threshold = strtof(optarg, NULL); clipper_threshold = strtof(optarg, NULL);
printf("Running with a clipper threshold of %f\n", clipper_threshold); printf("Running with a clipper threshold of %.3f\n", clipper_threshold);
break;
case 'l': //Soft Clipper
soft_clipper_threshold = strtof(optarg, NULL);
printf("Running with a soft clipper threshold of %.3f\n", soft_clipper_threshold);
break; break;
case 'P': //Polar case 'P': //Polar
polar_stereo = 1; polar_stereo = 1;
@@ -249,7 +249,7 @@ int main(int argc, char **argv) {
break; break;
case 'R': // Preemp case 'R': // Preemp
preemphasis_tau = strtof(optarg, NULL)*0.000001; preemphasis_tau = strtof(optarg, NULL)*0.000001;
printf("Running with a premp of %f\n", preemphasis_tau); printf("Running with a premp of %1.f µs\n", preemphasis_tau/0.000001);
break; break;
case 'v': // Version case 'v': // Version
show_version(); show_version();
@@ -398,11 +398,12 @@ int main(int argc, char **argv) {
} }
// #endregion // #endregion
Oscillator pilot_osc; // #region Setup Filters/Modulaltors/Oscillators
Oscillator osc;
if(polar_stereo == 1) { if(polar_stereo == 1) {
init_oscillator(&pilot_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 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 { } else {
init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier init_oscillator(&osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
} }
FMModulator sca_mod; FMModulator sca_mod;
@@ -410,7 +411,7 @@ int main(int argc, char **argv) {
HilbertTransformer hilbert; // An Hilbert shifts a signal in quadrature, generating the I/Q data HilbertTransformer hilbert; // An Hilbert shifts a signal in quadrature, generating the I/Q data
init_hilbert(&hilbert); init_hilbert(&hilbert);
DelayLine monoDelay; // Hilbert introduces a delay of 99 samples, this should be here to sync the mono with stereo to a sample DelayLine monoDelay; // Hilbert introduces a delay, this should be here to sync the mono with stereo to a sample
init_delay_line(&monoDelay, (HILBERT_TAPS-1)/2); init_delay_line(&monoDelay, (HILBERT_TAPS-1)/2);
ResistorCapacitor preemp_l, preemp_r; ResistorCapacitor preemp_l, preemp_r;
@@ -421,6 +422,11 @@ int main(int argc, char **argv) {
init_lpf(&lpf_l, LPF_CUTOFF, SAMPLE_RATE); init_lpf(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_lpf(&lpf_r, LPF_CUTOFF, SAMPLE_RATE); init_lpf(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
FrequencyFilter hpf_l, hpf_r;
init_hpf(&hpf_l, HPF_CUTOFF, SAMPLE_RATE);
init_hpf(&hpf_r, HPF_CUTOFF, SAMPLE_RATE);
// #endregion
signal(SIGINT, stop); signal(SIGINT, stop);
signal(SIGTERM, stop); signal(SIGTERM, stop);
@@ -459,10 +465,14 @@ int main(int argc, char **argv) {
float current_mpx_in = mpx_in[i]; float current_mpx_in = mpx_in[i];
float current_sca_in = sca_in[i]; float current_sca_in = sca_in[i];
float ready_l = apply_freqeuncy_filter(&lpf_l, r_in); float ready_l = apply_frequency_filter(&lpf_l, r_in);
float ready_r = apply_freqeuncy_filter(&lpf_r, l_in); float ready_r = apply_frequency_filter(&lpf_r, l_in);
ready_l = apply_frequency_filter(&hpf_l, ready_l);
ready_r = apply_frequency_filter(&hpf_r, ready_r);
ready_l = apply_pre_emphasis(&preemp_l, ready_l)*2; ready_l = apply_pre_emphasis(&preemp_l, ready_l)*2;
ready_r = apply_pre_emphasis(&preemp_r, ready_r)*2; ready_r = apply_pre_emphasis(&preemp_r, ready_r)*2;
ready_l = soft_clip(ready_l, soft_clipper_threshold);
ready_r = soft_clip(ready_r, soft_clipper_threshold);
ready_l = hard_clip(ready_l, clipper_threshold); ready_l = hard_clip(ready_l, clipper_threshold);
ready_r = hard_clip(ready_r, clipper_threshold); ready_r = hard_clip(ready_r, clipper_threshold);
@@ -471,8 +481,8 @@ int main(int argc, char **argv) {
float stereo = (ready_l - ready_r) / 2.0f; // Also Stereo to Mono but a bit diffrent float stereo = (ready_l - ready_r) / 2.0f; // Also Stereo to Mono but a bit diffrent
if(polar_stereo == 1) { if(polar_stereo == 1) {
if(ssb) { if(ssb) {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc, 1); // Get stereo carrier via multiplication float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 1); // Get stereo carrier via multiplication
float stereo_carrier_cos = get_oscillator_cos_sample(&pilot_osc); // Get Carrier Q of I/Q 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; stereo += 0.2;
@@ -482,7 +492,7 @@ int main(int argc, char **argv) {
if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_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(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
} else { } else {
float stereo_carrier = get_oscillator_sin_sample(&pilot_osc); float stereo_carrier = get_oscillator_sin_sample(&osc);
output[i] = mono*MONO_VOLUME + output[i] = mono*MONO_VOLUME +
((stereo+0.2)*stereo_carrier)*STEREO_VOLUME; ((stereo+0.2)*stereo_carrier)*STEREO_VOLUME;
if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME; if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
@@ -490,9 +500,9 @@ int main(int argc, char **argv) {
} }
} else { } else {
if(ssb) { if(ssb) {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc, 2); // Get stereo carrier via multiplication float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 2); // Get stereo carrier via multiplication
float stereo_carrier_cos = get_oscillator_cos_multiplier_ni(&pilot_osc, 2); // Get Carrier Q of I/Q float stereo_carrier_cos = get_oscillator_cos_multiplier_ni(&osc, 2); // Get Carrier Q of I/Q
float pilot = get_oscillator_sin_sample(&pilot_osc); float pilot = get_oscillator_sin_sample(&osc);
float stereo_i, stereo_q; float stereo_i, stereo_q;
apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // Compute I/Q apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // Compute I/Q
@@ -503,8 +513,8 @@ int main(int argc, char **argv) {
if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_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(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
} else { } else {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc,2); float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc,2);
float pilot = get_oscillator_sin_sample(&pilot_osc); float pilot = get_oscillator_sin_sample(&osc);
output[i] = mono*MONO_VOLUME + output[i] = mono*MONO_VOLUME +
pilot*PILOT_VOLUME + pilot*PILOT_VOLUME +
(stereo*stereo_carrier)*STEREO_VOLUME; (stereo*stereo_carrier)*STEREO_VOLUME;