radio95/fm95
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KubaPro010
2025-01-23 10:03:39 +01:00
parent 43521d90a8
commit 7096a6e572
13 changed files with 106 additions and 300 deletions
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2
.vscode/.server-controller-port.log vendored
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@@ -1,5 +1,5 @@
{
"port": 13452,
"time": 1737492785115,
"time": 1737621156918,
"version": "0.0.3"
}

3
.vscode/settings.json vendored
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@@ -11,6 +11,7 @@
"math.h": "c",
"options.h": "c",
"random": "c",
"__locale": "c"
"__locale": "c",
"complex": "c"
}
}

21
README.md
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@@ -7,40 +7,25 @@ STCode is a simple stereo encoder for FM, it uses pasimple and math to:
- Generate the stereo pilot in phase to the stereo subcarrier
- Generate the stereo diffrence signal using DSB-SC ((L-R)/2)
All that in about 3.5% cpu usage on a RPI-5 (lpf makes it 10, but stereo tool has 3 threads which do 100% cpu usage anyway, one 200)!
All that in about 3.5% cpu usage on a RPI-5 (stereo tool has 3 threads which do 100% cpu usage anyway, one 200)
Also nearly no latency, not like Stereo Tool (or mpxgen which doesn't even work)
As far as i've tested it (29-31 december) it's been fine but after a fix it was great, so i'd redecommend you
As far as i've tested it (29-31 december) it's been fine but after a fix it was great, so i'd recommend this to you
Also i'd recommend to use the SSB version because it's more spectrum effiecent
# SSB-STCode
This is a version of the stereo code but instead of DSB-SC it transmits some kind of VSG (mostly USB with a bit of LSB), about 600 hz of usb is left, just as god intended (Hilbert isn't perfect so i got some usb but managed to turn it into more into LSB)
This also has a cpu usage of 20% with lpf, but goes to 13-15% without the lpf
but SSB has slightly more cpu usage
# PSTCode
This is a yet another version of a Stereo encoder, however for the OIRT band which is in use in Russia, Belarus and other countries
Haven't tested it nor plan to
# CrosbySTCode
This is a stereo coder however with a diffrent system, let me yap some:
In the 1950-1960s the FCC had to decide between two stereo coding systems, we had the Zenith/GE system and the Crosby system, what was the diffrence?
The Zenith system had a 19 khz pilot and a 38 khz dsb-sc modulated stereo l-r signal, sounds familliar? yeah that's why you haven't heard of the crosby system
The crosby system on the other hand had a (better) decision of modulation the l-r signal into 50 khz with FM, why was it rejected? becuase of SCA, 67 and 41 khz were used up by stereo, 41 khz was also used up on the Zenith system but who cares
Also it doesnsn't sound bad, how may ask where did i find a decoder for it? Made it myself in GNU radio, it's even easier, if they chose 35 khz instead of 50, then we'd be using this, also i like this one because it has FM, not AM so if some idiot has a transmitter with hardware pre-emphasis then the stereo won't be affected by it
# SCAMod
SCAMod is a simple FM modulator which can be used to modulate a secondary audio stream, has similiar cpu usage and latency as STCode
Has a fine quality, but as it goes for 12 khz fm signals
# StereoSCAMod
Stereo SCA, like normal SCA but encodes L-R onto 80 khz, only demodulator of this right now is gnu radio
# MonoPass
want to keep mono for a reason but have the lpf and preemphasis, do so

14
lib/filters.c
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@@ -1,6 +1,11 @@
#include "filters.h"
void init_rc(ResistorCapacitor *rc, float tau, float sample_rate) {
void init_rc(ResistorCapacitor *rc, float alpha) {
rc->prev_sample = 0.0f;
rc->alpha = alpha;
}
void init_rc_tau(ResistorCapacitor *rc, float tau, float sample_rate) {
rc->prev_sample = 0.0f;
rc->alpha = exp(-1 / (tau * sample_rate));
}
@@ -8,15 +13,16 @@ void init_rc(ResistorCapacitor *rc, float tau, float sample_rate) {
float apply_pre_emphasis(ResistorCapacitor *rc, float sample) {
float audio = sample-rc->alpha*rc->prev_sample;
rc->prev_sample = audio;
return audio*2;
return audio;
}
void init_low_pass_filter(ResistorCapacitor *rc, float cutoff_frequency, float sample_rate) {
init_rc(&rc, (-1 / (sample_rate * log(sample_rate/(sample_rate+(1/(M_2PI*cutoff_frequency)))))), sample_rate);
float RC = 1.0f / (M_2PI * cutoff_frequency);
init_rc(&rc, 1.0f / (1.0f + sample_rate * RC));
}
float apply_low_pass_filter(ResistorCapacitor *rc, float sample) {
float output = rc->alpha*sample+(1-rc->alpha)*rc->prev_sample;
float output = rc->alpha*sample+(1.0f-rc->alpha)*rc->prev_sample;
rc->prev_sample = output;
return output;
}

7
lib/filters.h
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@@ -5,21 +5,18 @@
#include <stdlib.h>
#include "constants.h"
#define FIR_PHASES 32
#define FIR_TAPS 32
typedef struct {
float alpha;
float prev_sample;
} ResistorCapacitor;
void init_rc(ResistorCapacitor *pe, float tau, float sample_rate);
void init_rc(ResistorCapacitor *pe, float alpha);
void init_rc_tau(ResistorCapacitor *pe, float tau, float sample_rate);
float apply_pre_emphasis(ResistorCapacitor *pe, float sample);
void init_low_pass_filter(ResistorCapacitor *lp, float cutoff_frequency, float sample_rate);
float apply_low_pass_filter(ResistorCapacitor *lp, float sample);
typedef struct {
float *buffer;
int write_idx; // Write position

6
src/crosby_stereo_coder.c → satire/crosby_stereo_coder.c
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@@ -123,11 +123,11 @@ int main() {
init_oscillator(&osc, 50000.0, SAMPLE_RATE);
#ifdef PREEMPHASIS
ResistorCapacitor preemp_l, preemp_r;
init_rc(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf_l, lpf_r;
ResistorCapacitor lpf_l, lpf_r;
init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
#endif

6
src/stereo_sca_mod.c → satire/stereo_sca_mod.c
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@@ -125,11 +125,11 @@ int main() {
init_fm_modulator(&mod_stereo, 80000, 6000, SAMPLE_RATE);
#ifdef PREEMPHASIS
ResistorCapacitor preemp_l, preemp_r;
init_rc(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf_l, lpf_r;
ResistorCapacitor lpf_l, lpf_r;
init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
#endif

14
src/mono_passthrough.c
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@@ -31,7 +31,7 @@
volatile sig_atomic_t to_run = 1;
float clip(float sample) {
float hard_clip(float sample) {
if (sample > CLIPPER_THRESHOLD) {
return CLIPPER_THRESHOLD; // Clip to the upper threshold
} else if (sample < -CLIPPER_THRESHOLD) {
@@ -105,10 +105,10 @@ int main() {
#ifdef PREEMPHASIS
ResistorCapacitor preemp;
init_rc(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf;
ResistorCapacitor lpf;
init_low_pass_filter(&lpf, LPF_CUTOFF, SAMPLE_RATE);
#endif
@@ -132,17 +132,17 @@ int main() {
#ifdef LPF
float lowpassed = apply_low_pass_filter(&lpf, in);
float preemphasized = apply_pre_emphasis(&preemp, lowpassed);
float current_input = clip(preemphasized);
float current_input = hard_clip(preemphasized);
#else
float preemphasized = apply_pre_emphasis(&preemp, in);
float current_input = clip(preemphasized);
float current_input = hard_clip(preemphasized);
#endif
#else
#ifdef LPF
float lowpassed = apply_low_pass_filter(&lpf, in);
float current_input = clip(lowpassed);
float current_input = hard_clip(lowpassed);
#else
float current_input = clip(in);
float current_input = hard_clip(in);
#endif
#endif

2
src/options.h
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@@ -1,5 +1,5 @@
#define PREEMPHASIS
// #define LPF
#define LPF
#define buffer_maxlength 12288
#define buffer_tlength_fragsize 8192

28
src/polar_stereo_coder.c
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@@ -33,7 +33,7 @@
volatile sig_atomic_t to_run = 1;
float clip(float sample) {
float hard_clip(float sample) {
if (sample > CLIPPER_THRESHOLD) {
return CLIPPER_THRESHOLD; // Clip to the upper threshold
} else if (sample < -CLIPPER_THRESHOLD) {
@@ -122,11 +122,11 @@ int main() {
init_oscillator(&stereo_osc, 31250.0, SAMPLE_RATE);
#ifdef PREEMPHASIS
ResistorCapacitor preemp_l, preemp_r;
init_rc(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf_l, lpf_r;
ResistorCapacitor lpf_l, lpf_r;
init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
#endif
@@ -156,23 +156,23 @@ int main() {
float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left);
float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right);
float current_left_input = clip(preemphasized_left);
float current_right_input = clip(preemphasized_right);
float current_left_input = hard_clip(preemphasized_left);
float current_right_input = hard_clip(preemphasized_right);
#else
float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in);
float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in);
float current_left_input = clip(preemphasized_left);
float current_right_input = clip(preemphasized_right);
float current_left_input = hard_clip(preemphasized_left);
float current_right_input = hard_clip(preemphasized_right);
#endif
#else
#ifdef LPF
float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
float current_left_input = clip(lowpassed_left);
float current_right_input = clip(lowpassed_right);
float current_left_input = hard_clip(lowpassed_left);
float current_right_input = hard_clip(lowpassed_right);
#else
float current_left_input = clip(l_in);
float current_right_input = clip(r_in);
float current_left_input = hard_clip(l_in);
float current_right_input = hard_clip(r_in);
#endif
#endif
@@ -180,10 +180,10 @@ int main() {
float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent
float stereo_carrier = get_oscillator_sin_sample(&stereo_osc);
// 14 db is somewhere around 20% of a 1 volt signal
// -14 db is somewhere around 20% of a 1 volt signal
mpx[i] = mono * MONO_VOLUME +
((stereo+0.2) * stereo_carrier)*STEREO_VOLUME;
((stereo+0.2) * stereo_carrier)*STEREO_VOLUME; // the 0.2 add DC, you know what happens then? Carrier wave
}
if (pa_simple_write(output_device, mpx, sizeof(mpx), &pulse_error) < 0) {

14
src/sca_mod.c
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@@ -36,7 +36,7 @@
volatile sig_atomic_t to_run = 1;
float clip(float sample) {
float hard_clip(float sample) {
if (sample > CLIPPER_THRESHOLD) {
return CLIPPER_THRESHOLD; // Clip to the upper threshold
} else if (sample < -CLIPPER_THRESHOLD) {
@@ -113,10 +113,10 @@ int main() {
init_fm_modulator(&mod, FREQUENCY, DEVIATION, SAMPLE_RATE);
#ifdef PREEMPHASIS
ResistorCapacitor preemp;
init_rc(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf;
ResistorCapacitor lpf;
init_low_pass_filter(&lpf, LPF_CUTOFF, SAMPLE_RATE);
#endif
@@ -140,17 +140,17 @@ int main() {
#ifdef LPF
float lowpassed = apply_low_pass_filter(&lpf, in);
float preemphasized = apply_pre_emphasis(&preemp, lowpassed);
float current_input = clip(preemphasized);
float current_input = hard_clip(preemphasized);
#else
float preemphasized = apply_pre_emphasis(&preemp, in);
float current_input = clip(preemphasized);
float current_input = hard_clip(preemphasized);
#endif
#else
#ifdef LPF
float lowpassed = apply_low_pass_filter(&lpf, in);
float current_input = clip(lowpassed);
float current_input = hard_clip(lowpassed);
#else
float current_input = clip(in);
float current_input = hard_clip(in);
#endif
#endif

216
src/ssb_stereo_coder.c
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@@ -1,216 +0,0 @@
#include <stdio.h>
#include <pulse/simple.h>
#include <pulse/error.h>
#include <stdlib.h>
#include <math.h>
#include <stdint.h>
#include <signal.h>
#include <string.h>
#include "../lib/constants.h"
#include "../lib/oscillator.h"
#include "../lib/filters.h"
#include "../lib/hilbert.h"
#include "options.h"
//#define USB
#define SAMPLE_RATE 192000 // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000)
#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
#define BUFFER_SIZE 512
#define CLIPPER_THRESHOLD 0.525 // Adjust this as needed
#define MONO_VOLUME 0.45f // L+R Signal
#define PILOT_VOLUME 0.09f // 19 KHz Pilot
#define STEREO_VOLUME 0.45f // L-R signal possibly can be set to .9 because im not sure if usb will be 2 times stronger than dsb-sc
#ifdef PREEMPHASIS
#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america
#endif
#ifdef LPF
#define LPF_CUTOFF 15000
#endif
volatile sig_atomic_t to_run = 1;
float clip(float sample) {
if (sample > CLIPPER_THRESHOLD) {
return CLIPPER_THRESHOLD; // Clip to the upper threshold
} else if (sample < -CLIPPER_THRESHOLD) {
return -CLIPPER_THRESHOLD; // Clip to the lower threshold
} else {
return sample; // No clipping
}
}
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 (size_t i = 0; i < num_samples/2; i++) {
left[i] = input[i * 2];
right[i] = input[i * 2 + 1];
}
}
static void stop(int signum) {
(void)signum;
printf("\nReceived stop signal. Cleaning up...\n");
to_run = 0;
}
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_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_FLOAT32NE,
.channels = 1,
.rate = SAMPLE_RATE
};
pa_buffer_attr input_buffer_atr = {
.maxlength = buffer_maxlength,
.fragsize = buffer_tlength_fragsize
};
pa_buffer_attr output_buffer_atr = {
.maxlength = buffer_maxlength,
.tlength = buffer_tlength_fragsize,
.prebuf = buffer_prebuf
};
printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
pa_simple *input_device = pa_simple_new(
NULL,
"StereoEncoder",
PA_STREAM_RECORD,
INPUT_DEVICE,
"Audio Input",
&stereo_format,
NULL,
&input_buffer_atr,
NULL
);
if (!input_device) {
fprintf(stderr, "Error: cannot open input device.\n");
return 1;
}
printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
pa_simple *output_device = pa_simple_new(
NULL,
"StereoEncoder",
PA_STREAM_PLAYBACK,
OUTPUT_DEVICE,
"MPX",
&mono_format,
NULL,
&output_buffer_atr,
NULL
);
if (!output_device) {
fprintf(stderr, "Error: cannot open output device.\n");
pa_simple_free(input_device);
return 1;
}
Oscillator pilot_osc;
init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
HilbertTransformer hilbert;
init_hilbert(&hilbert);
DelayLine monoDelay;
init_delay_line(&monoDelay, 99);
#ifdef PREEMPHASIS
ResistorCapacitor preemp_l, preemp_r;
init_rc(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf_l, lpf_r;
init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
#endif
signal(SIGINT, stop);
signal(SIGTERM, stop);
int pulse_error;
float input[BUFFER_SIZE*2]; // Input from device, interleaved stereo
float left[BUFFER_SIZE+64], right[BUFFER_SIZE+64]; // Audio, same thing as in input but ininterleaved, ai told be there could be a buffer overflow here
float mpx[BUFFER_SIZE]; // MPX, this goes to the output
while (to_run) {
if (pa_simple_read(input_device, input, sizeof(input), &pulse_error) < 0) {
fprintf(stderr, "Error reading from input device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
uninterleave(input, left, right, BUFFER_SIZE*2);
for (int i = 0; i < BUFFER_SIZE; i++) {
float sin38 = get_oscillator_sin_multiplier_ni(&pilot_osc, 2);
float cos38 = get_oscillator_cos_multiplier_ni(&pilot_osc, 2); // Stereo carrier should be a harmonic of the pilot which is in phase, best way to generate the harmonic is to multiply the pilot's phase by two, so it is mathematically impossible for them to not be in phase
float pilot = get_oscillator_sin_sample(&pilot_osc); // This is after because if it was before then the stereo would be out of phase by one increment, so [GET STEREO] ([GET PILOT] [INCREMENT PHASE])
float l_in = left[i];
float r_in = right[i];
#ifdef PREEMPHASIS
#ifdef LPF
float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left);
float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right);
float current_left_input = clip(preemphasized_left);
float current_right_input = clip(preemphasized_right);
#else
float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in);
float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in);
float current_left_input = clip(preemphasized_left);
float current_right_input = clip(preemphasized_right);
#endif
#else
#ifdef LPF
float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
float current_left_input = clip(lowpassed_left);
float current_right_input = clip(lowpassed_right);
#else
float current_left_input = clip(l_in);
float current_right_input = clip(r_in);
#endif
#endif
float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono
float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent
float stereo_i, stereo_q;
apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // I/Q, the Quadrature data is 90 degrees apart from the In-phase data
#ifdef USB
float signal = (stereo_i*cos38+stereo_q*(sin38*0.775f)); // Compute LSB/USB, as the Hilbert isn't perfect, i'll have to a bit silence down the Q carrier in order to make it better, also, it is just perfect as FM Stereo LSB shouldn't be fully LSB
#else
float signal = (stereo_i*cos38-stereo_q*(sin38*0.775f)); // Compute LSB/USB, as the Hilbert isn't perfect, i'll have to a bit silence down the Q carrier in order to make it better, also, it is just perfect as FM Stereo LSB shouldn't be fully LSB
#endif
mpx[i] = delay_line(&monoDelay, mono) * MONO_VOLUME +
pilot * PILOT_VOLUME +
signal*STEREO_VOLUME;
}
if (pa_simple_write(output_device, mpx, sizeof(mpx), &pulse_error) < 0) {
fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
pa_simple_free(output_device);
exit_hilbert(&hilbert);
return 0;
}

73
src/stereo_coder.c
View File

@@ -7,11 +7,17 @@
#include <signal.h>
#include <string.h>
#include "options.h"
//#define SSB
//#define USB
#include "../lib/constants.h"
#include "../lib/oscillator.h"
#include "../lib/filters.h"
#include "options.h"
#ifdef SSB
#include "../lib/hilbert.h"
#endif
#define SAMPLE_RATE 192000 // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000)
@@ -22,7 +28,7 @@
#define MONO_VOLUME 0.45f // L+R Signal
#define PILOT_VOLUME 0.09f // 19 KHz Pilot
#define STEREO_VOLUME 0.45f // L-R signal
#define STEREO_VOLUME 0.45f // L-R signal possibly can be set to .9 because im not sure if usb will be 2 times stronger than dsb-sc
#ifdef PREEMPHASIS
#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america
@@ -34,7 +40,7 @@
volatile sig_atomic_t to_run = 1;
float clip(float sample) {
float hard_clip(float sample) {
if (sample > CLIPPER_THRESHOLD) {
return CLIPPER_THRESHOLD; // Clip to the upper threshold
} else if (sample < -CLIPPER_THRESHOLD) {
@@ -107,7 +113,7 @@ int main() {
"StereoEncoder",
PA_STREAM_PLAYBACK,
OUTPUT_DEVICE,
"MPX",
"MPX Output",
&mono_format,
NULL,
&output_buffer_atr,
@@ -121,13 +127,20 @@ int main() {
Oscillator pilot_osc;
init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
#ifdef SSB
HilbertTransformer hilbert;
init_hilbert(&hilbert);
DelayLine monoDelay;
init_delay_line(&monoDelay, 99);
#endif
#ifdef PREEMPHASIS
ResistorCapacitor preemp_l, preemp_r;
init_rc(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
#ifdef LPF
LowPassFilter lpf_l, lpf_r;
ResistorCapacitor lpf_l, lpf_r;
init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
#endif
@@ -148,7 +161,10 @@ int main() {
uninterleave(input, left, right, BUFFER_SIZE*2);
for (int i = 0; i < BUFFER_SIZE; i++) {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc, 2); // Stereo carrier should be a harmonic of the pilot which is in phase, best way to generate the harmonic is to multiply the pilot's phase by two, so it is mathematically impossible for them to not be in phase
float sin38 = get_oscillator_sin_multiplier_ni(&pilot_osc, 2); // Stereo carrier should be a harmonic of the pilot which is in phase, best way to generate the harmonic is to multiply the pilot's phase by two, so it is mathematically impossible for them to not be in phase
#ifdef SSB
float cos38 = get_oscillator_cos_multiplier_ni(&pilot_osc, 2);
#endif
float pilot = get_oscillator_sin_sample(&pilot_osc); // This is after because if it was before then the stereo would be out of phase by one increment, so [GET STEREO] ([GET PILOT] [INCREMENT PHASE])
float l_in = left[i];
float r_in = right[i];
@@ -159,32 +175,45 @@ int main() {
float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left);
float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right);
float current_left_input = clip(preemphasized_left);
float current_right_input = clip(preemphasized_right);
float current_left_input = hard_clip(preemphasized_left);
float current_right_input = hard_clip(preemphasized_right);
#else
float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in);
float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in);
float current_left_input = clip(preemphasized_left);
float current_right_input = clip(preemphasized_right);
float current_left_input = hard_clip(preemphasized_left);
float current_right_input = hard_clip(preemphasized_right);
#endif
#else
#ifdef LPF
float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
float current_left_input = clip(lowpassed_left);
float current_right_input = clip(lowpassed_right);
float current_left_input = hard_clip(lowpassed_left);
float current_right_input = hard_clip(lowpassed_right);
#else
float current_left_input = clip(l_in);
float current_right_input = clip(r_in);
float current_left_input = hard_clip(l_in);
float current_right_input = hard_clip(r_in);
#endif
#endif
float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono
float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent
mpx[i] = mono * MONO_VOLUME +
#ifdef SSB
float stereo_i, stereo_q;
apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // I/Q, the Quadrature data is 90 degrees apart from the In-phase data
#ifdef USB
float signal = (stereo_i*cos38+stereo_q*(sin38*0.775f)); // Compute USB, as the Hilbert isn't perfect, i'll have to a bit silence down the Q carrier in order to make it better, also, it is just perfect as FM Stereo LSB shouldn't be fully LSB
#else
float signal = (stereo_i*cos38-stereo_q*(sin38*0.775f)); // Compute LSB
#endif
mpx[i] = delay_line(&monoDelay, mono) * MONO_VOLUME +
pilot * PILOT_VOLUME +
(stereo * stereo_carrier) * STEREO_VOLUME; // DSB-SC modulation
signal*STEREO_VOLUME;
#else
mpx[i] = mono*MONO_VOLUME +
pilot*PILOT_VOLUME +
(stereo*sin38)*STEREO_VOLUME;
#endif
}
if (pa_simple_write(output_device, mpx, sizeof(mpx), &pulse_error) < 0) {
@@ -196,5 +225,9 @@ int main() {
printf("Cleaning up...\n");
pa_simple_free(input_device);
pa_simple_free(output_device);
#ifdef SSB
exit_hilbert(&hilbert);
exit_delay_line(&monoDelay);
#endif
return 0;
}