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mirror of https://github.com/radio95-rnt/fm95.git synced 2026-02-26 19:23:51 +01:00

add a version with rds

This commit is contained in:
2024-12-30 12:26:27 +01:00
parent d5673650a5
commit 9d06f8d80e
3 changed files with 210 additions and 10 deletions

1
compile_stereo_rds Executable file
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@@ -0,0 +1 @@
gcc stereo_coder_rds.c -lpulse -lpulse-simple -lm -o stereo_coder

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@@ -9,13 +9,12 @@
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
#define BUFFER_SIZE 512
#define MONO_VOLUME 0.5f
#define PILOT_VOLUME 0.025f
#define STEREO_VOLUME 0.275f
#define MONO_VOLUME 0.5f // L+R Signal
#define PILOT_VOLUME 0.025f // 19 KHz Pilot
#define STEREO_VOLUME 0.275f // L-R signal
#define TWO_BUFFER_SIZE (BUFFER_SIZE*2) // Don't touch this
volatile int to_run = 1;
volatile sig_atomic_t to_run = 1;
const float format_scale = 1.0f / 32768.0f;
void stereo_s16le_to_float(const int16_t *input, float *left, float *right, size_t num_samples) {
@@ -66,12 +65,12 @@ int main() {
const float STEREO_FREQ = 38000.0f;
// Define formats and buffer atributes
pa_sample_spec input_format = {
pa_sample_spec stereo_format = {
.format = PA_SAMPLE_S16LE,
.channels = 2,
.rate = SAMPLE_RATE
};
pa_sample_spec output_format = {
pa_sample_spec mono_format = {
.format = PA_SAMPLE_S16LE,
.channels = 1,
.rate = SAMPLE_RATE
@@ -95,7 +94,7 @@ int main() {
PA_STREAM_RECORD,
INPUT_DEVICE,
"Audio Input",
&input_format,
&stereo_format,
NULL,
&input_buffer_atr,
NULL
@@ -113,7 +112,7 @@ int main() {
PA_STREAM_PLAYBACK,
OUTPUT_DEVICE,
"MPX",
&output_format,
&mono_format,
NULL,
&output_buffer_atr,
NULL
@@ -148,7 +147,7 @@ int main() {
float mono = (left[i] + right[i]) / 2.0f;
float stereo = (left[i] - right[i]) / 2.0f;
mpx[i] = mono*MONO_VOLUME +
(stereo * stereo_carrier)*STEREO_VOLUME +
(pilot * PILOT_VOLUME);

200
stereo_coder_rds.c Normal file
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@@ -0,0 +1,200 @@
#include <stdio.h>
#include <pulse/simple.h>
#include <stdlib.h>
#include <math.h>
#include <stdint.h>
#include <signal.h>
#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
#define RDS_INPUT "RDS.monitor"
#define BUFFER_SIZE 512
#define MONO_VOLUME 0.5f // L+R Signal
#define PILOT_VOLUME 0.025f // 19 KHz Pilot
#define STEREO_VOLUME 0.275f // L-R signal
#define RDS_VOLUME 0.04f // RDS Signal
volatile sig_atomic_t to_run = 1;
const float format_scale = 1.0f / 32768.0f;
void stereo_s16le_to_float(const int16_t *input, float *left, float *right, size_t num_samples) {
for (size_t i = 0; i < num_samples/2; i++) {
left[i] = input[i * 2] * format_scale;
right[i] = input[i * 2 + 1] * format_scale;
}
}
void mono_s16le_to_float(const int16_t *input, float *output, size_t num_samples) {
for (size_t i = 0; i < num_samples; i++) {
output[i] = input[i * 2] * format_scale;
}
}
void float_array_to_s16le(const float *input, int16_t *output, size_t num_samples) {
for (size_t i = 0; i < num_samples; i++) {
output[i] = (int16_t)((fminf(fmaxf(input[i], -1.0f), 1.0f)) * 32767.0f);
}
}
#define M_2PI (3.14159265358979323846 * 2.0)
// Track phase continuously to maintain frequency accuracy
typedef struct {
float phase;
float phase_increment;
} Oscillator;
void init_oscillator(Oscillator *osc, float frequency, float sample_rate) {
osc->phase = 0.0f;
osc->phase_increment = (M_2PI * frequency) / sample_rate;
}
float get_next_sample(Oscillator *osc) {
float sample = sinf(osc->phase);
osc->phase += osc->phase_increment;
if (osc->phase >= M_2PI) {
osc->phase -= M_2PI;
}
return sample;
}
static void stop(int signum) {
(void)signum;
printf("\nReceived stop signal. Cleaning up...\n");
to_run = 0;
}
int main() {
printf("STCode : Stereo encoder made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
const float SAMPLE_RATE = 192000.0f; // Don't go lower than 176 KHz
const float PILOT_FREQ = 19000.0f;
const float STEREO_FREQ = 38000.0f;
const float RDS_FREQ = 57000.0f;
// Define formats and buffer atributes
pa_sample_spec stereo_format = {
.format = PA_SAMPLE_S16LE,
.channels = 2,
.rate = SAMPLE_RATE
};
pa_sample_spec mono_format = {
.format = PA_SAMPLE_S16LE,
.channels = 1,
.rate = SAMPLE_RATE
};
pa_buffer_attr input_buffer_atr = {
.maxlength = 4096,
.fragsize = 2048
};
pa_buffer_attr output_buffer_atr = {
.maxlength = 4096,
.tlength = 2048,
.prebuf = 0
};
printf("Connecting to input devices... (%s, %s)\n", INPUT_DEVICE, RDS_INPUT);
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;
}
pa_simple *input_device_rds = pa_simple_new(
NULL,
"StereoEncoder",
PA_STREAM_RECORD,
RDS_INPUT,
"Audio Input",
&mono_format,
NULL,
&input_buffer_atr,
NULL
);
if (!input_device_rds) {
fprintf(stderr, "Error: cannot open input device.\n");
pa_simple_free(input_device);
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);
pa_simple_free(input_device_rds);
return 1;
}
Oscillator pilot_osc, stereo_osc, rds_osc;
init_oscillator(&pilot_osc, PILOT_FREQ, SAMPLE_RATE);
init_oscillator(&stereo_osc, STEREO_FREQ, SAMPLE_RATE);
init_oscillator(&rds_osc, RDS_FREQ, SAMPLE_RATE);
signal(SIGINT, stop);
signal(SIGTERM, stop);
int16_t input[BUFFER_SIZE*2], input_rds[BUFFER_SIZE];
float rds[BUFFER_SIZE];
float left[BUFFER_SIZE], right[BUFFER_SIZE];
float mpx[BUFFER_SIZE];
int16_t output[BUFFER_SIZE];
while (to_run) {
if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) {
fprintf(stderr, "Error reading from input device.\n");
break;
}
if (pa_simple_read(input_device_rds, input_rds, sizeof(input_rds), NULL) < 0) {
fprintf(stderr, "Error reading from input device.\n");
break;
}
stereo_s16le_to_float(input, left, right, sizeof(input));
mono_s16le_to_float(input_rds, rds, sizeof(input_rds));
for (int i = 0; i < BUFFER_SIZE; i++) {
float pilot = get_next_sample(&pilot_osc);
float stereo_carrier = get_next_sample(&stereo_osc);
float rds_carrier = get_next_sample(&rds_osc);
float mono = (left[i] + right[i]) / 2.0f;
float stereo = (left[i] - right[i]) / 2.0f;
float rds_sample = rds[i];
mpx[i] = mono*MONO_VOLUME +
(stereo * stereo_carrier)*STEREO_VOLUME +
(pilot * PILOT_VOLUME) +
(rds_sample * rds_carrier)*RDS_VOLUME;
}
float_array_to_s16le(mpx, output, BUFFER_SIZE);
if (pa_simple_write(output_device, output, sizeof(output), NULL) < 0) {
fprintf(stderr, "Error writing to output device.\n");
break;
}
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
pa_simple_free(output_device);
return 0;
}