radio95/fm95
0
1
Fork 0
mirror of https://github.com/radio95-rnt/fm95.git synced 2026-02-26 19:23:51 +01:00
Code Issues Activity

comments

Browse Source
This commit is contained in:
KubaPro010
2024-12-30 16:28:00 +01:00
parent 645873564f
commit 58c725ed8e
1 changed files with 10 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
stereo_coder.c
View File

@@ -28,12 +28,14 @@ float clip(float sample) {
}
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];
}
}
#define M_2PI (3.14159265358979323846 * 2.0)
// Track phase continuously to maintain frequency accuracy
@@ -65,14 +67,14 @@ static void stop(int signum) {
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 108 KHz, becuase it (53000*2) and (38000+15000)
const float PILOT_FREQ = 19000.0f;
const float STEREO_FREQ = 38000.0f;
const float PILOT_FREQ = 19000.0f; // Don't touch this
const float STEREO_FREQ = 38000.0f; // This too
// Define formats and buffer atributes
pa_sample_spec stereo_format = {
.format = PA_SAMPLE_FLOAT32NE,
.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
.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,
@@ -81,7 +83,7 @@ int main() {
};
pa_buffer_attr input_buffer_atr = {
.maxlength = 4096,
.maxlength = 4096, // You can lower this to 512, but this is fine, it's sub-second delay, you're probably not gonna notice unless you're looking for it
.fragsize = 2048
};
pa_buffer_attr output_buffer_atr = {
@@ -134,9 +136,9 @@ int main() {
signal(SIGINT, stop);
signal(SIGTERM, stop);
float input[BUFFER_SIZE*2]; // Input from device
float left[BUFFER_SIZE], right[BUFFER_SIZE]; // Audio
float mpx[BUFFER_SIZE]; // MPX
float input[BUFFER_SIZE*2]; // Input from device, interleaved stereo
float left[BUFFER_SIZE], right[BUFFER_SIZE]; // Audio, same thing as in input but ininterleaved
float mpx[BUFFER_SIZE]; // MPX, this goes to the output
while (to_run) {
if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) {
fprintf(stderr, "Error reading from input device.\n");