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Code Issues Activity

add alsa output to pstcode, scamod

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This commit is contained in:
KubaPro010
2025-01-25 19:58:14 +01:00
parent efd83adf39
commit 634ba05d61
7 changed files with 89 additions and 448 deletions
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39
src/polar_stereo_coder.c
View File

@@ -17,6 +17,7 @@
#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
// #define ALSA_OUTPUT // Output, not input or both
#define BUFFER_SIZE 512
#define CLIPPER_THRESHOLD 0.525 // Adjust this as needed
@@ -75,11 +76,13 @@ int main() {
.maxlength = buffer_maxlength,
.fragsize = buffer_tlength_fragsize
};
#ifndef ALSA_OUTPUT
pa_buffer_attr output_buffer_atr = {
.maxlength = buffer_maxlength,
.tlength = buffer_tlength_fragsize,
.prebuf = buffer_prebuf
};
#endif
printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
@@ -101,6 +104,7 @@ int main() {
printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
#ifndef ALSA_OUTPUT
pa_simple *output_device = pa_simple_new(
NULL,
"PolarStereoEncoder",
@@ -117,6 +121,33 @@ int main() {
pa_simple_free(input_device);
return 1;
}
#else
snd_pcm_hw_params_t *output_params;
snd_pcm_t *output_handle;
int output_error = snd_pcm_open(&output_handle, OUTPUT_DEVICE, SND_PCM_STREAM_PLAYBACK, 0);
if(output_error < 0) {
fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
pa_simple_free(input_device);
return 1;
}
snd_pcm_hw_params_malloc(&output_params);
snd_pcm_hw_params_any(output_handle, output_params);
snd_pcm_hw_params_set_access(output_handle, output_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(output_handle, output_params, SND_PCM_FORMAT_FLOAT); // Same as pulse's Float32NE
snd_pcm_hw_params_set_channels(output_handle, output_params, 1);
unsigned int rate = SAMPLE_RATE;
int dir;
snd_pcm_hw_params_set_rate_near(output_handle, output_params, &rate, &dir);
snd_pcm_uframes_t frames = BUFFER_SIZE;
snd_pcm_hw_params_set_period_size_near(output_handle, output_params, &frames, &dir); // i don't have a clue why like this
output_error = snd_pcm_hw_params(output_handle, output_params);
if(output_error < 0) {
fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
snd_pcm_close(output_handle);
pa_simple_free(input_device);
return 1;
}
#endif
Oscillator stereo_osc;
init_oscillator(&stereo_osc, 31250.0, SAMPLE_RATE);
@@ -186,14 +217,22 @@ int main() {
((stereo+0.2) * stereo_carrier)*STEREO_VOLUME; // the 0.2 add DC, you know what happens then? Carrier wave
}
#ifndef ALSA_OUTPUT
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;
}
#else
snd_pcm_writei(output_handle, mpx, sizeof(mpx));
#endif
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
#ifndef ALSA_OUTPUT
pa_simple_free(output_device);
#else
snd_pcm_drain(output_handle);
snd_pcm_free(output_handle);
return 0;
}

49
src/sca_mod.c
View File

@@ -1,6 +1,4 @@
#include <stdio.h>
#include <pulse/simple.h>
#include <pulse/error.h>
#include <stdlib.h>
#include <math.h>
#include <stdint.h>
@@ -18,9 +16,16 @@
#define INPUT_DEVICE "SCA.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
// #define ALSA_OUTPUT // Output, not input or both
#define BUFFER_SIZE 512
#define CLIPPER_THRESHOLD 1 // Adjust this as needed, this also limits deviation, so if you set this to 0.5 then the deviation will be limited to half
#include <pulse/simple.h>
#include <pulse/error.h>
#ifdef ALSA_OUTPUT
#include <alsa/asoundlib.h>
#endif
#define VOLUME 0.1f // SCA Volume
#define VOLUME_AUDIO 1.0f // SCA Audio volume
#define FREQUENCY 67000 // SCA Frequency
@@ -66,11 +71,13 @@ int main() {
.maxlength = buffer_maxlength,
.fragsize = buffer_tlength_fragsize
};
#ifndef ALSA_OUTPUT
pa_buffer_attr output_buffer_atr = {
.maxlength = buffer_maxlength,
.tlength = buffer_tlength_fragsize,
.prebuf = buffer_prebuf
};
#endif
printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
@@ -91,7 +98,7 @@ int main() {
}
printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
#ifndef ALSA_OUTPUT
pa_simple *output_device = pa_simple_new(
NULL,
"SCAMod",
@@ -108,6 +115,33 @@ int main() {
pa_simple_free(input_device);
return 1;
}
#else
snd_pcm_hw_params_t *output_params;
snd_pcm_t *output_handle;
int output_error = snd_pcm_open(&output_handle, OUTPUT_DEVICE, SND_PCM_STREAM_PLAYBACK, 0);
if(output_error < 0) {
fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
pa_simple_free(input_device);
return 1;
}
snd_pcm_hw_params_malloc(&output_params);
snd_pcm_hw_params_any(output_handle, output_params);
snd_pcm_hw_params_set_access(output_handle, output_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(output_handle, output_params, SND_PCM_FORMAT_FLOAT); // Same as pulse's Float32NE
snd_pcm_hw_params_set_channels(output_handle, output_params, 1);
unsigned int rate = SAMPLE_RATE;
int dir;
snd_pcm_hw_params_set_rate_near(output_handle, output_params, &rate, &dir);
snd_pcm_uframes_t frames = BUFFER_SIZE;
snd_pcm_hw_params_set_period_size_near(output_handle, output_params, &frames, &dir); // i don't have a clue why like this
output_error = snd_pcm_hw_params(output_handle, output_params);
if(output_error < 0) {
fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
snd_pcm_close(output_handle);
pa_simple_free(input_device);
return 1;
}
#endif
FMModulator mod;
init_fm_modulator(&mod, FREQUENCY, DEVIATION, SAMPLE_RATE);
@@ -157,14 +191,23 @@ int main() {
signal[i] = modulate_fm(&mod, current_input)*VOLUME;
}
#ifndef ALSA_OUTPUT
if (pa_simple_write(output_device, signal, sizeof(signal), &pulse_error) < 0) {
fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
#else
snd_pcm_writei(output_handle, signal, sizeof(signal));
#endif
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
#ifndef ALSA_OUTPUT
pa_simple_free(output_device);
#else
snd_pcm_drain(output_handle);
snd_pcm_free(output_handle);
#endif
return 0;
}

4
src/stereo_coder.c
View File

@@ -23,7 +23,7 @@
#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
#define ALSA_OUTPUT // Output, not input or both
// #define ALSA_OUTPUT // Output, not input or both
#define BUFFER_SIZE 512
#define CLIPPER_THRESHOLD 0.525 // Adjust this as needed
@@ -89,11 +89,13 @@ int main() {
.maxlength = buffer_maxlength,
.fragsize = buffer_tlength_fragsize
};
#ifndef ALSA_OUTPUT
pa_buffer_attr output_buffer_atr = {
.maxlength = buffer_maxlength,
.tlength = buffer_tlength_fragsize,
.prebuf = buffer_prebuf
};
#endif
int open_pulse_error;