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add code to smooth out frequencies between audio frames

This commit is contained in:
Paul Dino Jones 2024-12-04 11:20:21 -05:00
parent c8e40e9b80
commit a2ff24afd7
1 changed files with 21 additions and 6 deletions
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27
src/monocoque/devices/sound/usb_generic_shaker_pulse.c
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@ -12,7 +12,7 @@
#define FORMAT PA_SAMPLE_S16LE
#define SAMPLE_RATE (44100)
#define AMPLITUDE 1
#define AMPLITUDE .5
#define DURATION 1.0
#ifndef M_PI
@ -46,31 +46,46 @@ void gear_sound_stream(pa_stream *s, size_t length, void *userdata) {
pa_stream_write(s, buffer, length, NULL, 0LL, PA_SEEK_RELATIVE);
}
double get_closest_frequency(size_t num_samples, double input_frequency) {
double samples_per_cycle = SAMPLE_RATE / input_frequency;
double adjusted_frequency = SAMPLE_RATE / (num_samples / floor(num_samples/samples_per_cycle) );
return adjusted_frequency;
}
void engine_sound_stream(pa_stream *s, size_t length, void *userdata) {
SoundData* data = (SoundData*)userdata;
double freq = data->curr_frequency;
size_t num_samples = length / sizeof(int16_t);
num_samples = (size_t) (DURATION * SAMPLE_RATE);
freq = get_closest_frequency( num_samples, freq );
//size_t num_samples = (size_t) (DURATION * SAMPLE_RATE);
int16_t buffer[num_samples];
//double t = 0;
for (size_t i = 0; i < num_samples; i++) {
double t = data->phase;
double sample = AMPLITUDE * 32767.0 * sin(2.0 * M_PI * freq * t);
double sample = AMPLITUDE * 32767.0 * sin( 2.0 * M_PI * freq * t );
buffer[i] = (int16_t)sample;
t += 1.0 / SAMPLE_RATE;
if (t >= DURATION) {
t = -DURATION;
if (t >= SAMPLE_RATE) {
t = 0;
}
data->phase = t;
}
data->phase = round(data->phase);
pa_stream_write(s, buffer, length, NULL, 0LL, PA_SEEK_RELATIVE);
}
void stream_success_cb(pa_stream *stream, int success, void *userdata) {
return;
}