Make fft work #1
2 changed files with 95 additions and 27 deletions
115
shadermeh.c
115
shadermeh.c
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@ -21,47 +21,101 @@ static GLfloat vertex_buffer[] = {
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};
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static GLubyte audio_buffer[AUDIO_SAMPLES * AUDIO_CHANNELS];
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static float audio_sample_data[AUDIO_SAMPLES];
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static fftw_complex fftw_in[AUDIO_SAMPLES];
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static fftw_complex fftw_out[AUDIO_SAMPLES];
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size_t sample_pointer = 0;
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size_t sample_data_pointer = 0;
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size_t sample_rate = 0;
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static float *audio_sample_data;
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static float *audio_receive_data;
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static double *fftw_in;
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static fftw_complex *fftw_out;
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static fftw_plan plan;
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static int try_fetch_audio(void)
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static int try_fetch_audio(float iTimeDelta)
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{
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size_t i, count = 0;
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int ret;
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/* To avoid generating stale images, we keep our own sample buffer,
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* which is then used to move a sliding window of data for the fft and
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* wave samples. We need to do this, as otherwise we would set an upper
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* limit of fps (20 at 4800kHz sample rate), which would not be good.
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* The size of the window is set in the header file. The with our
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* approach is that the buffer allows for drifting to occur within the
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* buffer limits. If you buffer is 3s long the delay can grow to 3s.
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* Choose your buffer size wisely for your application.
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*/
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size_t i;
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ssize_t ret = 0;
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memset(audio_receive_data, 0, AUDIO_BUFFER_SIZE *
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sizeof(*audio_receive_data));
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sample_pointer += (sample_rate * iTimeDelta);
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for (;;) {
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ret = read(STDIN_FILENO,
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(char *)audio_sample_data + count,
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sizeof(audio_sample_data) - count);
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ret = read(STDIN_FILENO, (char *)audio_receive_data,
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sizeof(*audio_receive_data)*AUDIO_BUFFER_SIZE);
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if (ret < 0) {
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if (errno == EINTR)
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continue;
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if (errno == EAGAIN)
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if (errno == EAGAIN || errno == EWOULDBLOCK)
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break;
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perror("stdin");
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return -1;
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}
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if (ret == 0)
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if (ret == 0 || ret % sizeof(float) != 0){
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break;
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}
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count += ret;
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ret /= 4;
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if((ret + sample_pointer) > AUDIO_BUFFER_SIZE){
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/* Not enough storage space to store all new audio data,
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* will override not output data with new one */
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memset(audio_sample_data, 0,
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AUDIO_BUFFER_SIZE * sizeof(*audio_sample_data));
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memcpy(audio_sample_data, audio_receive_data,
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ret * sizeof(*audio_sample_data));
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sample_pointer = 0;
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sample_data_pointer = ret;
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}else{
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memmove(audio_sample_data,
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&audio_sample_data[sample_pointer],
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(AUDIO_BUFFER_SIZE - sample_pointer)*
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sizeof(*audio_sample_data));
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if(sample_data_pointer <= sample_pointer){
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sample_data_pointer = 0;
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}else{
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sample_data_pointer -= sample_pointer;
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}
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sample_pointer = 0;
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size_t len = ret;
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if((ret + sample_data_pointer) >= AUDIO_BUFFER_SIZE){
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len = AUDIO_BUFFER_SIZE - sample_data_pointer;
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}
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memcpy(&audio_sample_data[sample_data_pointer],
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audio_receive_data, len * sizeof(float));
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sample_data_pointer += len;
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break;
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}
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}
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if((sample_pointer+AUDIO_FFT_SIZE) >= sample_data_pointer){
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fprintf(stderr, "shadermeh input to slow %zu > %zu! wrapping around!\n", sample_pointer+AUDIO_FFT_SIZE, sample_data_pointer);
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sample_pointer = 0;
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}
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for (i = 0; i < AUDIO_SAMPLES; ++i)
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fftw_in[i][0] = audio_sample_data[i];
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memset(fftw_in, 0, sizeof(*fftw_in) * AUDIO_BUFFER_SIZE);
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memset(fftw_out, 0, sizeof(*fftw_out) * AUDIO_BUFFER_SIZE);
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for (i = 0; i < AUDIO_FFT_SIZE; ++i)
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fftw_in[i] = audio_sample_data[sample_pointer+i];
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fftw_execute(plan);
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for (i = 0; i < AUDIO_SAMPLES; ++i) {
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float x = fftw_out[i][0], y = fftw_out[i][1];
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float a = sqrt(x * x + y * y);
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float a = cabs(fftw_out[i]);
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audio_buffer[i + AUDIO_SAMPLES] = audio_sample_data[i] * 127.0f + 127.0f;
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audio_buffer[i] = 127.0f + a * 127.0f;
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audio_buffer[i + AUDIO_SAMPLES] = audio_sample_data[sample_pointer+i] * 127.0f + 127.0f;
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audio_buffer[i] = log(fabsf(a)+1) * 50;
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}
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return 0;
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@ -159,11 +213,11 @@ static const struct option long_opts[] = {
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{ "height", required_argument, NULL, 'h' },
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{ "shader", required_argument, NULL, 's' },
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{ "to-stdout", no_argument, NULL, 'S' },
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{ "stdin-audio", no_argument, NULL, 'a' },
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{ "stdin-audio", required_argument, NULL, 'a' },
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{ NULL, 0, NULL, 0 },
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};
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static const char *short_opts = "w:h:s:Sa";
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static const char *short_opts = "w:a:h:s:S";
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static const char *usage_str =
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"shadermeh OPTIONS...\n"
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@ -174,7 +228,7 @@ static const char *usage_str =
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" --height, -h <pixels>\n"
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"\n"
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" --to-stdout, -S Poop raw RGB24 frames to stdout (blocking)\n"
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" --stdin-audio, -a Read raw PCM audio from stdin (non-blocking)\n"
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" --stdin-audio, -a <sample rate> Read raw PCM audio from stdin (non-blocking)\n"
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"\n"
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" --shader, -s <shader file>\n"
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"\n";
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@ -188,7 +242,7 @@ int main(int argc, char **argv)
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void *fb32 = NULL, *fb24 = NULL;
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const char *shader_file = NULL;
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GLint major, minor, prog;
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float iTime, iTimeDelta;
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float iTime, iTimeDelta = 0;
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bool have_audio = false;
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bool to_stdout = false;
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window *wnd;
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@ -218,6 +272,11 @@ int main(int argc, char **argv)
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break;
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case 'a':
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have_audio = true;
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sample_rate = strtol(optarg, NULL, 10);
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audio_sample_data = malloc(AUDIO_BUFFER_SIZE *
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sizeof(float));
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audio_receive_data = malloc(AUDIO_BUFFER_SIZE *
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sizeof(float));
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break;
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default:
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fputs(usage_str, stderr);
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@ -341,8 +400,12 @@ int main(int argc, char **argv)
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glBindSampler(0, sampler_sound);
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if (have_audio) {
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plan = fftw_plan_dft_1d(AUDIO_SAMPLES, fftw_in, fftw_out,
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FFTW_FORWARD, FFTW_ESTIMATE);
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fftw_in = fftw_alloc_real(AUDIO_BUFFER_SIZE);
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fftw_out = fftw_alloc_complex(AUDIO_BUFFER_SIZE);
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if(fftw_in == NULL || fftw_out == NULL)
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goto fail_vao;
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plan = fftw_plan_dft_r2c_1d(AUDIO_BUFFER_SIZE, fftw_in, fftw_out,
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FFTW_MEASURE);
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}
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/******************** framebuffer object ********************/
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@ -377,7 +440,7 @@ int main(int argc, char **argv)
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glClear(GL_COLOR_BUFFER_BIT);
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if (have_audio) {
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if (try_fetch_audio())
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if (try_fetch_audio(iTimeDelta))
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break;
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glBindTexture(GL_TEXTURE_2D, sound_tex);
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@ -451,6 +514,8 @@ fail_vao:
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window_make_current(NULL);
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free(fb32);
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free(fb24);
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fftw_free(fftw_in);
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fftw_free(fftw_out);
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window_destroy(wnd);
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return EXIT_SUCCESS;
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}
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@ -27,10 +27,13 @@
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#include <poll.h>
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#include <time.h>
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#include <fftw3.h>
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#include <math.h>
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#include <complex.h>
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#include <fftw3.h>
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#define AUDIO_SAMPLES (512)
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#define AUDIO_SAMPLES (4096)
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#define AUDIO_BUFFER_SIZE (sample_rate * 3)
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#define AUDIO_FFT_SIZE (AUDIO_SAMPLES * 2)
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#define AUDIO_CHANNELS (2)
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typedef struct {
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