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#include "compute.h"
#include "fft.h"
#include <math.h>
#define MAX_SAMPLES 2048
gfloat compute_level(const float *data, size_t nsamples, size_t nchan) {
double rate=44100; //TODO dynamique
size_t i;
float input[MAX_SAMPLES], output[128];
float level;
if (nsamples >= MAX_SAMPLES) {
printf("WARN : nsamples >= MAX_SAMPLES : %i >= %i\n", nsamples, MAX_SAMPLES);
nsamples=MAX_SAMPLES;
}
/* Just return the max peak
for (i=0;i<nsamples;i+=nchan) {
val=((float *)data)[i];
//printf("val==%i\n", val);
if (val<0) val=-val;
if (level<val) level=val;
}
*/
for (i=0;i<nsamples;i++) {
input[i]=data[i/**nchan*/];
/* printf("\r%f ", input[i]);
fflush(stdout);
*/
}
// printf("\n");
compute_spectrom(input, nsamples, rate, output);
level=0.f;
for (i=1;i<128;i++) {
level+=output[i];
}
level/=127.f;
//printf("%f\n", level);
return level;
}
// From Audacity
void compute_spectrom(float * data, int width, double rate, float *output) {
int i;
float processed[256]={0.0f};
//TODO : remove init here
float in[256]={0.0f};
float out[256]={0.0f};
int start = 0;
int windows = 0;
while (start + 256 <= width) {
for (i=0; i<256; i++)
in[i] = data[start + i];
// Hanning
for (i=0; i<256; i++)
in[i] *= 0.50 - 0.50 * cos(2 * M_PI * i / (256 - 1));
PowerSpectrum(in, out);
// Take real part of result
for (i=0; i<256/2; i++)
processed[i] += out[i];
start += 256/2;
windows++;
}
// Convert to decibels
// But do it safely; -Inf is nobody's friend
for (i = 0; i < 256/2; i++){
float temp=(processed[i] / 256 / windows);
if (temp > 0.0)
processed[i] = 10*log10(temp);
else
processed[i] = 0;
}
for(i=0;i<256/2;i++)
output[i] = processed[i];
}
/*
* PowerSpectrum
*
* This function computes the same as RealFFT, above, but
* adds the squares of the real and imaginary part of each
* coefficient, extracting the power and throwing away the
* phase.
*
* For speed, it does not call RealFFT, but duplicates some
* of its code.
*/
void PowerSpectrum(float *In, float *Out)
{
int i;
float theta = M_PI / 128;
float tmpReal[128];
float tmpImag[128];
float RealOut[128];
float ImagOut[128];
for (i = 0; i < 128; i++) {
tmpReal[i] = In[2 * i];
tmpImag[i] = In[2 * i + 1]; //FIXME : WTFFFF ?
tmpImag[i]=0;
}
FFT(128, 0, tmpReal, tmpImag, RealOut, ImagOut);
float wtemp = sin(0.5 * theta);
float wpr = -2.0 * wtemp * wtemp;
float wpi = -1.0 * sin(theta);
float wr = 1.0 + wpr;
float wi = wpi;
int i3;
float h1r, h1i, h2r, h2i, rt, it;
for (i = 1; i < 128 / 2; i++) {
i3 = 128 - i;
h1r = 0.5 * (RealOut[i] + RealOut[i3]);
h1i = 0.5 * (ImagOut[i] - ImagOut[i3]);
h2r = 0.5 * (ImagOut[i] + ImagOut[i3]);
h2i = -0.5 * (RealOut[i] - RealOut[i3]);
rt = h1r + wr * h2r - wi * h2i;
it = h1i + wr * h2i + wi * h2r;
Out[i] = rt * rt + it * it;
rt = h1r - wr * h2r + wi * h2i;
it = -h1i + wr * h2i + wi * h2r;
Out[i3] = rt * rt + it * it;
wr = (wtemp = wr) * wpr - wi * wpi + wr;
wi = wi * wpr + wtemp * wpi + wi;
}
rt = (h1r = RealOut[0]) + ImagOut[0];
it = h1r - ImagOut[0];
Out[0] = rt * rt + it * it;
rt = RealOut[128 / 2];
it = ImagOut[128 / 2];
Out[128 / 2] = rt * rt + it * it;
}
void audio2hsv_1(gint audio_level, gint *light_h, gint *light_s, gint *light_v) {
// Dummy code
*light_h=-audio_level;
*light_s=audio_level;
*light_v=65535;
}
void hsv2rgb(gint h, gint s, gint v, gint *r, gint *g, gint *b) {
/*
* Purpose:
* Convert HSV values to RGB values
* All values are in the range [0..65535]
*/
float F, M, N, K;
int I;
if ( s == 0 ) {
/*
* Achromatic case, set level of grey
*/
*r = v;
*g = v;
*b = v;
} else {
I = (int) h/(65535/6); /* should be in the range 0..5 */
F = h - I; /* fractional part */
M = v * (1 - s);
N = v * (1 - s * F);
K = v * (1 - s * (1 - F));
if (I == 0) { *r = v; *g = K; *b = M; }
if (I == 1) { *r = N; *g = v; *b = M; }
if (I == 2) { *r = M; *g = v; *b = K; }
if (I == 3) { *r = M; *g = N; *b = v; }
if (I == 4) { *r = K; *g = M; *b = v; }
if (I == 5) { *r = v; *g = M; *b = N; }
}
}
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