#include "compute.h"

#include "fft.h"
#include <math.h>

#define MIN_SAMPLES 256
#define MAX_SAMPLES 2048

//#define MAX(a,b) (a>b?a:b)
//#define MIN(a,b) (a<b?a:b)

//static inline float todB_a(const float *x);
void compute_spectrum(float *data, int width, float output[PSHalf]);


float compute_level(const float *data, size_t nsamples, int rate) {

	size_t i; 
	float input[MAX_SAMPLES], pwrspec[PSHalf];
	float value;
	int f, min_f_index, max_f_index;

	if (nsamples >= MAX_SAMPLES) {
		printf("WARN : nsamples >= MAX_SAMPLES : %i >= %i\n", nsamples, MAX_SAMPLES);
		nsamples=MAX_SAMPLES;
	}
	if (nsamples < MIN_SAMPLES) {
		printf("WARN : nsamples < MIN_SAMPLES : %i >= %i\n", nsamples, MIN_SAMPLES);
		return -120.f;
	}
		/* Replicate with symmetry the sound to obtain an input buffer of the minimal len
		for (i=0;i<MIN_SAMPLES;i++) {
			if ( (i/nsamples)%2==1 )
				input[i]=data[i]; // First channel only
			else
				input[i]=data[nsamples-i-1];
		}
		nsamples=MIN_SAMPLES;
	} else {*/
		for (i=0;i<nsamples;i++) {
			input[i]=data[i]; // First channel only
		}
	//}

	compute_spectrum(input, nsamples, pwrspec);

	// Compute the mean power for 200Hz to 2000Hz band
	min_f_index=((float)PSHalf)*200.f/(((float)rate)/2.f);
	max_f_index=((float)PSHalf)*2000.f/(((float)rate)/2.f);

	value=0.f;
	for (f=min_f_index;f<=max_f_index;f++) {
		value+=pwrspec[f];
	}
	// Mean value
	value=value/(max_f_index-min_f_index+1);

	return value;
}
/*
static inline float todB_a(const float *x){
  return (float)((*(int32_t *)x)&0x7fffffff) * 7.17711438e-7f -764.6161886f;
}
*/
// Adapted from Audacity 
void compute_spectrum(float *data, int width, float output[PSHalf]) {

	int i, start, windows;
	float temp;
	float in[PSNumS];
	float out[PSHalf];
	float processed[PSHalf]={0.0f};

	start = 0;
	windows = 0;
	while (start + PSNumS <= width) {
		// Windowing : Hanning
		for (i=0; i<PSNumS; i++)
			in[i] = data[start+i] *(0.50-0.50*cos(2*M_PI*i/(PSNumS-1)));

		// Returns only the real part of the result
		PowerSpectrum(in, out);

		for (i=0; i<PSHalf; i++)
			processed[i] += out[i];

		start += PSHalf;
		windows++;
	}
	// Convert to decibels
	// But do it safely; -Inf is nobody's friend
	for (i = 0; i < PSHalf; i++){
		temp=(processed[i] / PSNumS / windows);
		if (temp > 0.0)
			output[i] = 10*log10(temp);
		else
			output[i] = 0;
	}
}

void audio2hsv_1(float audio_level, float *light_h, float *light_s, float *light_v) {
	static float hue=0;
	static float value_decay[8]={0.f};
	static int decay_idx=0;

	int i;
	float level_norm, temp;

	// Testé avec micro externe sur lud-msi
	//[-38dB;0dB] -> [0.0;1.0]
	level_norm=(38.f+audio_level)/38.f;
	if (level_norm<0.0f) level_norm=0.f;
	//if (level_norm<0.1f) level_norm=0.f; //FIXME : ici cache misere pour le tremblement sur plancher bruit
	if (level_norm>1.f) level_norm=1.f;

	if (decay_idx>=8) decay_idx=0;
	value_decay[decay_idx++]=level_norm;

	hue=(hue+0.0002f);
	if (hue>1.f) hue-=1.f;

//	printf("%+3.1f %+1.3f\n", audio_level, level_norm);

	// Dummy code
//	*light_h=hue;
	temp=hue+level_norm/3.f;
	*light_h=temp-(int)temp; // Fractionnal part

	*light_s=1.f;

//	*light_v=level_norm;
	temp=0.f;
	for(i=0;i<8;i++) temp+=value_decay[i];
	temp/=8.f;
	*light_v=temp;
}