/*************************************************************************************
                   DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
                                 IMPERIAL COLLEGE LONDON

                      EE 3.19: Real Time Digital Signal Processing
                           Dr Paul Mitcheson and Daniel Harvey

                                 PROJECT: Frame Processing

                            ********* ENHANCE. C **********
                             Shell for speech enhancement

        Demonstrates overlap-add frame processing (interrupt driven) on the DSK.

 *************************************************************************************
                             By Danny Harvey: 21 July 2006
                             Updated for use on CCS v4 Sept 2010
 ************************************************************************************/
/*
 *	You should modify the code so that a speech enhancement project is built
 *  on top of this template.
 */
/**************************** Pre-processor statements ******************************/
//  library required when using calloc
#include <stdlib.h>
//  Included so program can make use of DSP/BIOS configuration tool.
#include "dsp_bios_cfg.h"

/* The file dsk6713.h must be included in every program that uses the BSL.  This
   example also includes dsk6713_aic23.h because it uses the
   AIC23 codec module (audio interface). */
#include "dsk6713.h"
#include "dsk6713_aic23.h"

// math library (trig functions)
#include <math.h>

/* Some functions to help with Complex algebra and FFT. */
#include "cmplx.h"
#include "fft_functions.h"

// Some functions to help with writing/reading the audio ports when using interrupts.
#include <helper_functions_ISR.h>

#define WINCONST 0.85185			/* 0.46/0.54 for Hamming window */
#define FSAMP 8000.0		/* sample frequency, ensure this matches Config for AIC */
#define FFTLEN 256					/* fft length = frame length 256/8000 = 32 ms*/
#define NFREQ (1+FFTLEN/2)			/* number of frequency bins from a real FFT */
#define OVERSAMP 4					/* oversampling ratio (2 or 4) */
#define FRAMEINC (FFTLEN/OVERSAMP)	/* Frame increment */
#define CIRCBUF (FFTLEN+FRAMEINC)	/* length of I/O buffers */
#define FRAME_TIME 2.5
#define MAX_COUNT 20000
#define MAX_FLOAT 3.4E+38
#define OUTGAIN 16000.0				/* Output gain for DAC */
#define INGAIN  (1.0/16000.0)		/* Input gain for ADC  */
#define NUM_M 4
#define NUM_ALPHA 4
// PI defined here for use in your code
#define PI 3.141592653589793
#define TFRAME FRAMEINC/FSAMP       /* time between calculation of each frame */

/******************************* Global declarations ********************************/

/* Audio port configuration settings: these values set registers in the AIC23 audio
   interface to configure it. See TI doc SLWS106D 3-3 to 3-10 for more info. */
DSK6713_AIC23_Config Config = { \
             /**********************************************************************/
             /*   REGISTER              FUNCTION                  SETTINGS         */
             /**********************************************************************/\
    0x0017,  /* 0 LEFTINVOL  Left line input channel volume  0dB                   */\
    0x0017,  /* 1 RIGHTINVOL Right line input channel volume 0dB                   */\
    0x01f9,  /* 2 LEFTHPVOL  Left channel headphone volume   0dB                   */\
    0x01f9,  /* 3 RIGHTHPVOL Right channel headphone volume  0dB                   */\
    0x0011,  /* 4 ANAPATH    Analog audio path control       DAC on, Mic boost 20dB*/\
    0x0000,  /* 5 DIGPATH    Digital audio path control      All Filters off       */\
    0x0000,  /* 6 DPOWERDOWN Power down control              All Hardware on       */\
    0x0043,  /* 7 DIGIF      Digital audio interface format  16 bit                */\
    0x008d,  /* 8 SAMPLERATE Sample rate control        8 KHZ-ensure matches FSAMP */\
    0x0001   /* 9 DIGACT     Digital interface activation    On                    */\
             /**********************************************************************/
};

// Codec handle:- a variable used to identify audio interface
DSK6713_AIC23_CodecHandle H_Codec;

float *inbuffer, *outbuffer;        /* Input/output circular buffers */
float *inframe, *outframe;          /* Input and output frames */
float *inwin, *outwin;              /* Input and output windows */
float ingain, outgain;				/* ADC and DAC gains */
float cpufrac;                      /* Fraction of CPU time used */
complex *fft_out;						/* FFT output */
float *noise;
float *power_in;
float *mag_in;
float* p_w;
float* prev_noise;
float* SNR;
volatile int io_ptr=0;              /* Input/ouput pointer for circular buffers */
volatile int frame_ptr=0;           /* Frame pointer */
volatile int frame_ctr = 0;
volatile int m_ptr = 0;
float snr_val = 0;
float total_snr = 0;
float lambda = 0.05;
float alpha[NUM_ALPHA] = {100, 50, 20, 5};
float avg = 0;
float sum = 0;
float *M[NUM_M];
float mag_N_X;
float K;
float time_constant = 40e-3;		/* Time constant in ms */
int started = 0;
 /******************************* Function prototypes *******************************/
void init_hardware(void);       /* Initialize codec */
void init_HWI(void);            /* Initialize hardware interrupts */
void ISR_AIC(void);             /* Interrupt service routine for codec */
void process_frame(void);       /* Frame processing routine */
void write_spectrum(void);
void get_noise(void);
void low_pass_filter(float* current, float* next);
void overestimation(void);
/********************************** Main routine ************************************/
void main()
{

    int k; // used in various for loops
    int counter = 1;
/*  Initialize and zero fill arrays */

    inbuffer	= (float *) calloc(CIRCBUF, sizeof(float));	/* Input array */
    outbuffer	= (float *) calloc(CIRCBUF, sizeof(float));	/* Output array */
    inframe		= (float *) calloc(FFTLEN, sizeof(float));	/* Array for processing*/
    outframe	= (float *) calloc(FFTLEN, sizeof(float));	/* Array for processing*/
    inwin		= (float *) calloc(FFTLEN, sizeof(float));	/* Input window */
    outwin		= (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    fft_out		= (complex *) calloc(FFTLEN, sizeof(complex));	/* FFT Output */
    power_in	= (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    p_w         = (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    mag_in		= (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    noise		= (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    prev_noise	= (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    SNR			= (float *) calloc(FFTLEN, sizeof(float));	/* Output window */
    for(k = 0; k < FFTLEN; ++k) {
        SNR[k] = 0;
    }
    /* initialize board and the audio port */
    init_hardware();

    /* initialize hardware interrupts */
    init_HWI();

/* initialize algorithm constants */

    for (k=0; k<FFTLEN; ++k)
    {
        inwin[k] = sqrt((1.0-WINCONST*cos(PI*(2*k+1)/FFTLEN))/OVERSAMP);
        outwin[k] = inwin[k];
    }
    ingain=INGAIN;
    outgain=OUTGAIN;

    for (k = 0; k < NUM_M; ++k) {
        M[k] = (float *) calloc(FFTLEN, sizeof(float));
    }

    K = exp(-TFRAME/time_constant);
    /* main loop, wait for interrupt */
    while(1)    {
        process_frame();
        counter++;
        snr_val = total_snr / counter;
    }
}

/********************************** init_hardware() *********************************/
void init_hardware()
{
    // Initialize the board support library, must be called first
    DSK6713_init();

    // Start the AIC23 codec using the settings defined above in config
    H_Codec = DSK6713_AIC23_openCodec(0, &Config);

    /* Function below sets the number of bits in word used by MSBSP (serial port) for
    receives from AIC23 (audio port). We are using a 32 bit packet containing two
    16 bit numbers hence 32BIT is set for  receive */
    MCBSP_FSETS(RCR1, RWDLEN1, 32BIT);

    /* Configures interrupt to activate on each consecutive available 32 bits
    from Audio port hence an interrupt is generated for each L & R sample pair */
    MCBSP_FSETS(SPCR1, RINTM, FRM);

    /* These commands do the same thing as above but applied to data transfers to the
    audio port */
    MCBSP_FSETS(XCR1, XWDLEN1, 32BIT);
    MCBSP_FSETS(SPCR1, XINTM, FRM);


}
/********************************** init_HWI() **************************************/
void init_HWI(void)
{
    IRQ_globalDisable();			// Globally disables interrupts
    IRQ_nmiEnable();				// Enables the NMI interrupt (used by the debugger)
    IRQ_map(IRQ_EVT_RINT1,4);		// Maps an event to a physical interrupt
    IRQ_enable(IRQ_EVT_RINT1);		// Enables the event
    IRQ_globalEnable();				// Globally enables interrupts

}

// Spectrum calculations for the new values
void write_spectrum(void) {
    unsigned int k;
    for(k = 0; k < FFTLEN; ++k) {
        if(power_in[k] < M[m_ptr][k] || M[m_ptr][k] == 0) {
            M[m_ptr][k] = power_in[k];
        }
    }
}

// Noise estimataion
void get_noise(void) {
    int k, i, min_i;
    float min_val;

    for(k = 0; k < FFTLEN; ++k) {
        min_i = 0;
        min_val = M[0][k];
        for(i = 1; i < NUM_M; ++i) {
            if (M[i][k] < min_val && M[i][k]!= 0) {
                min_val = M[i][k];
                min_i = i;
            }
        }
        noise[k] = M[min_i][k];
    }

    overestimation();
}



void overestimation(void) {
    int i;
    sum = 0;
    // Calcualte |signal^2/noise^2| for all k
    for (i = 0; i < FFTLEN; ++i) {
        if(noise[i] != 0) {
            SNR[i] = power_in[i] / noise[i];
            sum += SNR[i];
        }
    }

    // Calculate average
    sum /= FFTLEN;
    avg = sum;
    total_snr += sum;
    // Use SNRs to divide
    for (i = 0; i < FFTLEN; ++i) {
        // Normalising
        SNR[i] /= 2*sum;
        SNR[i] = SNR[i] > 1 ? 1 : SNR[i];
        noise[i] *= alpha[(int)(SNR[i] * (NUM_ALPHA-1))];
    }
}

// Low pass filter X(w)
void low_pass_filter(float* current, float* next) {
    int w;
    for (w = 0; w < FFTLEN; ++w) {
        current[w] = (1-K)*current[w] + K*next[w];
        next[w] = current[w];
    }
}

/******************************** process_frame() ***********************************/
void process_frame(void)
{
    int k, m;
    int io_ptr0;
    /* work out fraction of available CPU time used by algorithm */
    cpufrac = ((float) (io_ptr & (FRAMEINC - 1)))/FRAMEINC;

    /* wait until io_ptr is at the start of the current frame */
    while((io_ptr/FRAMEINC) != frame_ptr);

    /* then increment the framecount (wrapping if required) */
    if (++frame_ptr >= (CIRCBUF/FRAMEINC)) frame_ptr=0;

    /* save a pointer to the position in the I/O buffers (inbuffer/outbuffer) where the
    data should be read (inbuffer) and saved (outbuffer) for the purpose of processing */
    io_ptr0=frame_ptr * FRAMEINC;

    /* copy input data from inbuffer into inframe (starting from the pointer position) */

    m=io_ptr0;
    for (k=0;k<FFTLEN;k++)
    {
        inframe[k] = inbuffer[m] * inwin[k];
        if (++m >= CIRCBUF) m=0; /* wrap if required */
    }

    /************************* DO PROCESSING OF FRAME  HERE **************************/

    // Initialise the array fft_out for FFT
    for (k = 0; k < FFTLEN; ++k) {
        fft_out[k] = cmplx(inframe[k], 0.0);
    }

    // Perform the FFT
    fft(FFTLEN, fft_out);

    // calculate the power spectrum
    for (k = 0; k < FFTLEN; ++k) {
        power_in[k] = fft_out[k].r * fft_out[k].r + fft_out[k].i * fft_out[k].i;
    }

    low_pass_filter(power_in, p_w);
    low_pass_filter(noise, prev_noise);

    // Get average of fft_out and write to Spectrum
    write_spectrum();

    // Set the noise
    get_noise();

    if(frame_ctr > MAX_COUNT-1) {
        int i;
        frame_ctr = 0;
        if(++m_ptr == NUM_M) m_ptr = 0;
        for(i = 0; i < FFTLEN; ++i) {
            M[m_ptr][i] = power_in[i];
        }
    }

    // max(lambda, |N(w)/g(w)|
    for (k = 0; k < FFTLEN; ++k) {
        float g;
        mag_N_X = sqrt(1 - noise[k]/power_in[k]);
        g = mag_N_X > lambda ? mag_N_X : lambda;
        fft_out[k] = rmul(g, fft_out[k]);
    }

    // Back into time domain
    ifft(FFTLEN, fft_out);

    for (k = 0; k < FFTLEN; ++k) {
        outframe[k] = fft_out[k].r;
    }
    /********************************************************************************/

    /* multiply outframe by output window and overlap-add into output buffer */

    m=io_ptr0;

    for (k=0;k<(FFTLEN-FRAMEINC);k++)
    {                                           /* this loop adds into outbuffer */
        outbuffer[m] = outbuffer[m]+outframe[k]*outwin[k];
        if (++m >= CIRCBUF) m=0; /* wrap if required */
    }

    for (;k<FFTLEN;k++)
    {
        outbuffer[m] = outframe[k]*outwin[k];   /* this loop over-writes outbuffer */
        m++;
    }
}
/*************************** INTERRUPT SERVICE ROUTINE  *****************************/

// Map this to the appropriate interrupt in the CDB file

void ISR_AIC(void)
{
    short sample;
    /* Read and write the ADC and DAC using inbuffer and outbuffer */
    sample = mono_read_16Bit();
    inbuffer[io_ptr] = ((float)sample)*ingain;
        /* write new output data */
    mono_write_16Bit((int)(outbuffer[io_ptr]*outgain));

    /* update io_ptr and check for buffer wraparound */

    if (++io_ptr >= CIRCBUF) io_ptr=0;
    frame_ctr++;
    started = 1;
}

/************************************************************************************/