Merge pull request #2 from dan12n/project

Merging working project into master

1 files changed, 389 insertions, 0 deletions

diff --git a/Project/RTDSP/enhance.c b/Project/RTDSP/enhance.c
new file mode 100644
index 0000000..09b78d6
--- /dev/null
+++ b/Project/RTDSP/enhance.c
@@ -0,0 +1,389 @@
+/*************************************************************************************
+                   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] = {300, 400, 600, 1000};
+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;
+    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;
+    float temp;
+    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;
+}
+
+/************************************************************************************/