Working on proj

3 files changed, 302 insertions, 261 deletions

diff --git a/Project/RTDSP/.launches/RTDSP.launch b/Project/RTDSP/.launches/RTDSP.launch
index 6a5779e..5aa4ead 100644
--- a/Project/RTDSP/.launches/RTDSP.launch
+++ b/Project/RTDSP/.launches/RTDSP.launch
@@ -13,7 +13,7 @@
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 <stringAttribute key="com.ti.ccstudio.debug.core.MRU_PROGRAM_NAME" value="C:\github\RTDSP\Project\RTDSP\Debug\RTDSP.out"/>
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 <stringAttribute key="org.eclipse.cdt.launch.GLOBAL_VARIABLES" value="&lt;?xml version=&quot;1.0&quot; encoding=&quot;UTF-8&quot;?&gt;&#13;&#10;&lt;globalVariableList/&gt;&#13;&#10;"/>
diff --git a/Project/RTDSP/enhance.c b/Project/RTDSP/enhance.c
new file mode 100644
index 0000000..9829c00
--- /dev/null
+++ b/Project/RTDSP/enhance.c
@@ -0,0 +1,301 @@
+/*************************************************************************************
+			       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 1000
+#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
+// 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                    */\
+			 /**********************************************************************/
+};
+
+typedef struct {
+	complex *spec;
+	float mag_avg;
+} Spectrum;
+
+// 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 */
+volatile int io_ptr=0;              /* Input/ouput pointer for circular buffers */
+volatile int frame_ptr=0;           /* Frame pointer */
+volatile int frame_ctr =0;
+double avg = 0;
+Spectrum M[NUM_M];
+ /******************************* 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 */
+           
+/********************************** Main routine ************************************/
+void main()
+{      
+
+  	int k; // used in various for loops
+  
+/*  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 */
+	
+	/* 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) 
+  	{
+  		int i;
+  		M[k].spec = (complex *) calloc(FFTLEN, sizeof(complex));
+  		M[k].mag_avg = MAX_FLOAT;
+  		for(i = 0; i < FFTLEN; ++i) {
+  			M[k].spec[i].r = MAX_FLOAT;
+  			M[k].spec[i].i = MAX_FLOAT;
+  		}
+  	}  
+
+ 							
+  	/* main loop, wait for interrupt */  
+  	while(1) 	process_frame();
+}
+    
+/********************************** 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
+
+}
+        
+/******************************** process_frame() ***********************************/  
+void process_frame(void)
+{
+	int k, m; 
+	int io_ptr0;
+	avg = 0;
+	/* 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 **************************/
+
+	for (k = 0; k < FFTLEN; ++k) {
+		fft_out[k].i = 0.0;
+		fft_out[k].r = inframe[k];
+	}
+	
+	fft(FFTLEN, fft_out);
+	
+	for(k = 0; k < FFTLEN; ++k) {
+		avg += cabs(fft_out[k]);	
+	}
+	avg /= FFTLEN;
+	
+	if(avg < M[0].mag_avg) {
+		M[0].mag_avg = avg;
+		for(k = 0; k < FFTLEN; ++k) {
+			M[0].spec[k] = fft_out[k];
+		}
+	}
+	
+	for(k = 0; k < FFTLEN; ++k) {
+		fft_out[k] = M[0].spec[k];	
+	}
+	
+	ifft(FFTLEN, fft_out);
+	
+	if(frame_ctr > MAX_COUNT-1) {
+		int i;
+		frame_ctr = 0;
+		free(M[NUM_M-1].spec);
+		for(k = NUM_M-1; k > 0; ++k) {
+			M[k] = M[k-1];
+		}
+		M[0].spec = (complex *) malloc(FFTLEN * sizeof(complex));
+		M[0].mag_avg = MAX_FLOAT;
+  		for(i = 0; i < FFTLEN; ++i) {
+  			M[0].spec[i].r = MAX_FLOAT;
+  			M[0].spec[i].i = MAX_FLOAT;
+  		}
+	}
+	
+	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 */
+	frame_ctr++;
+	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;
+}
+
+/************************************************************************************/
diff --git a/Project/RTDSP/frame.c b/Project/RTDSP/frame.c
deleted file mode 100644
index 4c9954c..0000000
--- a/Project/RTDSP/frame.c
+++ /dev/null
@@ -1,260 +0,0 @@
-/*************************************************************************************
-			       DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
-					   		     IMPERIAL COLLEGE LONDON 
-
- 				      EE 3.19: Real Time Digital Signal Processing
-					       Dr Paul Mitcheson and Daniel Harvey
-
-				        		  LAB 6: Frame Processing
-
- 				            ********* F R A M E. C **********
-
-  				Demonstrates Frame Processing (Interrupt driven) on the DSK. 
-
- *************************************************************************************
- 				Updated for use on 6713 DSK by Danny Harvey: May-Aug 2006
-				Updated for ccsV4 Sept 2010
- ************************************************************************************/
-/*
- *	You should modify the code so that an FFT is applied to an input frame 
- *  which is then IFFT'd and sent to the audio port.
- */
-/**************************** Pre-processor statements ******************************/
-
-//  Included so program can make use of DSP/BIOS configuration tool.  
-#include <stdlib.h>
-#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>
- 
- 
-
-/******************************* 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                 */\
-    0x0001   /* 9 DIGACT     Digital interface activation    On                    */\
-			 /**********************************************************************/
-};
-
-// Codec handle:- a variable used to identify audio interface  
-DSK6713_AIC23_CodecHandle H_Codec;
-
-// PI defined here for use in your code
-#define E_PI 3.1415926535897932384626433832795028841971693993751058209749445923078164062
-
-#define BUFLEN 800  /* Frame buffer length must be even for real fft */
-
-
-/* Pointers to data buffers */                        
-float *input;
-float *intermediate;
-float *output;
-volatile int index = 0; 
-float mag[BUFLEN];
-complex C[BUFLEN];
- 
-
- /******************************* Function prototypes *******************************/
-void init_hardware(void);     
-void init_HWI(void);   
-void ISR_AIC(void);
-void init_arrays(void);
-void wait_buffer(void); 
-void dft_(int N, complex *X);
-void idft_(int N, complex *X);
-                       
-/********************************** Main routine ************************************/
-void main()
-{      
-
-  /* setup arrays */
-  init_arrays();  
-
-	/* initialize board and the audio port */
-  init_hardware();
-	
-  /* initialize hardware interrupts */
-  init_HWI();
-
-    
-  /* loop indefinitely, waiting for interrupts */  					
-  while(1) 
-  {
-  	  	wait_buffer();
-  };
-  
-}
-
-    
-    
-/********************************** 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
-
-} 
-/************************** Allocate memory for arrays *******************************/        
-void init_arrays(void)
-{
-	input        = (float *) calloc(BUFLEN, sizeof(float)); /* Input array */
-    output       = (float *) calloc(BUFLEN, sizeof(float)); /* Output array */
-    intermediate = (float *) calloc(BUFLEN, sizeof(float)); /* Array for processing*/
-}
-
-/*************************** INTERRUPT SERVICE ROUTINE  ******************************/
-
-// Map this to the appropriate interrupt in the DSP BIOS
-
-void ISR_AIC(void)
-{       
-	short sample;
-	float scale = 11585;  
-	
-	sample = mono_read_16Bit();
-
-	/* add new data to input buffer
-	   and scale so that 1v ~= 1.0 */
-	input[index] = ((float)sample)/scale;
-		
-	/* write new output data */
-	mono_write_16Bit((short)(output[index]*scale)); 
-	
-	/* update index and check for full buffer */
-	if (++index == BUFLEN)
-		index=0;		
-}
-
-/******************* Wait for buffer of data to be input/output **********************/
-void wait_buffer(void)
-{
-	float *p;  
-	int i;
-	/* wait for array index to be set to zero by ISR */
-	while(index);
-	
-	/* rotate data arrays */
-	p = input;
-	input = output;
-	output = intermediate;
-	intermediate = p;
-	
-	
-	/************************* DO PROCESSING OF FRAME  HERE **************************/                
-	
-	// Copy data elements to complex
-	for (i = 0; i < BUFLEN; ++i) {
-		C[i].i = 0;
-		C[i].r = intermediate[i];
-	}
-	
-	dft_(BUFLEN, C);
-	
-	// Calculate the magnitude of the complex numbers
-	for (i = 0; i < BUFLEN; ++i) { 
-		mag[i] = cabs(C[i]);
-	}
-    
-	idft_(BUFLEN, C);
-	for (i = 0; i < BUFLEN; ++i) {
-		intermediate[i] = C[i].r;
-	}
-
-	/**********************************************************************************/
-	
-	/* wait here in case next sample has not yet been read in */                          
-	while(!index);
-}        
-
-
-void dft_(int N, complex *X) {
-    int i, n, k;
-	complex* x = malloc(N*sizeof(X[0]));
-	for(i = 0; i < N; ++i) {
-		x[i] = X[i];
-	}
-    for(k = 0; k < N; ++k) {
-        X[k].r = 0.f;
-        X[k].i = 0.f;
-        for(n = 0; n < N; ++n) {
-            X[k].r += x[n].r * cos(2 * E_PI * k * n / N);
-            X[k].i -= x[n].r * sin(2 * E_PI * k * n / N);
-        }
-    }
-    
-    free(x);
-}
-
-void idft_(int N, complex *X) {
-    int i, n, k;
-	complex* x = malloc(N*sizeof(X[0]));
-	for(i = 0; i < N; ++i) {
-		x[i] = X[i];
-	}
-    for(n = 0; n < N; ++n) {
-        X[n].i = 0.f;
-		X[n].r = 0.f;
-        for(k = 0; k < N; ++k) {
-            X[n].r += x[k].r * cos(2 * E_PI * k * n / N) - x[k].i * sin(2 * E_PI * k * n / N);
-        }
-        X[n].r /= N;
-    }
-    
-    free(x);
-}