Cleaning files

4 files changed, 179 insertions, 191 deletions

diff --git a/lab1/RTDSP/volume.c b/lab1/RTDSP/volume.c
index 3da8881..43958f8 100644
--- a/lab1/RTDSP/volume.c
+++ b/lab1/RTDSP/volume.c
@@ -1,20 +1,20 @@
 /*************************************************************************************
-			       DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
-					   		     IMPERIAL COLLEGE LONDON 
+                   DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
+                                    IMPERIAL COLLEGE LONDON
 
- 				      EE 3.19: Real Time Digital Signal Processing
-					       Dr Paul Mitcheson and Daniel Harvey
+                       EE 3.19: Real Time Digital Signal Processing
+                           Dr Paul Mitcheson and Daniel Harvey
 
-				        LAB 1: Getting Started with the TI C6x DSP 
+                        LAB 1: Getting Started with the TI C6x DSP
 
- 				          ********* V O L U M E . C **********
+                           ********* V O L U M E . C **********
 
-             Part of the volume example. Demonstrates connecting the DSK 
-         			    to sinewave data stored on the host PC.
-				    Sinewave is then processed by increasing its gain.
-		      Additionally demonstrates calling an assembly routine from C. 
+             Part of the volume example. Demonstrates connecting the DSK
+                         to sinewave data stored on the host PC.
+                    Sinewave is then processed by increasing its gain.
+              Additionally demonstrates calling an assembly routine from C.
  *************************************************************************************
- 				Updated for use on 6713 DSK by Danny Harvey: May-Aug 2006
+                 Updated for use on 6713 DSK by Danny Harvey: May-Aug 2006
  ************************************************************************************/
 /* "@(#) DSP/BIOS 4.90.270 01-08-04 (bios,dsk6713-c04)" */
 
@@ -24,19 +24,19 @@
 #include "volume.h"
 
 /******************************* Global declarations ********************************/
-// note: for see volume.h for initialisaton of BUFSIZE, MINGAIN and BASELOAD 
+// note: for see volume.h for initialisaton of BUFSIZE, MINGAIN and BASELOAD
 // and defintion of PARMS
 
-int inp_buffer[BUFSIZE];       			 // processing data buffers 
+int inp_buffer[BUFSIZE];                    // processing data buffers
 int out_buffer[BUFSIZE];
-int gain = MINGAIN;                      // volume control variable 
-unsigned int processingLoad = BASELOAD;  // processing routine load value 
-
-struct PARMS str =						 // A structure of type PARMS is initialised. 
-{										 // This struct is not used in the logic of  
-    2934,								 // the program, it is included to demonstrate 
-    9432,								 // how to use watch windows on structs.
-    213,								
+int gain = MINGAIN;                      // volume control variable
+unsigned int processingLoad = BASELOAD;  // processing routine load value
+
+struct PARMS str =                         // A structure of type PARMS is initialised.
+{                                         // This struct is not used in the logic of
+    2934,                                 // the program, it is included to demonstrate
+    9432,                                 // how to use watch windows on structs.
+    213,
     9432,
     &str
 };
@@ -50,23 +50,23 @@ static void dataIO(void);
 /********************************** Main routine ************************************/
 void main()
 {
-    int *input = &inp_buffer[0];		// use pointers to point to first element in
-    int *output = &out_buffer[0];		// input and output buffers
+    int *input = &inp_buffer[0];        // use pointers to point to first element in
+    int *output = &out_buffer[0];        // input and output buffers
 
-    puts("volume example started\n");	// send a message to stdio 
+    puts("volume example started\n");    // send a message to stdio
 
-    // loop forever 
+    // loop forever
     while(TRUE)
-    {        
-         //  Read input data using a probe-point connected to a host file. 
+    {
+         //  Read input data using a probe-point connected to a host file.
          //  Write output data to a graph connected through a probe-point.
-         
+
         dataIO();
 
         #ifdef FILEIO
         puts("begin processing")        /****** deliberate syntax error ******/
         #endif
-        
+
         // process signal held in input buffer array (apply gain)
         // result is returned to output buffer array
         processing(input, output);
@@ -80,15 +80,15 @@ static int processing(int *input, int *output)
 {
     int size = BUFSIZE;
 
-	/* loop through length of input array mutiplying by gain. Put the result in 
-	 the output array. */
+    /* loop through length of input array mutiplying by gain. Put the result in
+     the output array. */
     while(size--){
         *output++ = *input++ * gain;
     }
-        
+
     // apply additional processing load by calling assembly function load()
     load(processingLoad);
-    
+
     return(TRUE);
 }
 
diff --git a/lab1/RTDSP/volume.h b/lab1/RTDSP/volume.h
index c70ae5d..9b46bf5 100644
--- a/lab1/RTDSP/volume.h
+++ b/lab1/RTDSP/volume.h
@@ -3,7 +3,7 @@
  *  All rights reserved. Property of Texas Instruments Incorporated.
  *  Restricted rights to use, duplicate or disclose this code are
  *  granted through contract.
- *  
+ *
  */
 /* "@(#) DSP/BIOS 4.90.270 01-08-04 (bios,dsk6713-c04)" */
 /*
@@ -31,11 +31,11 @@
 #define BASELOAD 1
 
 struct PARMS {
-	int Beta;
-	int EchoPower;
-	int ErrorPower;
-	int Ratio;
-	struct PARMS *Link;
+    int Beta;
+    int EchoPower;
+    int ErrorPower;
+    int Ratio;
+    struct PARMS *Link;
 };
 
 #endif /* __VOLUME_H */
diff --git a/lab2/RTDSP/sine.c b/lab2/RTDSP/sine.c
index aa2653e..561ca0c 100644
--- a/lab2/RTDSP/sine.c
+++ b/lab2/RTDSP/sine.c
@@ -1,33 +1,33 @@
 /*************************************************************************************
-			       DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
-					   		     IMPERIAL COLLEGE LONDON 
+                   DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
+                                    IMPERIAL COLLEGE LONDON
 
- 				      EE 3.19: Real Time Digital Signal Processing
-					       Dr Paul Mitcheson and Daniel Harvey
+                       EE 3.19: Real Time Digital Signal Processing
+                           Dr Paul Mitcheson and Daniel Harvey
 
-				        LAB 2: Learning C and Sinewave Generation 
+                        LAB 2: Learning C and Sinewave Generation
 
- 				             ********* S I N E . C **********
+                              ********* S I N E . C **********
 
-             	Demonstrates outputing data from the DSK's audio port. 
-   			  Used for extending knowledge of C and using look up tables.
+                 Demonstrates outputing data from the DSK's audio port.
+                 Used for extending knowledge of C and using look up tables.
 
  *************************************************************************************
-  				Updated for use on 6713 DSK by Danny Harvey: May-Aug 06/Dec 07/Oct 09
-				CCS V4 updates Sept 10
+                  Updated for use on 6713 DSK by Danny Harvey: May-Aug 06/Dec 07/Oct 09
+                CCS V4 updates Sept 10
  ************************************************************************************/
 /*
  *  Initialy this example uses the AIC23 codec module of the 6713 DSK Board Support
- *  Library to generate a 1KHz sine wave using a simple digital filter. 
- *	You should modify the code to generate a sine of variable frequency.
+ *  Library to generate a 1KHz sine wave using a simple digital filter.
+ *    You should modify the code to generate a sine of variable frequency.
  */
 /**************************** Pre-processor statements ******************************/
 
-//  Included so program can make use of DSP/BIOS configuration tool.  
+//  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 
+/* 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"
@@ -39,7 +39,7 @@
 #include "helper_functions_polling.h"
 
 
-// PI defined here for use in your code 
+// PI defined here for use in your code
 #define PI 3.141592653589793
 
 // Contains the size of the sine table
@@ -49,12 +49,12 @@
 
 /******************************* Global declarations ********************************/
 
-/* Audio port configuration settings: these values set registers in the AIC23 audio 
+/* 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         */ 
-			 /**********************************************************************/\
+             /**********************************************************************/
+             /*   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                   */\
@@ -65,27 +65,27 @@ DSK6713_AIC23_Config Config = { \
     0x004f,  /* 7 DIGIF      Digital audio interface format  32 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  
+// Codec handle:- a variable used to identify audio interface
 DSK6713_AIC23_CodecHandle H_Codec;
 
 /* Sampling frequency in HZ. Must only be set to 8000, 16000, 24000
-32000, 44100 (CD standard), 48000 or 96000  */ 
+32000, 44100 (CD standard), 48000 or 96000  */
 int sampling_freq = 16000;
 
 
-// Array of data used by sinegen to generate sine. These are the initial values.                        
+// Array of data used by sinegen to generate sine. These are the initial values.
 float y[3] = {0,0,0};
 float x[1] = {1}; // impulse to start filter
 
 float a0 = 1.4142; // coefficients for difference equation
 float b0 = 0.707;
 
-// Holds the value of the current sample 
-float sample; 
+// Holds the value of the current sample
+float sample;
 
 /* Left and right audio channel gain values, calculated to be less than signed 32 bit
  maximum value. */
@@ -93,7 +93,7 @@ Int32 L_Gain = 2100000000;
 Int32 R_Gain = 2100000000;
 
 
-/* Use this variable in your code to set the frequency of your sine wave 
+/* Use this variable in your code to set the frequency of your sine wave
    be carefull that you do not set it above the current nyquist frequency! */
 float sine_freq = 1000.0;
 
@@ -103,7 +103,7 @@ float table[SINE_TABLE_SIZE];
 // Current index in the table, that can be used to calculate the next index
 int sine_index = 0;
 
-     
+
 /******************************* Function prototypes ********************************/
 void init_hardware(void);
 void sine_init(void);
@@ -112,79 +112,72 @@ float sinegen(void);
 void main()
 {
 
-	// initialize board and the audio port
-	init_hardware();
-	
-	// Initialize the sine table
-	sine_init();
-	
-    // Loop endlessley generating a sine wave 
+    // initialize board and the audio port
+    init_hardware();
+
+    // Initialize the sine table
+    sine_init();
+
+    // Loop endlessley generating a sine wave
     while(1)
     {
- 		// Calculate next sample
- 		sample = sinegen();
- 		
-     	/* Send a sample to the audio port if it is ready to transmit.
+        // Calculate next sample
+        sample = sinegen();
+
+        /* Send a sample to the audio port if it is ready to transmit.
            Note: DSK6713_AIC23_write() returns false if the port if is not ready */
 
         //  send to LEFT channel (poll until ready)
         while (!DSK6713_AIC23_write(H_Codec, ((Int32)(sample * L_Gain))))
         {};
-		// send same sample to RIGHT channel (poll until ready)
+
+        // send same sample to RIGHT channel (poll until ready)
         while (!DSK6713_AIC23_write(H_Codec, ((Int32)(sample * R_Gain))))
         {};
-        
-		// Set the sampling frequency. This function updates the frequency only if it 
-		// has changed. Frequency set must be one of the supported sampling freq.
-		set_samp_freq(&sampling_freq, Config, &H_Codec);
-	
-	}
 
+        // Set the sampling frequency. This function updates the frequency only if it
+        // has changed. Frequency set must be one of the supported sampling freq.
+        set_samp_freq(&sampling_freq, Config, &H_Codec);
+    }
 }
 
 /******************************* init_hardware() ************************************/
 void init_hardware()
 {
-    // Initialize the board support library, must be called first 
+    // Initialize the board support library, must be called first
     DSK6713_init();
-    
-    // Start the codec using the settings defined above in config 
+
+    // Start the codec using the settings defined above in config
     H_Codec = DSK6713_AIC23_openCodec(0, &Config);
 
-	/* Defines number of bits in word used by MSBSP for communications with AIC23
-	 NOTE: this must match the bit resolution set in in the AIC23 */
-	MCBSP_FSETS(XCR1, XWDLEN1, 32BIT);
-	
-	/* Set the sampling frequency of the audio port. Must only be set to a supported 
-	   frequency (8000/16000/24000/32000/44100/48000/96000) */
-	
-	DSK6713_AIC23_setFreq(H_Codec, get_sampling_handle(&sampling_freq));
+    /* Defines number of bits in word used by MSBSP for communications with AIC23
+       NOTE: this must match the bit resolution set in in the AIC23 */
+    MCBSP_FSETS(XCR1, XWDLEN1, 32BIT);
 
+    /* Set the sampling frequency of the audio port. Must only be set to a supported
+       frequency (8000/16000/24000/32000/44100/48000/96000) */
+    DSK6713_AIC23_setFreq(H_Codec, get_sampling_handle(&sampling_freq));
 }
 
-/******************************* sin_init() ************************************/
+/******************************* sin_init() *****************************************/
 void sine_init()
 {
-	int i;
-	for(i = 0; i < SINE_TABLE_SIZE; i++)
-	{
-		table[i] = sin(2 * PI * i / SINE_TABLE_SIZE);
-	}
+    int i;
+    for(i = 0; i < SINE_TABLE_SIZE; i++)
+    {
+        table[i] = sin(2 * PI * i / SINE_TABLE_SIZE);
+    }
 }
 
-/********************************** sinegen() ***************************************/   
+/********************************** sinegen() ***************************************/
 float sinegen(void)
 {
-	// temporary variable used to output values from function
-	float wave;
-	
-	wave = table[(int)((sine_freq / (float)sampling_freq) * (float)SINE_TABLE_SIZE * sine_index) % SINE_TABLE_SIZE];
-	
-	sine_index++;
-
-    return(wave);
-    
-}
+    // temporary variable used to output values from function
+    float wave;
 
+    wave = table[(int)((sine_freq / (float)sampling_freq) * (float)SINE_TABLE_SIZE * sine_index) % SINE_TABLE_SIZE];
 
+    sine_index++;
 
+    return wave;
+}
diff --git a/lab3/RTDSP/intio.c b/lab3/RTDSP/intio.c
index 7c33ee8..4ee527a 100644
--- a/lab3/RTDSP/intio.c
+++ b/lab3/RTDSP/intio.c
@@ -1,32 +1,32 @@
 /*************************************************************************************
-			       DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
-					   		     IMPERIAL COLLEGE LONDON 
+                   DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
+                                    IMPERIAL COLLEGE LONDON
 
- 				      EE 3.19: Real Time Digital Signal Processing
-					       Dr Paul Mitcheson and Daniel Harvey
+                       EE 3.19: Real Time Digital Signal Processing
+                           Dr Paul Mitcheson and Daniel Harvey
 
-				        		  LAB 3: Interrupt I/O
+                                  LAB 3: Interrupt I/O
 
- 				            ********* I N T I O. C **********
+                             ********* I N T I O. C **********
 
-  Demonstrates inputing and outputing data from the DSK's audio port using interrupts. 
+  Demonstrates inputing and outputing data from the DSK's audio port using interrupts.
 
  *************************************************************************************
- 				Updated for use on 6713 DSK by Danny Harvey: May-Aug 2006
-				Updated for CCS V4 Sept 10
+                 Updated for use on 6713 DSK by Danny Harvey: May-Aug 2006
+                Updated for CCS V4 Sept 10
  ************************************************************************************/
 /*
- *	You should modify the code so that interrupts are used to service the 
+ *    You should modify the code so that interrupts are used to service the
  *  audio port.
  */
 /**************************** Pre-processor statements ******************************/
 
 #include <stdlib.h>
-//  Included so program can make use of DSP/BIOS configuration tool.  
+//  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 
+/* 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"
@@ -40,17 +40,17 @@
 // Some functions to help with configuring hardware
 #include "helper_functions_polling.h"
 
-// PI defined here for use in your code 
+// PI defined here for use in your code
 #define PI 3.141592653589793
 
 /******************************* Global declarations ********************************/
 
-/* Audio port configuration settings: these values set registers in the AIC23 audio 
+/* 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         */ 
-			 /**********************************************************************/\
+             /**********************************************************************/
+             /*   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                   */\
@@ -61,19 +61,19 @@ DSK6713_AIC23_Config Config = { \
     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  
+// Codec handle:- a variable used to identify audio interface
 DSK6713_AIC23_CodecHandle H_Codec;
 
 
 /* Sampling frequency in HZ. Must only be set to 8000, 16000, 24000
-32000, 44100 (CD standard), 48000 or 96000  */ 
+32000, 44100 (CD standard), 48000 or 96000  */
 int sampling_freq = 16000;
 
-/* Use this variable in your code to set the frequency of your sine wave 
+/* Use this variable in your code to set the frequency of your sine wave
    be carefull that you do not set it above the current nyquist frequency! */
 float sine_freq = 2000.0;
 
@@ -90,104 +90,99 @@ Int32 R_Gain = 2100000000;
 unsigned sine_phase_ind = 0;
 
  /******************************* Function prototypes ********************************/
-void init_hardware(void);     
-void init_HWI(void); 
+void init_hardware(void);
+void init_HWI(void);
 float sinegen(void);
 void ISR_AIC(void);
 void sine_init(void);
 /********************************** Main routine ************************************/
 void main(){
-
-	// initialize board and the audio port
+    // initialize board and the audio port
     init_hardware();
 
-  	sine_init();
-  	/* initialize hardware interrupts */
-  	init_HWI();
-
-  	/* loop indefinitely, waiting for interrupts */  					
-  	while(1) {
-  	
-  	};
+    sine_init();
+    /* initialize hardware interrupts */
+    init_HWI();
 
+    /* loop indefinitely, waiting for interrupts */
+    while(1) {};
 }
-        
-/********************************** init_hardware() **********************************/  
+
+/********************************** init_hardware() **********************************/
 void init_hardware()
 {
-    // Initialize the board support library, must be called first 
+    // Initialize the board support library, must be called first
     DSK6713_init();
-    
-    // Start the AIC23 codec using the settings defined above in config 
+
+    // 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);	
+    /* 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);
+    /* 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);
+    /* 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() **************************************/  
+/********************************** 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_XINT1,4);		// Maps an event to a physical interrupt
-	IRQ_enable(IRQ_EVT_XINT1);		// Enables the event
-	IRQ_globalEnable();				// Globally enables interrupts
-
-} 
+    IRQ_globalDisable();            // Globally disables interrupts
+    IRQ_nmiEnable();                // Enables the NMI interrupt (used by the debugger)
+    IRQ_map(IRQ_EVT_XINT1,4);        // Maps an event to a physical interrupt
+    IRQ_enable(IRQ_EVT_XINT1);        // Enables the event
+    IRQ_globalEnable();                // Globally enables interrupts
+}
 
 //Populates the table with appropriate sine values
 void sine_init(void)
 {
-	int i;
-	for(i = 0; i < SINE_TABLE_SIZE; i++)
-	{
-		table[i] = sin(2 * PI * i / SINE_TABLE_SIZE);
-	}
+    int i;
+    for(i = 0; i < SINE_TABLE_SIZE; i++)
+    {
+        table[i] = sin(2 * PI * i / SINE_TABLE_SIZE);
+    }
 }
 
 //Returns the sample according to the sampling frequency
 float sinegen(void)
 {
-	unsigned sample_index = sine_phase_ind * sine_freq * SINE_TABLE_SIZE / sampling_freq;
-	sample_index = sample_index % SINE_TABLE_SIZE;
-	sine_phase_ind++;
-	return table[sample_index];
+    unsigned sample_index = sine_phase_ind * sine_freq * SINE_TABLE_SIZE / sampling_freq;
+    sample_index = sample_index % SINE_TABLE_SIZE;
+    sine_phase_ind++;
+    return table[sample_index];
 }
 
-/******************** INTERRUPT SERVICE ROUTINE ***********************/  
+/******************** INTERRUPT SERVICE ROUTINE ***********************/
 
 /********************************** Ex2 ***************************************/
 void ISR_AIC(void)
 {
-	// temporary variable used to output values from function
-	float wave_out, wave;
-	sine_phase_ind = sine_phase_ind % sampling_freq;
-	wave = sinegen();
-	wave_out = wave < 0 ? wave : -wave;
+    // temporary variable used to output values from function
+    float wave_out, wave;
+    sine_phase_ind = sine_phase_ind % sampling_freq;
+    wave = sinegen();
+    wave_out = wave < 0 ? wave : -wave;
 
-	mono_write_16Bit((short)(wave_out*32767));
+    mono_write_16Bit((short)(wave_out*32767));
 }
 
 
 /********************************** Ex1 ***************************************/
 /*void ISR_AIC()
 {
-	short mono_out, mono_in;
-	mono_in = mono_read_16Bit();
-	mono_out = mono_in; //mono_in < 0 ? -mono_in : mono_in;
-	mono_write_16Bit(mono_out);
+    short mono_out, mono_in;
+    mono_in = mono_read_16Bit();
+    mono_out = mono_in; //mono_in < 0 ? -mono_in : mono_in;
+    mono_write_16Bit(mono_out);
 }
 */