dss.c

i2s usb audio demo lpc2138

#include "LPC23xx.h"                       /* LPC23xx definitions  */
#include "type.h"

#include "../l3/inc/l3.h"
#include "../Application/inc/envelope.h"

// This is a simple example of direct digital synthesis of a sine wave for the LPC2138
// The default configuration is an accumulator size of 2^32
// This is overkill but it's the default register size
// Also uses 100000 kHz sample rate (interrupt is set for 10 uS)
// Nothing particularily sacred about this value, can be set slower
// For technical details see http://alfa.iele.polsl.gliwice.pl/~izi/DDS(1).pdf
// "Direct Digital Synthesis: A Tool for Periodic Wave Generation (Part 1)" by Lionel Cordesses"

// The sine table has 256 elements, you can easily use 128, 256, 512, 1024, 2048 (files are provided in folder)
// The number of elements defines the lowest achievable frequency

// There is some phase jitter in some tables at certain frequencies, tables could be cleaned up 

// change numbers if accumulator size or sample rate is changed
// there is no good reason to change accumulator size from 32 bits
// since it is native register size

// accumulator size / sample rate
// 2^32 / 100000 kHz
// 429496729 / 100000 = 42950
//#define freq_mult 42950 
#define freq_mult 97391 /* 429496729/44100 */ 	 

//#define freq_mult 48696 /* 429496729/88200 */ /* interrupt happens 88200/sec */	 

extern unsigned int songindex;
extern int I2S_muteFlag;


DWORD sampleCount = 0; 
unsigned int freq;		 
unsigned int accum=0;

/* functions exported from this file */
	 void nextTone(void);

void nextTone(void)
{
    signed short sample, tone; 
	DWORD buffer=0;
	static long j=3, direction =1;

	/* phase increment = accumulator size / sample rate * frequency	 */
	if(I2S_muteFlag == 0)	   /* if inside envelope play note */
	{

    accum += (freq_mult) * (freq);

	sample = sine_table1024[accum>>22]; // 10 bit sine pointer, use for 1024 table
 										   
	/* send the sample to the I2S, fixed support 16-bit stereo	*/

	  sampleCount++;												
	 tone = adjustTone(sample, sampleCount);

	  if (direction == 0) tone = tone>>1; 	   /* adjust amplitude */	

	  buffer =  (tone*(0x4-j));						  /* Adjust left channel */
//	  buffer =  (tone);	                
	  buffer &= (0xFFFF);		                      /* clear sign extention on negative numbers */
 	  buffer = buffer | (tone*j << 16 );		      /* Adjust right channel */
// 	  buffer = buffer | (tone << 16 );  
      I2S_TX_FIFO = buffer; /* write to I2S */
	}							  
	else
	{
      I2S_TX_FIFO = 0; /* write to I2S */
	}
							 /* ADSR envelope = attackCount + decayCount + sustainCount + releaseCount */
    if (sampleCount >= releaseCount) /* Set mute flag every envelope/44100 Hz */
 	{
	   I2S_muteFlag = 1;
	   FIO2CLR = 0x1;
	   sampleCount = 0;

	   if (direction ==1)j++;
	   else j--;
	   if (j == 4) direction = 0 ;	 /* four notes in front going left */
	   if (j == 0) direction = 1 ;	 /* two notes in back going right */												   
    }												   
}