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A Simple Interactive Program For Real Time Fir Digital Filters Written In The C Language

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Conference

1997 Annual Conference

Location

Milwaukee, Wisconsin

Publication Date

June 15, 1997

Start Date

June 15, 1997

End Date

June 18, 1997

ISSN

2153-5965

Page Count

9

Page Numbers

2.43.1 - 2.43.9

Permanent URL

https://peer.asee.org/6782

Download Count

927

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Paper Authors

author page

James E. Cross

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Abstract
NOTE: The first page of text has been automatically extracted and included below in lieu of an abstract

Session 1220

A Simple Interactive Program for Real-Time FIR Digital Filters Written in the C-Language

James E. Cross Southern University

Abstract

Introduction

The main design task in using a digital signal processor (DSP) for the real-time processing of signals is that of algorithm development. An algorithm must be developed that will transform the signal in a manner to meet the design specifications. This paper is primarily concerned with the design of such an algorithm that is user friendly. The fundamentals of real-time digital signal processing will first be presented. A discussion of possible computer languages for digital signal processors will be given. The approach to developing a user friendly algorithm will be divided into two parts, that of first developing a non-real-time algorithm as a prototype followed by a real-time algorithm. Finally, some test results will be given, followed by some concluding remarks.

Some Fundamentals of Real-time Digital Signal Processing

The main components of a real-time digital signal processor is an analog to digital (A/D) converter, a computer and a digital to analog (D/A) converter. A number of vendors sell analog interface boards(AIB) that include an A/D converter and a D/A converter. Also included is an anti-aliasing filter and signal reconstruction circuitry. It is possible to use a sound card for this purpose. For the sampled analog signal to be accurately reproduced, the requirement is that the signal must be sampled at a rate that is greater than twice the highest frequency present in the signal. This is called the Nyquist rate. Since the highest frequency for sound is about 20KHz, to be certain of reproducing all frequencies accurately, the signal must be sampled at a rate greater than 40KHz. A signal with frequencies going only to 8 KHz is usually considered adequate for telephone quality sound whereas a CD player is expected to responded to the highest possible sound frequencies. A 44.1 Khz sampling rate is used for CD- quality sound. Some vendors use a sampling rate of 48 KHz for their analog interfacing boards. There has been considerable interest recently in the direct sampling of radio signals. This is as opposed to having the radio signal demodulate so that the sound signal is recovered and then sampling the lower frequency sound signal. Considering the Nyquist sampling rate, a 10 MHz signal must be sampled at a rate above 20 MHz.

Another important consideration, especially for filter design, is the speed and memory requirements of the computer to execute the associated algorithm. Several vendors supply special computers or processors that are designed specifically for digital signal processing. These are called digital signal processors or DSPs. The convolution function is the most common function used in digital filters. The convolution function is written as y = h*x where h,

Cross, J. E. (1997, June), A Simple Interactive Program For Real Time Fir Digital Filters Written In The C Language Paper presented at 1997 Annual Conference, Milwaukee, Wisconsin. https://peer.asee.org/6782

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