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DSP Lab 1

This document introduces digital signal processing and MATLAB. It discusses sampling analog signals into digital form, representing signals in different domains like time and frequency. It explains digital filtering techniques in time/space domains like linear, causal and stable filters. The Fourier transform is used to convert between time/space and frequency domains. MATLAB is used to generate and analyze sample waveforms to familiarize with its DSP and signal processing toolbox functions.

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139 views3 pages

DSP Lab 1

This document introduces digital signal processing and MATLAB. It discusses sampling analog signals into digital form, representing signals in different domains like time and frequency. It explains digital filtering techniques in time/space domains like linear, causal and stable filters. The Fourier transform is used to convert between time/space and frequency domains. MATLAB is used to generate and analyze sample waveforms to familiarize with its DSP and signal processing toolbox functions.

Uploaded by

Anonymous PfQKLcTt
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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LAB NO - 01

OBJECTIVE:
Introduction to digital signal processing and to become familiarize with MATLAB software
general function and signal processing tool box function.

DIGITAL SIGNAL PROCESSING


Digital signal processing (DSP) is the use of digital processing, such as by computers, to perform a
wide variety of signal processing operations. The signals processed in this manner are a sequence of
numbers that represent samples of a continuous variable in a domain such as time, space, or
frequency.
Digital signal processing and analog signal processing are subfields of signal processing. DSP
applications include audio and speech signal processing, sonar, radar and other sensor
array processing, spectral estimation, statistical signal processing, digital image processing, signal
processing for telecommunications, control of systems, biomedical engineering, seismic data
processing, among others.
Digital Signal Processing refers to techniques for improving the accuracy and reliability of digital
communication.
It is a matrix based powerful software package for scientific & engineering computation &
utilization
Complex numerical problem can be solved in a correction of time that requires in other high
level languages.
It provide an interactive environment with hundreds of built in functions for technical
computation graphics & animations.
In MATLAB user can create his own function.

SIGNAL SAMPLING
The increasing use of computers has resulted in the increased use of, and need for, digital signal
processing. To digitally analyze and manipulate an analog signal, it must be digitized with an analogto-digital converter. Sampling is usually carried out in two stages, discretization and quantization.
Discretization means that the signal is divided into equal intervals of time, and each interval is
represented by a single measurement of amplitude. Quantization means each amplitude measurement
is approximated by a value from a finite set. Rounding real numbers to integers is an example.

DOMAIN

In DSP, engineers usually study digital signals in one of the following domains: time domain (onedimensional signals), spatial domain (multidimensional signals), frequency domain,
and wavelet domains. They choose the domain in which to process a signal by making an informed
assumption (or by trying different possibilities) as to which domain best represents the essential
characteristics of the signal. A sequence of samples from a measuring device produces a temporal or
spatial domain representation, whereas a discrete Fourier transform produces the frequency domain
information, that is, the frequency spectrum.

TIME & SPACE DOMAIN


The most common processing approach in the time or space domain is enhancement of the input
signal through a method called filtering. Digital filtering generally consists of some linear
transformation of a number of surrounding samples around the current sample of the input or output
signal. There are various ways to characterize filters; for example:

A "linear" filter is a linear transformation of input samples; other filters are "non-linear". Linear
filters satisfy the superposition condition, i.e. if an input is a weighted linear combination of
different signals, the output is a similarly weighted linear combination of the corresponding
output signals.

A "causal" filter uses only previous samples of the input or output signals; while a "non-causal"
filter uses future input samples. A non-causal filter can usually be changed into a causal filter by
adding a delay to it.

A "time-invariant" filter has constant properties over time; other filters such as adaptive
filters change in time.

A "stable" filter produces an output that converges to a constant value with time, or remains
bounded within a finite interval. An "unstable" filter can produce an output that grows without
bounds, with bounded or even zero input.

A "finite impulse response" (FIR) filter uses only the input signals, while an "infinite impulse
response" filter (IIR) uses both the input signal and previous samples of the output signal. FIR
filters are always stable, while IIR filters may be unstable.

FREQUENCY DOMAIN

Signals are converted from time or space domain to the frequency domain usually through
the Fourier transform. The Fourier transform converts the signal information to a magnitude
and phase component of each frequency. Often the Fourier transform is converted to the
power spectrum, which is the magnitude of each frequency component squared.

The most common purpose for analysis of signals in the frequency domain is analysis of
signal properties. The engineer can study the spectrum to determine which frequencies are
present in the input signal and which are missing.

In addition to frequency information, phase information is often needed. This can be obtained
from the Fourier transform. With some applications, how the phase varies with frequency can
be a significant consideration.

EXAMPLES:

CONCLUSION:
In this lab draw the different wave form of function on MAT Lab. Such as real
number,sine wave, cosine wave, square wave, sine and cosine wave. Using MATLAB these function
program code write in MATLAB and give the suitable result on graph.

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