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Basic Components of A DSP System Generic Structure

The document discusses the basic components of a digital signal processing (DSP) system. It describes how a DSP system typically consists of three main components: 1) an analog-to-digital converter that converts an analog input signal into digital form, 2) a digital system that performs operations on the digital signal, and 3) a digital-to-analog converter that converts the digital output back to an analog signal. It also discusses how continuous time signals are converted to discrete time signals through sampling and the relationship between the continuous and discrete time domains and representations.

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119 views

Basic Components of A DSP System Generic Structure

The document discusses the basic components of a digital signal processing (DSP) system. It describes how a DSP system typically consists of three main components: 1) an analog-to-digital converter that converts an analog input signal into digital form, 2) a digital system that performs operations on the digital signal, and 3) a digital-to-analog converter that converts the digital output back to an analog signal. It also discusses how continuous time signals are converted to discrete time signals through sampling and the relationship between the continuous and discrete time domains and representations.

Uploaded by

Md. Asaduzzaman Raz
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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BASIC COMPONENTS OF A DSP SYSTEM

Generic structure:

• In its most general form, a DSP system will consist of three main components, as illustrated in Figure.
• The analog-to-digital (A/D) converter transforms the analog signal xa(t) at the system input into a digital
signal xd [n]. An A/D converter can be thought of as consisting of a sampler (creating a discretetime
signal), followed by a quantizer (creating discrete levels).
• The digital system performs the desired operations on the digital signal xd[n] and produces a
corresponding output yd [n] also in digital form.
• The digital-to-analog (D/A) converter transforms the digital output yd[n] into an analog signal ya(t)
suitable for interfacing with the outside world.
• In some applications, the A/D or D/A converters may not be required; we extend the meaning of DSP
systems to include such cases.

Discrete-time signals are typically written as a function of an index n (for example, x(n) or xn may
represent a discretisation of x(t) sampled every T seconds). In contrast to Continuous signal systems,
where the behaviour of a system is often described by a set of linear differential equations, discrete-time
systems are described in terms of difference equations. Most Monte Carlo simulations utilize a discrete-
timing method, either because the system cannot be efficiently represented by a set of equations, or
because no such set of equations exists. Transform-domain analysis of discrete-time systems often makes
use of the Z transform.
Discrete time processing of continuous time signals:
Even though this course is primarily about the discrete time signal processing, most signals we encounter
in daily life are continuous in time such as speech, music and images. Increasingly discrete-time signals
processing algorithms are being used to process such signals. For processing by digital systems, the
discrete time signals are represented in digital form with each discrete time sample as binary word.
Therefore we need the analog to digital and digital to analog interface circuits to convert the continuous
time signals into discrete time digital form and vice versa. As a result it is necessary to develop the
relations between continuous time and discrete time representations.

1. Sampling of continuous time signals:


Let {xc(t)} be a continuous time signal that is sampled uniformly at t = nT generating the sequence
{x[n]} where
x[n] = xc(nT ), −∞ < n < ∞, T >0

T is called sampling period, the reciprocal of T is called the sampling fre-quency fs = 1/T . The frequency
domain representation of {xc(t)} is given by its Fourier transform.

where the frequency-domain representation of {x[n]} is given by its discrete time fourier transform.
To establish relationship between the two representation, we use impulse train sampling. This should be
understood as mathematically convenient method for understanding sampling. Actual circuits can not
produce contin-uous time impulses. A periodic impulse train is given by:
xp(t) = xc(t)p(t)

using sampling property of the impulse f (t)δ(t − t0) = f (t0)δ(t − t0), we get

From multiplication property, we know that:

Xp(jΩ) = 2π [Xc(jΩ) ₃ P (jΩ)]


The Fourier transform of a impulse train is given by
Where Ωs = 2T
Using the property that X (jΩ) δ(Ω − Ω0) = X (j(Ω − Ω0)) it follows tha
Xp(jΩ) = 1/t Xc(jΩ − kΩs)
Thus Xp(jΩ) is a periodic function of Ω with period Ωs, consisting of super-position of shifted replicas of
Xc(jΩ) scaled by 1/T . Figure 8.3 illustrates this for two cases.

If Ωm < (Ωs − Ωm) or equivalently Ωs > 2Ωm there is no overlap between shifted replicas of Xc(jΩ),
whereas with Ωs < 2Ωm, there is overlap. Thus if Ωs > 2Ωm, Xc(jΩ) is faithfully replicated in Xp(jΩ) and
can be recovered from xp(t) by means of lowpass filtering with gain T and cut of frequency between Ωm
and ΩsΩm. This result is known as Nyquist sampling theorem.
A major application of discrete-time systems is in the processing of continuous-time signals.

The overall system is equivalent to a continuous-time system, since it transforms the continuous-time
input signal xs(t) into the continuous time signal yr(t).

Sampling Theorem: Let xc(t) be a bandlimited signal with Xc(jΩ) = 0,


for |Ω| > Ωm. Then Xc(t) is uniquely determined by its samples x[n] =
xc(nT), −∞ < n < ∞, if

The frequency 2Ωm is called Nyquist rate, while the frequency Ωm is called
the Nyquist frequency.
The signal xc(t) can be reconstructed by passing xp(t) through a lowpass
filter.

The effect of underselling: Aliasing


We have seen earlier that spectrum Xc(jΩ) is not faithfully copied when Ωs <
2Ωm. The terms in overlap. The signal xc(t) is no longer recoverable

From xp(t). This effect, in which individual terms in equation overlap is called aliasing.
For the ideal low pass signal
Hence xr(nT) = xc(nT), n = 0,±1,±2.......
Thus at the sampling instants the signal values of the original and reconstructed
Signals are same for any sampling frequency.

Source : http://msk1986.files.wordpress.com/2013/09/dsp-unit-1.pdf

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