MODULE-1_ DSP Algorithm and Architecture – 21EC723

DSP Algorithm and Architecture

Course Code: 21EC723

Module-1

Introduction to Digital Signal Processing

 Introduction, A Digital Signal – Processing system,

 Major features of programmable Digital signal processors,

 The Sampling Process,

 Discrete Time Sequences,

 Discrete Fourier Transform (DFT) and Fast Fourier Transform (FFT),

 Linear Time-Invariant Systems,

 Digital Filters,

 Decimation and Interpolation.

Section 1.3, 2.1 to 2.8 of Text 1

Text book referred: Digital Signal Processing”, Avatar Singh and S Srinivasan, Thomson Learning,
2004 .

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DSP Algorithm and Architecture 21EC723

UNIT-1

Introduction to Digital Signal Processing

1.1 What is DSP?

DSP is a technique of performing the mathematical operations on the signals in digital domain.
As real time signals are analog in nature we need first convert the analog signal to digital, then we
have to process the signal in digital domain and again converting back to analog domain. Thus ADC is
required at the input side whereas a DAC is required at the output end. A typical DSP system is as
shown in figure 1.1.

1.2 Need for DSP

Analog signal Processing has the following drawbacks:

 They are sensitive to environmental changes
 Aging
 Uncertain performance in production units
 Variation in performance of units
 Cost of the system will be high
 Scalability
If Digital Signal Processing would have been used we can overcome the above shortcomings of ASP.

1.3 A Digital Signal Processing System

A computer or a processor is used for digital signal processing. Anti aliasing filter is a LPF
which passes signal with frequency less than or equal to half the sampling frequency in order to avoid
Aliasing effect. Similarly at the other end, reconstruction filter is used to reconstruct the samples from
the staircase output of the DAC (Figure 1.2).

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1.4 The Sampling Process

ADC process involves sampling the signal and then quantizing the same to a digital value. In
order to avoid Aliasing effect, the signal has to be sampled at a rate at least equal to the Nyquist rate.

Where, fs is the sampling frequency, fm is the maximum frequency component in the message
signal. If the sampling of the signal is carried out with a rate less than the Nyquist rate, the higher
frequency components of the signal cannot be reconstructed properly. The plots of the reconstructed
outputs for various conditions are as shown in figure 1.4.

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1.5 Discrete Time Sequences

Sampling Interval T, in the above equation replacing t by nT we get, x (nT) = A cos (2

where n= 0,1, 2,..etc
For simplicity denote x (nT) as x (n)
 x (n) = A cos (2πfnT) where n= 0,1, 2,..etc
We have fs=1/T also θ ΠfnT
 πfnT)= A cos (2πfn/fs) = A cos πn
θ called as digital frequency.
θ = 2πfT = 2πf/fs radians

Fig 1.5 A Cosine Waveform

A sequence that repeats itself after every period N is called a periodic sequence.
Consider a periodic sequence x (n) with period N x (n)=x (n+N) n=……..,-1,0,1,2,……..
Frequency response gives the frequency domain equivalent of a discrete time sequence. It is denoted
as X(ejθ)=∑x(n) e-jnθ

Frequency response of a discrete sequence involves both magnitude response and phase response.

1.6 Discrete Fourier Transform and Fast Fourier Transform

1.6.1 DFT Pair:

DFT is used to transform a time domain sequence x (n) to a frequency domain sequence X
(K).The equations that relate the time domain sequence x (n) and the corresponding frequency domain
sequence X (K) are called DFT Pair and is given by,

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1.6.2 The Relationship between DFT and Frequency Response:

We have,

From the above expression it is clear that we can use DFT to find the Frequency response of a

the signal record length.

It is clear from the expression of
samples N has to be a large value. Although DFT is an efficient technique of obtaining the frequency
response of a sequence, it requires more number of complex operations like additions and
multiplications.
Thus many improvements over DFT were proposed. One such technique is to use the
periodicity property of the twiddle factor e- . Those algorithms were called as Fast Fourier
Transform Algorithms. The following table depicts the complexity involved in the computation using
DFT algorithms.

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FFT algorithms are classified into two categories via

1. Decimation in Time FFT
2. Decimation in Frequency FFT
In decimation in time FFT the sequence is divided in time domain successively till we reach
the sequences of length 2. Whereas in Decimation in Frequency FFT, the sequence X(K) is divided
successively. The complexity of computation will get reduced considerably in case of FFT algorithms.

1.7 Linear Time Invariant Systems

A system which satisfies superposition theorem is called as a linear system and a system that
has same input output relation at all times is called a Time Invariant System. Systems, which satisfy
both the properties, are called LTI systems.

LTI systems are characterized by its impulse response or unit sample response in time domain whereas
it is characterized by the system function in frequency domain.

1.7.1 Convolution
Convolution is the operation that related the input output of an LTI system, to its unit sample
response. The output of the system y (n) for the input x (n) and the impulse response of the system

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being h (n) is given as y (n) = x(n) * h(n) = ∑ -k), x(n) is the input of the system, h(n) is the
impulse response of the system, y(n) is the output of the system.

1.7.2 Z Transformation
Z Transformations are used to find the frequency response of the system. The Z Transform for
a discrete sequence x (n) is given by, X(Z)= ∑x(n) z-n

1.7.3 The System Function

An LTI system is characterized by its System function or the transfer function. The system
function of a system is the ratio of the Z transformation of its output to that of its input. It is denoted as
H (Z) and is given by H (Z) = Y (Z)/ X (Z).
The magnitude and phase of the transfer function H (Z) gives the frequency response of the
system. From the transfer function we can also get the poles and zeros of the system by solving its
numerator and denominator respectively.

1.8 Digital Filters

Filters are used to remove the unwanted components in the sequence. They are characterized
by the impulse response h (n). The general difference equation for an Nth order filter is given by
∑aky(n-k)+ ∑ k x(n-k)
A typical digital filter structure is as shown in figure 1.7.

Fig 1.7 Structure of a Digital Filter

Values of the filter coefficients vary with respect to the type of the filter. Design of a digital filter
involves determining the filter coefficients. Based on the length of the impulse response, digital filters
are classified into two categories via Finite Impulse Response (FIR) Filters and Infinite Impulse
Response (IIR) Filters.

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1.8.1 FIR Filters

FIR filters have impulse responses of finite lengths. In FIR filters the present output depends
only on the past and present values of the input sequence but not on the previous output sequences.
Thus they are non recursive hence they are inherently stable.FIR filters possess linear phase response.
Hence they are very much applicable for the applications requiring linear phase response.
The difference equation of an FIR filter is represented as

The frequency response of an FIR filter is given as

The major drawback of FIR filters is, they require more number of filter coefficients to realize a
desired response as compared to IIR filters. Thus the computational time required will also be more.

1.8.2 IIR Filters

Unlike FIR filters, IIR filters have infinite number of impulse response samples. They are
recursive filters as the output depends not only on the past and present inputs but also on the past
outputs. They generally do not have linear phase characteristics. Typical system function of such
filters is given by,

Stability of IIR filters depends on the number and the values of the filter coefficients. The major
advantage of IIR filters over FIR is that, they require lesser coefficients compared to FIR filters for the
same desired response, thus requiring less computation time.

1.8.3 FIR Filter Design

Frequency response of an FIR filter is given by the following expression,

Design procedure of an FIR filter involves the determination of the filter coefficients bk.

1.8.4 IIR Filter Design

IIR filters can be designed using two methods viz using windows and direct method. In this
approach, a digital filter can be designed based on its equivalent analog filter. An analog filter is
designed first for the equivalent analog specifications for the given digital specifications. Then using
appropriate frequency transformations, a digital filter can be obtained. The filter specifications consist
of passband and stopband ripples in dB and Passband and Stopband frequencies in rad/sec.

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Fig 1.11 Lowpass Filter Specifications

Direct IIR filter design methods are based on least squares fit to a desired frequency response. These
methods allow arbitrary frequency response specifications.

1.9 Decimation and Interpolation

Decimation and Interpolation are two techniques used to alter the sampling rate of a sequence.
Decimation involves decreasing the sampling rate without violating the sampling theorem whereas
interpolation increases the sampling rate of a sequence appropriately by considering its neighboring
samples.

1.9.1 Decimation
Decimation is a process of dropping the samples without violating sampling theorem. The
factor by which the signal is decimated is called as decimation factor and it is denoted by M. It is
given by,

Fig 1.12 Decimation Process

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1.9.2 Interpolation
Interpolation is a process of increasing the sampling rate by inserting new samples in between.
The input output relation for the interpolation, where the sampling rate is increased by a factor L, is
given as,

Fig 1.13 Interpolation Process

Problems:

1. Obtain the transfer function of the IIR filter whose difference equation is given by y (n)=
0.9y (n-1)+0.1x (n)
y (n)= 0.9y (n-1)+0.1x (n)
Taking Z transformation both sides
Y (Z) = 0.9 Z-1 Y (Z) + 0.1 X (Z)
Y (Z) [1- 0.9 Z-1] = 0.1 X (Z)
The transfer function of the system is given by the expression,
H (Z)= Y(Z)/X(Z)
= 0.1/ [ 1- 0.9 Z-1]
Realization of the IIR filter with the above difference equation is as shown in figure.

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DSP Algorithm and Architecture 21EC723

2. Let x(n)= [0 3 6 9 12] be interpolated with L=3. If the filter coefficients of the
filters are bk=[1/3 2/3 1 2/3 1/3], obtain the interpolated sequence

After inserting zeros,

w (m) = [0 0 0 3 0 0 6 0 0 9 0 0 12]
bk=[1/3 2/3 1 2/3 1/3]
We have,
y(m)= -k) = b-2 w(m+2)+ b-1 w(m+1)+ b0 w(m)+ b1 w(m-1)+ b2 w(m-2)
Substituting the values of m, we get
y(0)= b-2 w(2)+ b-1 w(1)+ b0 w(0)+ b1 w(-1)+ b2 w(-2)= 0
y(1)= b-2 w(3)+ b-1 w(2)+ b0 w(1)+ b1 w(0)+ b2 w(-1)=1
y(2)= b-2 w(4)+ b-1 w(3)+ b0 w(2)+ b1 w(1)+ b2 w(0)=2
Similarly we get the remaining samples as,
y (n) = [ 0 1 2 3 4 5 6 7 8 9 10 11 12]

Recommended Questions
1. Explain with the help of mathematical equations how signed numbers can be
multiplied. The sequence x(n) = [3,2,-2,0,7].It is interpolated using interpolation
sequence bk=[0.5,1,0.5] and the interpolation factor of 2. Find the interpolated
sequence y(m).
2. An analog signal is sampled at the rate of 8KHz. If 512 samples of this signal are used
to compute DFT X(k) determine the analog and digital frequency spacing between
adjacent X(k0 elements. Also, determine analog and digital frequencies corresponding
to k=60.
3. With a neat diagram explain the scheme of the DSP system.
4. What is DSP? What are the important issues to be considered in designing and
implementing a DSP system? Explain in detail.
5. Why signal sampling is required? Explain the sampling process.
6. Define decimation and interpolation process. Explain them using block diagrams and
equations. With a neat diagram explain the scheme of a DSP system.
7. With an example explain the need for the low pass filter in decimation process.
8. For the FIR filter y(n)=(x(n)+x(n-1)+x(n-2))/3. Determine i) System Function ii)
Magnitude and phase function iii) Step response iv) Group Delay.
9. List the major architectural features used in DSP system to achieve high speed program
execution.

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DSP Algorithm and Architecture 21EC723

10. Explain how to simulate the impulse responses of FIR and IIR filters.
11. Explain the two method of sampling rate conversions used in DSP system, with suitable
block diagrams and examples. Draw the corresponding spectrum.
12. Assuming X(K) as a complex sequence determine the number of complex real
multiplies for computing IDFT using direct and Radix-2 FT algorithms.
13. With a neat diagram explain the scheme of a DSP system. (June.12, 8m)
14. With an example explain the need for the low pass filter in decimation process.
(June.12, 4m)
15. For the FIR filter y(n)=(x(n)+x(n-1)+x(n-2))/3. Determine i) System Function ii)
Magnitude and phase function iii) Step response iv) Group Delay. (June.12, 8m)
16. List the major architectural features used in DSP system to achieve high speed program
execution. (Dec.11, 6m).
17. Explain how to simulate the impulse responses of FIR and IIR filters. (Dec.11, 6m).
18. Explain the two method of sampling rate conversions used in DSP system, with suitable
block diagrams and examples. Draw the corresponding spectrum. (Dec.11, 8m).
19. Explain with the help of mathematical equations how signed numbers can be
multiplied. (July.11, 8m).

20. With a neat diagram explain the scheme of the DSP system. (Dec.10-Jan.11, 8m)
(July.11, 8m).

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