CAPITAL UNIVERSITY JHARKHAND

DIGITAL SIGNAL PROCESSING

Q1 Ans:
In the context of digital signal processing (DSP), a digital signal is a
discrete time, quantized amplitude signal. In other words, it is a
sampled signal consisting of samples that take on values from a discrete
set (a countable set that can be mapped one-to-one to a subset of
integers).
Q2 Ans:
A system is any process that generates an output signal in response to
an input signal. Continuous signals are usually represented with
parentheses, while discrete signals use brackets.
Q3 Ans:
 Sampling
The audio data we wish to treat will generally be present in the form of
electric oscillations. These can either come from a microphone
recording acoustic sound waves, from a tape, or an electronic
instrument or device. To convert these oscillations to a form that can
be treated by a computer they have to be digitised, i.e. reduced from
an analog form (time- and value-continuous) to digital (time- and value-
discrete). This process is called A/D conversion or sampling .
 Power and Energy
The notion of energy and power for digital signals is only slightly related
to the physical units pertaining to analog electrical signals. They are,
nevertheless, related to the perceived loudness, albeit in a non-trivial
way
  Fourier Transform
After A/D conversion, the signal x(n) is represented in the time domain
as a sequence of numbers. According to Fourier’s ingenious idea , any
function, even nonperiodic ones, can be expressed as a sum of
(periodic) sinusoids (which can not be further decomposed, i.e. they are
``pure’’.
 Convolution, Filtering and Linear Systems
 Windowing.
Q4 ans:
DSP is used primarily in areas of the audio signal, speech processing,
RADAR, seismology, audio, SONAR, voice recognition, and some
financial signals. For example, Digital Signal Processing is used for
speech compression for mobile phones, as well as speech transmission
for mobile phones.
Q6 Ans:
Energy signal is a signal whose energy is finite and power is zero
whereas Power signal is a signal whose power is finite and energy is
infinite.
Q7 Ans:
Discrete-time systems, “A set of connected parts or models which takes
discrete-time signals as input, known as excitation, processes it under
certain set of rules and algorithms to have a desired output of another
discrete-time signal, known as response homogeneous.
Q9 Ans:
A linear system follows the laws of superposition. This law is necessary
and sufficient condition to prove the linearity of the system. Apart from
this, the system is a combination of two types of laws – Law of
additivity. Law of homogeneity.
Q10 Ans:
A static system is a memoryless system. A dynamic system is a system
in which output at any instant of time depends on the input sample at
the same time as well as at other times.
Q11 Ans:
A causal system is one whose output depends only on the present and
the past inputs. A noncausal system’s output depends on the future
inputs. In a sense, a noncausal system is just the opposite of one that
has memory.
Q12 Ans:
Even signals are symmetric around vertical axis, and Odd signals are
symmetric about origin. Even Signal: A signal is referred to as an even if
it is identical to its time-reversed counterparts; x(t) = x(-t). Odd Signal: A
signal is odd if x(t) = -x(-t).
Q13 Ans:
Signals which can be defined exactly by a mathematical formula are
known as deterministic signals. A signal is said to be non-deterministic if
there is uncertainty with respect to its value at some instant of time.
Non-deterministic signals are random in nature hence they are called
random signals.
Q14 Ans:
A signal is said to be periodic signal if it has a definite pattern and
repeats itself at a regular interval of time. Whereas, the signal which
does not at the regular interval of time is known as an ’aperiodic signal
or non-periodic signal.
Q16 Ans:
A system is called a stable system in case a finite change in the input
causes a finite change in the output. Examples of unstable systems: a
ball on top of an sphere. In theory you can put the small ball on the top
of the sphere such that its center gravity is right above the center of the
sphere.
Q17 Ans:
A system is said to be time invariant if the response of the system to an
input is not a function of time. On the other hand a system is time
variant if the response to an input alters with time i.e. the system has
varying response to the same input at different instants of time.
Q18 Ans:
A recursive system is a system in which current output depends on
previous output(s) and input(s) but in non-recursive system current
output does not depend on previous output(s).
Q21 Ans:
In mathematics and signal processing, the Z-transform converts a
discrete-time signal, which is a sequence of real or complex numbers,
into a complex frequency-domain representation. It can be considered
as a discrete-time equivalent of the Laplace transform.
Q23 Ans:
Region of Convergence
Region of Convergence is the range of complex variable Z in the Z-
plane. The Z- transformation of the signal is finite or convergent. So,
ROC represents those set of values of Z, for which XZ has a finite value.
Q24 Ans:
Linearity
It states that when two or more individual discrete signals are
multiplied by constants, their respective Z-transforms will also be
multiplied by the same constants.
Time Shifting
Time shifting property depicts how the change in the time domain in
the discrete signal will affect the Z-domain.
Time Scaling
Time Scaling property tells us, what will be the Z-domain of the signal
when the time is scaled in its discrete form.
Successive Differentiation
Successive Differentiation property shows that Z-transform will take
place when we differentiate the discrete signal in time domain, with
respect to time.
Q26 Ans:
Properties of Linear Convolution
Commutative Law: (Commutative Property of Convolution) x(n) * h(n) =
h(n) * x(n)
Associate Law: (Associative Property of Convolution)
Distribute Law: (Distributive property of convolution) x(n) * [ h1(n) +
h2(n) ] = x(n) * h1(n) + x(n) * h2(n)
Q27 Ans:
The Discrete-Time Fourier Transform (DTFT) is the cornerstone of all
DSP, because it tells us that from a discrete set of samples of a
continuous function, we can create a periodic summation of that
function’s Fourier ttransform
Q29 Ans:
Properties of the DTFT
Linearity.
Symmetry.
Time Scaling.
Time Shifting.
Convolution.
Time Differentiation.
Parseval’s Relation.
Modulation (Frequency Shift)
Q31 Ans:
The discrete Fourier transform (DFT) converts a finite sequence of
equally-spaced samples of a function into a same-length sequence of
equally-spaced samples of the discrete-time Fourier transform (DTFT),
which is a complex-valued function of frequency.
Q32 Ans:
An N-point DFT is expressed as the multiplication , where is the original
input signal, is the N-by-N square DFT matrix, and. Is the DFT of the
signal.
Q34 Ans:
Properties of Discrete Fourier Transform(DFT)
PROPERTIES OF DFT.
Periodicity.
Linearity.
Circular Symmetries of a sequence.
Symmetry Property of a sequence.
 Symmetry property for real valued x(n) i.e xI(n)=0.
Circular Convolution.
Multiplication.
Q35 Ans:
Linearity
The transform of a sum is the sum of the transforms: DFT(x+y) = DFT(x)
+ DFT(y). Likewise, a scalar product can be taken outside the transform:
DFT(c*x) = c*DFT(x). These follow directly from the fact that the DFT
can be represented as a matrix multiplication.
Q37 Ans:
For the computation of N-point DFT, N2 complex multiplications and
N[N-1] Complex additions are required. If the value of N is large than
the number of into lakhs. This proves inefficiency of direct DFT
computation.
Q41 Ans:
Infinite impulse response is a property applying to many linear time-
invariant systems that are distinguished by having an impulse response
h(t) which does not become exactly zero past a certain point, but
continues indefinitely.
Q43 Ans:
The Impulse Invariance Method is used to design a discrete filter that
yields a similar frequency response to that of an analog filter. Discrete
filters are amazing for two very significant reasons: You can separate
signals that have been fused and, You can use them to retrieve signals
that have been distorted.
Q44 Ans:
Disadvantages of bi-linear transformation method :
The mapping is non linear in this method because of this frequency
warping effect takes place
Q45 Ans:
Prewarping. Frequency warping follows a known pattern, and there is a
known relationship between the warped frequency and the known
frequency. We can use a technique called prewarping to account for
the nonlinearity, and produce a more faithful mapping.
Q49 Ans:
The frequency response of the Butterworth filter is maximally flat (i.e.
has no ripples) in the passband and rolls off towards zero in the
stopband. When viewed on a logarithmic Bode plot, the response
slopes off linearly towards negative infinity.
Q52 Ans:
This is useful for illustrating how the filter causes impulsive inputs to
spread out in the time domain. The magnitude and phase plots show
and , respectively, plotted with so that the frequency ranges from up
to , which are the minimum and maximum frequencies representable in
a digital system.
Q53 Ans:
The main difference between analog and digital filters is that analog
filters process analog signals directly, whereas digital filters need to first
convert analog signals to digital signals, before processing. After
processing, the signal needs to be converted again from digital to
analog signals.
Q54 Ans:
Advantages of digital filter :
Digital filter has characteristic like linear phase response.
The performance of the digital filter does not vary with environmental
parameters.
Digital filters used at very low frequencies, for example in a biomedical
application so-called as an adaptive filter because the frequency
response can be possible to adjust automatically with an
implementation of the programmable processor.
The digital filters are portable.
Disadvantages of digital filter :
The signal bandwidth of the input signal is limited by ADC and DAC.
The bandwidth of the digital filter is much lower than an analogue filter.
The accuracy of the digital filter depends on the word length used to
encode them in binary form.
It required more design and development time compared to an
analogue filter.
Q55 Ans:
Impulse invariance is a technique for designing discrete-time infinite-
impulse-response (IIR) filters from continuous-time filters in which the
impulse response of the continuous-time system is sampled to produce
the impulse response of the discrete-time system.
Q57 Ans:
Poles represent frequencies that cause the denominator of a transfer
function to equal zero, and they generate a reduction in the slope of
the system’s magnitude response.
Q59 Ans:
The bilinear transformation is a mathematical mapping of variables. In
digital filtering, it is a standard method of mapping the s or analog
plane into the z or digital plane. It transforms analog filters, designed
using classical filter design techniques, into their discrete equivalents.

Q63 Ans:
In signal processing, a finite impulse response (FIR) filter is a filter
whose impulse response (or response to any finite length input) is of
finite duration, because it settles to zero in finite time. This is in
contrast to infinite impulse response (IIR) filters, which may have
internal feedback and may continue to respond indefinitely (usually
decaying).
FIR filters can be discrete-time or continuous-time, and digital or
analog.
Q64 Ans:
There are two fundamental types of digital filters: finite impulse
response (FIR) and infinite impulse response (IIR). As the terminology
suggests, these classifications refer to the filter’s impulse response.
Q65 Ans:
The five main types of frequency filters are the high pass, low-pass, all-
pass, band pass, and notch filters. Their characteristics are determined
by the type and values of circuit components used as well as their
arrangement. The classification is based on the frequency range that a
filter allows to passes through.
Q66 Ans:
 Windowing
Apply window to truncated inverse Fourier transform of specified
“brick wall” filter
 Multiband with Transition Bands
Equiripple or least squares approach over sub-bands of the
frequency range.
 Constrained Least Squares
Minimize squared integral error over entire frequency range
subject to maximum error constraints
 Arbitrary Response
Arbitrary responses, including nonlinear phase and complex filters
 Raised Cosine
Lowpass response with smooth, sinusoidal transition
Q67 Ans:
Ans: A digital filter is causal if its impulse response h (n) =0 for n<0. A
digital filter is stable if its impulse response is absolutely summable.
Q68 Ans:
The necessary and sufficient condition for IIR filters to be stable is that
all poles are inside the unit circle. In contrast, FIR filters are always
stable because the FIR filters do not have poles. You can determine if
pole-zero pairs are close enough to cancel out each other effectively.
Q70 Ans:
The main reason behind the cause delay distortion is that the frequency
and velocity of the signal in-network medium do not match up. Due to
this, there occurs a delay in the signals across the network and they
appear to be significantly distorted.
In signal processing, phase distortion or phase-frequency distortion is
distortion, that is, change in the shape of the waveform, that occurs
when (a) a filter’s phase response is not linear over the frequency range
of interest, that is, the phase shift introduced by a circuit or device is
not directly proportional to frequency, or (b) the zero-frequency
intercept of the phase-frequency characteristic is not 0 or an integral
multiple of 2π radians.
Q72 Ans:
They can easily be designed to be “linear phase” (and usually are).
They are simple to implement.
They are suited to multi-rate applications.
They have desirable numeric properties.
They can be implemented using fractional arithmetic.
Q73 Ans:
The primary disadvantage of FIR filters is that they often require a much
higher filter order than IIR filters to achieve a given level of
performance. Correspondingly, the delay of these filters is often much
greater than for an equal performance IIR filter.
Q78 Ans:
There are essentially three well-known methods for FIR filter design
namely: The window method 1. The frequency sampling technique 2.
Optimal filter design methods.
Q79 Ans:
Gibbs phenomenon Is the oscillatory behavior of the Fourier series of a
piecewise continuously differentiable periodic function around a jump
discontinuity. The function’s Nth partial Fourier series (formed by
summing its N th lowest constituent sinusoids) produces large peaks
around the jump which overshoot and undershoot the function’s actual
values. This approximation error approaches a limit of about 9% of the
jump as more sinusoids are used, though the infinite Fourier series sum
does eventually converge almost everywhere except the point of
discontinuity.
The Gibbs phenomenon was observed by experimental physicists, but
was believed to be due to imperfections in the measuring apparatus,
and it is one cause of ringing artifacts in signal processing.
Q80 Ans:
In an ideal window function the: Main lobe width is small (high-
frequency resolution) Side lobe level is high (good noise suppression,
high detection ability) Side lobe roll-off rate is high.
Q81 Ans:
CLASSIFICATION OF SIGNAL PROCESSING
 1) ASP (Analog signal Processing) : If the input signal given to the
system is analog then system does analog signal processing. Ex
Resistor, capacitor or Inductor, OP-AMP etc
2) DSP (Digital signal Processing) : If the input signal given to the
system is digital then system does digital signal processing. Ex
Digital Computer, Digital Logic Circuits etc. The devices called as
ADC (analog to digital Converter) converts Analog signal into
digital and DAC (Digital to Analog Converter) does vice-versa.
Most of the signals generated are analog in nature. Hence these signals
are converted to digital form by the analog to digital converter. Thus
AD Converter generates an array of samples and gives it to the digital
signal processor. This array of samples or sequence of samples is the
digital equivalent of input analog signal. The DSP performs signal
processing operations like filtering, multiplication, transformation or
amplification etc operations over these digital signals. The digital
output signal from the DSP is given to the DAC.
Q82 Ans:
Factors of influencing DSP
 Arithmetic Format.
 Data Width.
 Speed.
 Memory Organization.
 Ease of Development.
 Multiprocessor Support.
 Power Consumption and Management.
 Cost.
Q85 Ans:
The Harvard architecture is a computer architecture with separate
storage and signal pathways for instructions and data. It contrasts with
the von Neumann architecture, where program instructions and data
share the same memory and pathways.
The term originated from the Harvard Mark I relay-based computer,
which stored instructions on punched tape (24 bits wide) and data in
electro-mechanical counters. These early machines had data storage
entirely contained within the central processing unit, and provided no
access to the instruction storage as data. Programs needed to be
loaded by an operator; the processor could not initialize itself.
Modern processors appear to the user to be von Neumann machines,
with the program code stored in the same main memory as the data.
For performance reasons, internally and largely invisible to the user,
most designs have separate processor caches for the instructions and
data, with separate pathways into the processor for each. This is one
form of what is known as the modified Harvard architecture.
Q86 Ans:

Q87 Ans:
Q88 Ans:
Block diagram of VLIW Architecture
Timing Space Diagram of VLIW Architecture

Q89 Ans:
There are three types of MAC addresses, which are:
Unicast MAC Address
Multicast MAC address
Broadcast MAC address
Q91 Ans:
The number of overlapable operations of which an instruction is
comprised is known as the depth of the pipeline. The minimum depth is
three (fetch, decode, execute), typical values are four or five, but by
dividing the arithmetic operation into stages the maximum depth may
be larger.