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DEPARTMENT OF ECE
Structures of IIR – Analog filter design – Discrete time IIR filter from analog
filter – IIR filter design by Impulse Invariance, Bilinear transformation,
Approximation of derivatives – (HPF, BPF, BRF) filter design using frequency
translation
Structures of FIR – Linear phase FIR filter – Filter design using windowing
techniques, Frequency sampling techniques – Finite word length effects in digital
Filters
UNIT V APPLICATIONS
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CS2403-DIGITAL SIGNAL PROCESSING DEPARTMENT OF CSE
PART A (Q&A)
Q. 1. What is DSP?
Ans. The digital signal processing systems have many advantages. Even though there are
certain disadvantages as follows
1. Bandwidth limitations : In case of DSP, if input signal is having wide bandwidth then it
demands for high speed ADC. This is because to avoid aliasing effect, the sampling rate
should be atleast twice the bandwidth. Thus such signals require fast digital signal
processors. But always there is a practical limitation in the speed of processors and ADC.
2. System complexity : The digital signal processing system makes use of converters like
ADC and DAC. This increases the system complexity compared to analog systems.
Similarly in many applications the time required for this conversion is more.
3. Power Consumption: A typical digital signal processing chip contains more than 4 lakh
transistors. Thus power dissipation is more in caps systems compared to analog systems.
2. Instrumentation and control like spectral analysis, noise reduction, data compression.
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7. Consumer applications like digital audio and video, power like monitor.
Ans. A ‘signal’ may be defined as a physical quantity which varies with time, space or
any independent variable Example — voltage, current A ‘system may be defined as a
combination of devices and networks or subsystem chosen to do a desired action
Example Electrical N/W, mechanical system
Ans. There are various types of signals Every signal is having its own characteristic The
processing of signal mainly depends on the characteristics of that particular signal So
classification of signal is necessary Broadly the signal are classified as follows
Ans In mathematics, the sinc, function, denoted by sinc(x) and sometimes as Sa (x),
has two definitions, In digital processing and information theory, the normalized sinc
function is commonly defined by
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In both cases, the value of the function at the removable singularity at zero, usually
calculated by l’Hospital rule, is something specified explicitly as the limit value 1 The
sinc function is analytic everywhere.
Many signals that posses infinite energy, have a finite average power. The average
either finite or infinite. If P is finite (and non zero), the signal is called a power signal.
Ans. A system is called linear, if superposition principle applies to that system. This
means that linear system may be defined as one whose response to the sum of the
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Here x1(t) is the input or excitation and y1(t) is its output or response and
Here x2 (t) is the input or excitation and y2(t) is its output or response
Linearity property for both continuous time and discrete time systems may be written
For any non-linear system, the principle of super-position does not hold true and
Q. 9. Define periodic and non periodic signals Give an example in each case.
Ans A periodic signal repeats after fixed period But non-periodic signal never repeats
Periodic signal like x(t) sin wt and Non periodic signal like A discrete time
signal is periodic, if its frequency can be expressed as a ratio of two integers i.e.
Here k and N are integer and N is the period of discrete time signal
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As the name indicates, time scaling operations are related to the change in time scale.
There are two types of time scaling operations.
Up scaling (Expansion)
• Upscaling (Amplification)
• Downscahng (Attenuation)
• Addition
• Multiplication
Q 11 What is the difference between static and dynamic discrete time signals?
Ans. There can be static and dynamic discrete time systems but cannot be signals
Ans. A discrete time unit signal is denoted by U(n) Its value is unity for all positive
values of n. That means its value is one for n 0. While for other values of n, its value is
zero.
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Ans. A discrete time unit ramp function is denoted as Ur (n) and it is defined as
Figure below shows the graphical representation of a discrete unit ramp function.
Ans. A system may be defined as a set of elements or functional blocks which are
connected together and produces an output in response to an input signal. The response of
the system depends upon transfer function of the system.
Mathematically it is defined by
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Q. 15. State the necessary and sufficient condition for stability of LTI systems
Here h(k)= h(n) is the impulse response of LTI system Thus equation (1) give the
Q. 16. What are the constraints on the transfer function if it were to represent
a causal LTI system?
Ans. If h(n) is the response of released LTI system to a unit impulse applied at n = 0, it
follows that h(n) = 0 for n < 0 is both a necessary and a sufficient condition for causality
Hence on LTI system is causal if and only if its impulse response is zero for negative
values of n.
Ans. If a system has both the linearity and time in varience properties, then the system is
called as linear time m varient (LTI) system
Q 18. What are the conditions for the region of convergence of a causal LTI system?
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Ans. A discrete time LTI system is causal if and only if the ROC of its transfer function is
the extension of a circle, in including infinite
A discrete time LTI systems which has a rational transfer function H(z) will be causal if
and only if.
(z) The ROC is the extension of a circle outside the outermost pole and
(ii) Units H(z) expressed as a ratio of polynomials in z, the order of the numerator should
be smaller than order of denomenator.
Ans. A continuous time signal x(t) can be completely respresented in its sampled form
and recoverd back from the sample form if the sampling frequency
where ‘W’ is the maximum frequency of the continuous time signal x(t)
Ans.
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Ans.
Ans. A system is called time invariant if its input output characteristics do not charge
with time. A LTI discrete time system satisfies boths the linearity and the time invariance
properties.
To test if any given systems is time invariant, first apply an arbitrary sequence x (n) and
find y (n).
y (n) = T [x (n)]
Now delay the input sequence by k samples and find output sequence denote it as. y(n,k)
T[x(n-k)]
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For all possible values of k, the systems is the invariant on the other hand
the output.
Ans. An even signal is that type of signal which exhibits symmetry in the time domain
This type of signal is identical about the origin Mathematically, an even signal must
satisfy the following condition.
Similarly, an odd signal is that type of signal which exhibits anti-symmetry. This type of
signal is not identical about the origin Actually, the signal is identical to its negative
Mathematically, an odd signal must satisfy the following condition
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Q. 24. What is the frequency response of a discrete LTI system? Derive the
frequency response of a system whose impulse response is given by h(n) = a” u(n —
1) for (a) <1.
Ans. The frequency response of a linear time invariant discrete time system can be
obtained by applying a spectrum of the input sinusoids to the system. The frequency
response gives the gain and phase response of the system to the input sinusoids at all
frequencies. Let us consider, the inpulse response of an LTI discrete time system is h(n)
and the input x(n) to the system is complex exponential e1u. The output of the system
y(n) can be
Given
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Q. 25. Determine the power and energy of the unit step sequence.
Consequently, the unit step sequence is a power signal. Its energy is infinite.
Q. 26. Sketch the block diagram representation of the discrete time system
described by the input-output relation.
where x(n) is the input and y (n) is the output of the system.
Ans. According to the question, the output y(n) is obtained by multiplying the input x(n)
by 0.5, multiplying the previous input x(n -1) by 0.5, adding the two products and then
adding the previous output y(n -1) multiplied by 1/4. Figure below illustrates this block
diagram realization of the system. A simple rearrangement of equation is
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Q. 27. Determine if the system y(n) = x(— n) is time variant or time invariant.
Now if we delay the output y (n), by k units in time, the result will be
Q. 28. Determine if the system described by the following input-output equation are
linear or nonlinear i.e.
Ans. For two input sequences x1(n) and x2(n), the corresponding outputs are
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By comparing (i) with (ii), we can conclude that the system is linear.
Q. 29. Determine the system y(n) = x (2n) is causal or non causal system.
Q. 30. Consider the special case of a finite duration sequence given as:
Ans. Sequence x(n) is non-zero for the time instants n = - 1,0,2, we need three impulses
at delay k = — 1, 0 ,2
Ans.
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Ans. A system is causal if the response or output does not begin before the input function
is applied. This means that if input is applied at t=t0 then for.
Causal systems are physically realizable. On the other hand if the response of the system
to an input depends on the future values of that input, then the system is non- causal or
anticipatory. Non-causal systems cannot be implemented practically. This means that
there is no system possible practically which can produce its output before input is
applied.
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Ans. The Z-transform of a discrete time signal x(n) is defined as the power series
The z-transform is a infinite power series, it exists only for those values of z for
which this series converges. The region of convergence (ROC) of X(z) is the set of all s
values ofz for which X(z) attains a finite value. Thus any time we cite a z-transform.
We should also indicate its ROC.
Ans. The z-transform is an infinite power series, it exists only for those values of z
for which the series converges. The region of convergence (ROC) of X (z) is set of all
values of z for which X (z) attains a finite value. The ROC of a finite duration signal is
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the entire z-plane, except possibly the point . These points are excluded
because z-n (when n > 0) becomes unbounded for z = and z n (when n > 0) becomes
unbounded for z = 0.
Ans. Let us consider a sequence x(n) having z-transforrn with ROC that includes the
unit circle. If X(z) is sampled at the N equally spaced points on the unit circle. If X(z) is
We obtain
Expression is (2) identical to the Fourier transform X(w) evaluated at the N. equally
spaced. Frequencies
If the sequence x(n) has a finite duration of length N or less, the sequence can be
recovered from its N-point DFT. Hence its Z-transform is uniquely determined by its N-
point DFI’. Consequently, X(z) can be expressed as a function of the DFT {X(k)} as
follows
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When evaluated on the unit circle (3) yields the Fourier transform of the finite
duration sequence in terms of its DFT in the form:
This expression for Fourier transform is a polynomial interpolation formula for X(w)
expressed in terms of the, values {x(k)) of the polynomial at a set of equally spaced
discrete frequencies
2. It is used for the analysis of discrete time systems in frequency domain which in
generally more efficient than time domain analysis.
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Q. 37. What are the conditions for the region of convergence of a non causal LTI
system.
Ans. The condition for non-causal of discrete time LTI system is that the impulse
response of a causal discrete time LTI system js given as
Ans. The convolution property for Z-transforms is very important for systems
analysis and design. In words : The transform of the convolution is the product of the
transforms.
Proof. This is somewhat easier (and more general) to prove for noncausal sequences.
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Q.39. State the correlation property of two sequence in z-domain. Give its ROC.
Ans. Correlation is a measure of the degree to which two signals are similar. The
Q. 40. Find out the Z-transform for the following discrete time sequence.
Ans.
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Q. 41. Determine to z-transform of the following signal and sketch the pole zero
pattern:
Ans.
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Q. 43, What are the various methods to find out inverse z transform?
Ans.
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Q. 46. Determine the pole-zero plot for the system described by difference equation
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Where T = 1/fs is the sampling period (in units of time e.g. seconds) and fs is the
Let
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This is precisely the definition of the unilateral Z-transform of the discrete function x[n]
Comparing the last two equations, we find the relationship between the unilateral Z-
transform and the Laplace transform of the sampled signal:
Q. 49. Find out the Z-transform for the following discrete time sequence
Ans.
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Q. 1. Define DFT.
where k = 0, 1, 2 N—I.
Ans. The Discrete Time Fourier Transform of a discrete line signal x(n)
is expressed as
DTFT is periodic units period . So any interval of length is sufficient for the
Ans. If
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Comparing each summation term with definition of DTFT then we can write
Q. 4 Explain the symmetry properties of DFTs which provide basis for fast
algorithms.
Ans. Most approaches for improving the efficiency of computation of DFT, exploits
Ans. The process of lengthening a sequence by adding zero valued samples is called
appending with zeros or zero padding. This is done to equate linear convolution units
circular convolutions in case of DFT.
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Ans. The quantization errors in the direct computation of DFT is in particular the effect
of round off errors due to the multiplications performed in the DFT with fixed point
arithmetic, for example.
[x(n)] is a complex valued sequence. Assume that the real and imaginary components
of x(n) and are represented by b bits. Consequently the computation of the product
x(n) W requires four real multiplications. Each real multiplication is rounded from 2b bits
to b bits and hence there are four quantization errors for each complex valued
multiplication.
In the direct computation of the DFT, there are N complex valued multiplications for each
point in the DFT. Therefore the total number of real multiplications in the computation of
a single point in the DFT is 4N. Consequently there are 4N quantization errors.
Ans. The main advantage of in-place computation is reduction in the memory size
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‘a’ & ‘b’ are inputs and ‘A’ and ‘B’ are outputs of butterfly. For anyone input ‘a’ and ‘b’
two memory locations are required for each. One memory location to store real part and
other memory location to store imagining part. So for both inputs ‘a’ & ‘b’ = 2 + 2 = 4
memory location are required.
Thus outputs ‘A’ & ‘B’ are calculated by using the values ‘a’ & ‘b’ stored inmemory.
Once the computation of ‘A’ & ‘B’ done then values of ‘a’ & ‘b’ are not required. Instead
of storing ‘A’ & ‘B’ at other memory locations, there values are stored at the same place
where ‘a’ & ‘b’ were stored. That means ‘A’ & ‘B’ are stored in the place of ‘a’ & ‘b’.
This is called as in-place computation.
Ans.
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Ans. Linear if itering, correlation analysis and spectrum analysis are same
important applications of FFT algorithm.
(a) Find the Fourier Transform X(cv). (b) Find the N-point DFT
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Ans.
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n=0 n=0
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N2 = 642 = 4096
Number of complex multiplications required using FFT is
(N/2) log N = ((64/2) log 64 = 192
speed improvement factor (4096/192) = 21.33.
Q.22. Define discrete linear convolution.
The discrete convolution of the two discrete variable function x(n) and h(n) is the
discrete variable function y(n) given by the summation
∞
y(n)= ∑ x(k) h(n-k)
k=-∞
Q.23. What are the properties of DIT FFT?
1.Computation are done in place. Once a butterfly structure operation is performed on a
pair of complex numbers(a,b) to produce (A,B) there is no need to save the input pair
(a,b). Hence we can store the results(A,B) in the same location as(a,b).
2. Data x(n) after decimation is stored in reverse order.
Q.24.The direct computation of DFT of a sequence X(n) requires 4N2 real
multiplications
and N(4N-2) real additions.
Q.25. The direct computation of DFT of a sequence X(n) requires N2 complex
multiplications and N(N-1) complex additions.
Q.26. What are the advantages of FFT algorithm?
Fast fourier transform reduces the computation time. In DFT computation, number of
multiplication is N2 and the number of addition is N(N-1). In FFT algorithm, number of
multiplication is only N/2(log2N) . Hence FFT reduces the number of elements (adder,
multiplier Z &delay elements). This is achieved by effectively utilizing the symmetric
and periodicity properties of Fourier transform.
Ans. The mapping of frequency from 1 to is approximately linear for small value
of For the higher frequencies, however the relation between Q x o becomes
highly non-linear. This introduces the distortion in the frequency scale of digital filter
relative to analog filter. This effect is known as wraping effect.
The influence of the wraping effect on the amplitude response can be demonstrated by
considering on analog filter with no. of passband centered at regular derived digital filter
has some numbers of passbands but the centre frequencies and the bandwidth of higher
frequencies passband in digital domain tends to be reduced.
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1. Anti-aliasing filter must be used which is a low pass filter to remove high frequency
noise contain in input signal. It avoids aliasing effect also.
2.Sample and hold circuit is used to keep the voltage level constant.
3.Output signal of Digital to analog converter is analog, that is a continuous signal. But it
contain high frequency components. Such high frequency components are
4.Amplifiers are used sometimes to bring the voltage level of input signal upto required
level for distortionless transmission.
Q. 3. Show for 3rd order butterworth low pass filter the Location of its poles and
zeroes in a s—plane.
Ans.
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Q.1. What is the basic difference between cascade form and direct form structures
for FIR systems?
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Ans. Basis occurs in the usage of memory space in bats coses. Cascade form is basically
in need of series memory. No of memory space required less in case of direct-2 form of
FIR w.r.t. cascade form start use of FIR systems.
Ans. The direct form II realization requires only the layer of M or N storage elements.
When compared to direct form I realisation the direct form II uses minimum number of
storage elements and hence said to be a Canonic striictur JJrweves wJ’ the Jc is
performed sequentially, the direct form II needs two adders instead of one adder required
for the direct form I.
Though the direct form I and II are commonly employed, they have two drawbacks viz (i)
they lock hardware flexibility and (ii) due to finite precision arithmetic, the sensitivity of
the co-efficients to quantisation effects increases with the order of the filter. This
sensitivity may change the co-efficient values and hence the frequency response, thereby
causing the filter to become unstable. To overcome these effects, the cascade and parallel
realizations can be implemented.
Q. 3. Compare different form structures of filter realization from the point of view
of speed and memory requirement.
Ans. The structural representation provides the relations between some pertinent internal
variable with the input and output that in turn provide the keys to implementations. There
are various form of structural representations of a digital filter.
In case of direct I form structure realization separate delay for both input and output
signal samples. So more memory is utilized by this form. For example.
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In case of direct-Il form structure realization only one delay is required for both input and
output signal samples. Therefore it is more efficient in term of memory requirements. For
example:
Ans. The following are some of the finite word length effects in digital filters:
5 Limit cycles
Ans. Rounding is the process of reducing size of a binary number to finite size of ‘b’ bits
such that the rounded b-bit number is closest to the original unquantized number.
For Example:
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Error due to rounding: The quantization error is fixed point number due to rounding is
defined as
The range of error due to rounding for all the three formats (i.e. one’s complement, two’s
complement and sign-magnitude) of fixed point presentation is same.
In fixed point representation the range of error made by rounding a number to ‘b’ bits is
Ans. In fixed point representation the bits allowed for integer part and fractional part and
so the position of binary point is fixed. The main drawback of this representation is that,
due to the fixed integer and fraction part, too large and to small values cannot be
represented. The bit to the right represent the fractional part of the number and those to
the left represent the integer part.
For Example:
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The negative numbers are represented in three different form for fixed point arithmetic
1. Sign-magnitude form.
2. One’s-complement form
3. Two’s-complement form.
1. Sign Magnitude form: In this form, the MSB is used to represent the given no. is
positive or negative. Let ‘N’ be the length of binary bits, then (N-i) bit will represent
magnitude and MS represent sign.
For example:
2. One’s complement form: In this form the positive number is represented as in the sign
magnitude notation. But the negative number is obtained by complementing all thebits of
the positive number.
For example:
3. Two’s complement form: In this form positive numbers are represented as in sign
magnitude and one’s complement. The negative number is obtained by complementing
all the bits of the +ve number and adding one to the least significant bit.
For example:
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Where M is called mantissa and it will be in binary fraction format. The value of M will
be in the range of and E is called exponent and it is either a positive or
negative integer.
In this form, both mantissa and exponent uses one bit for representing sign. Usu
- ally the LSB is mantissa and exponent is used to represent the sign. A ‘I’ in the LSB
represent negative sign and a ‘0’ in the LSB represent positive sign.
The floating point representation is explained by considering a five bit mantissa and three
bit exponent with a total size of eight bits. In mantissa the LSB is used to represent the
sign and other four bits are used to represent a binary fraction number. In exponent the
LSB is used to represent the sign and the other two bits are used to represent a binary
integer number.
Q 8 What is windowing?
Ans. One possible way of finding on FIR filter which approximates H(eJw) will be to
truncate the infinite fourier series at Abrupt truncation of the series would
lead to oscillations m the pass-band and stop-band These oscillations may be reduced by
use of less abrupt truncation of the fourier series. This can be achieved by multiplying the
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infinite impulse response with a finite weighing sequence w(n) called as window. This
process is called as windowing.
Ans 1. The infinite duration impulse response can be converted to a finite duration
impulse response by trucating the infinite series at But this results in undesirable
oscillations in the pass-band and step-band of the digital filter. This is due to slow
convergence of the Fourier series near the point of discontinuity. These undesirable
oscillations can be reduced by using a set of time limited weighing functions z e referred
as windowing function.
2 The windowmg function consists of main lobe which contains most of the energy of
window function and side lobes which decay rapidly
4 Window function have side lobes that decrease in energy rapidly as tends to
Ans.
Q.11. In what cases FIR filters will be preferred over IIR filters?
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Q. 12. What will happen if length of windows is increased in design of FIR filters?
need to be calculated. A more memory space used for it. More lengths of window means
more accuracy in transation process.
Ans. Let us consider the example of a lowpass filter having desired frequency
response as depicted of in figure (a). This response has the cut off frequency at
wc.
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Note : In figure oscillations or ringing takes place near band-edge of the filter. These
oscillations or ringing is generated because of sidebes m the frequency response of
the window function This oscillatory behaviour (i e ringing effect) near the band edge of
the filter is known as Gibbs phenonzenon Thus, the ringing effect takes place because of
sidelobes in W(w). These sidelobes are generated because of abrupt discontinuity (in case
of rectangular window) of the window function. In case of rectangular window, the
sidelobes are larger m size because the discontinuity is abrupt Therefore, ringing effect is
maximum in rectangular window
Therefore, different window functions are developed which consists of taper and decays
gradually toward zero This reduces sidelobes and hence ringing effect in H(w)
Q. 14. What are the essential features of a good window for FIR filters?
5. The trade off between main lobe widths and side lobe level can be adjusted.
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6. Smoother ends.
Ans. For FIR filter unit impulse response for symmetric system are given by:
For FIR filters m is finite i.e. may be odd, symmetric and antisymmetric conditions so in
hR filters m is infinite.
So it does not satisfies linear phase condition of eq. (1) and (2). Due to ‘rn’ tends to ‘00’
hR filters cannot have linear phase.
Ans. FIR is a finite impulse response. FIR system has an impulse response that is zero
outside of same finite time interval. FIR system has a finite memory of length M samples.
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1’s complement.
Ans. fraction (7/8) = (0.111) in sign magnitude , 1’s complement and 2’s complement
Fraction (-7/8) = (1.111) in sign magnitude
= (1.000) in 1’s complement
= (1.001) in 2’s complement
Q.28. The filter coefficient H = -0.673 is represented by sign magnitude fixed point
arithmetic. I the word length is 6 bits , compute the quantization error due to
truncation.
Ans. (0.673) = (0.1010110…)
(-0.673) = (1.1010110…)
after truncating to 6 bits we get
(1.101011) = -0.671875
Quantization error = xq – x
= (-0.671875)-(-0.673)
= 0.001125
Q.29. Give the expression for the signal to quantization noise ratio and calculate
the improvement with an increase of 2 bits to the existing bit.
Ans. SNR = 6b – 1.24dB,With an increase of 2 bits, increase in SNR is approximately
12dB.
Q.30. Why rounding is preferred over truncation in realizing digital filters?
Ans. 1. The quantization error due to rounding is independent of the type of arithmetic.
2. The mean of rounding error is Zero. 3. The variance of rounding error signal is low.
Q.31.What is product quantization error? Or What is round-off noise error?
Ans. Product quantization error arises at the output of a multiplier. Multiplication of a
‘b’ bit data with a ‘b’ bit coefficient results in a product having 2b bits. Since a ‘b’ bit
register is used , the multiplier output must be rounded or truncated to ‘b’ bits which
produces an error. This error is known as product quantization error.
UNIT V APPLICATIONS
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recombination of the set of sub band signals, one can approximately generate the
original speech signal.
17. What is meant by Image Enhancement?
Image enhancement is to improve the appearance of images for human perception
by making some features of the image like edges or contrast, more prominent
relative to others. It is done for the purpose of image analysis or for display
18. What is the use of adaptive filters?
Adaptive filters are capable of adjusting their co-efficient continuously during
transmission of data and this is done by operating on the received signal in
accordance with some algorithm. These filters are used for adaptive equalization of
channel output and adaptive prediction in adaptive Differential pulse code
modulation.
19. Give the advantages of digital recording.
1. A high signal to noise ratio limited by A to D conversion accuracy
2. Absence of wow- flutter speed variation
3. Elimination of harmonic distortion at upper signal extremity
4. Removal of amplitude variations caused by changes in tape magnetization
5. Avoiding inter channel cross talk
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