UNIT III FINITE IMPULSE RESPONSE DIGITAL FILTERS

PART A

1.What are the different types of filters based on impulse response?

Based on impulse response the filters are of two types
1. IIR filter
2. FIR filter
The IIR filters are of recursive type, whereby the present output sample depends on the present
input, past input samples and output samples.
The FIR filters are of non recursive type, whereby the present output sample depends on the
present input sample and previous input samples.

2. What are the different types of filters based on frequency response?

Based on frequency response the filters can be classified as
1. Lowpass filter
2. Highpass filter
3. Bandpass filter
4. Bandreject filter

3.Distinguish between FIR filters and IIR filters.

FIR filter
These filters can be easily designed to have perfectly linear phase.
FIR filters can be realized recursively and non-recursively.
Greater flexibility to control the shape of their magnitude response.
Errors due to round off noise are less severe in FIR filters, mainly because feedback is not used.
IIR filter
These filters do not have linear phase.
IIR filters are easily realized recursively.
Less flexibility, usually limited to specific kind of filters.
The round off noise in IIR filters is more.

4. What are the design techniques of designing FIR filters?

There are three well known methods for designing FIR filters with linear phase .They are
(1.)Window method (2.)Frequency sampling method (3.)Optimal or minimax design.

5.What is Gibbs phenomenon?

One possible way of finding an FIR filter that approximates H(ejw) would be to truncate the
infinite Fourier series at n=(N-1/2).Direct truncation of the series will lead to fixed percentage
overshoots and undershoots before and after an approximated discontinuity in the frequency
response.
6. List the steps involved in the design of FIR filters using windows.
1.For the desired frequency response Hd(w), find the impulse response hd(n) using Equation
hd(n)=1/2 Hd(w)ejwndw
-
2.Multiply the infinite impulse response with a chosen window sequence w(n) of length N to
obtain filter coefficients h(n),i.e.,
h(n)= hd(n)w(n) for |n|(N-1)/2
= 0 otherwise

3.Find the transfer function of the realizable filter

(N-1)/2
H(z)=z-(N-1)/2 [h(0)+ h(n)(zn+z-n)]
n=0
7. What are the desirable characteristics of the window function?
The desirable characteristics of the window are
1.The central lobe of the frequency response of the window should contain most of the energy
and should be narrow.
2.The highest side lobe level of the frequency response should be small.
3.The side lobes of the frequency response should decrease in energy rapidly as tends to .

8. What is the necessary and sufficient condition for linear phase characteristic in FIR
filter?
The necessary and sufficient condition for linear phase characteristic in FIR filter is, the impulse
response h(n) of the system should have the symmetry property i.e.,
H(n) = h(N-1-n)
where N is the duration of the sequence.

9.What are the advantages of Kaiser window?

o It provides flexibility for the designer to select the side lobe level and N
o It has the attractive property that the side lobe level can be varied continuously from the low
value in the Blackman window to the high value in the rectangular window

10.What is the principle of designing FIR filter using frequency sampling method?
In frequency sampling method the desired magnitude response is sampled and a linear phase
response is specified .The samples of desired frequency response are identified as DFT
coefficients. The filter coefficients are then determined as the IDFT of this set of samples.

11. For what type of filters frequency sampling method is suitable?

Frequency sampling method is attractive for narrow band frequency selective filters where only a
few of the samples of the frequency response are non zero.

12.What are the advantages and disadvantages of FIR filters?

Advantages:
1. FIR filters have exact linear phase.
2. FIR filters are always stable.
3. FIR filters can be realized in both recursive and non recursive structure.
4. Filters with any arbitrary magnitude response can be tackled using FIR sequence.
Disadvantages:
1. For the same filter specifications the order of FIR filter design can be as high as 5 to 10 times
that in an IIR design.
2. Large storage requirement is requirement
3. Powerful computational facilities required for the implementation.

13.Draw the direct form realization of FIR system.

14.Draw the direct form realization of a linear Phase FIR system for N odd
15. Draw the direct form realization of a linear Phase FIR system for N even.

16. Draw the M stage lattice filter.

17. When cascade form realization is preferred in FIR filters?

The cascade form realization is preferred when complex zeros with absolute magnitude is less
than one.

18.State the equations used to convert the lattice filter coefficients to direct form FIR Filter
coefficient.
m(0) = 1
m(m) = km
m-1(m-k)m(k) = m-1(k) + m(m)

19. State the equations used to convert the FIR filter coefficients to the lattice filter
Coefficient.
For an M_stage filter , m-1(0) =1 and km = m(m)
m(m-k)m-1(k) = m(k) - m(m) , 1km-1
1-m2 (m)

19.What is filter?
Filter is a frequency selective device, which amplifies particular range of frequencies and
attenuate particular range of frequencies.

20. Comment on the passband and stopband characteristics of butterworth and chebyshev
filters
1. Butterworth filter have monotonically decreasing response. Chebyshev filters have ripples in
passband and monotonic response in stopband
2. For the same order, transition band chebyshev filter is narrow than that in butterworth filter.
3. Poles of butterworth filter lie on the circle, whereas poles of chebyshev filters lie on the ellipse
4. For the same specification, order of the chebyshev filter is low compared to butterworth filter.

21. Mention the disadvantage of bilinear transformation technique.

1. There is nonlinearity in the frequency relationship. This nonlinearity is large at high
frequencies
2. Impulse response and phase response of the analog filter is not preserved during bilinear
mapping.

22. What is frequency wrapping?

The frequency relationship in bilinear transformation is given as,

Mapping between _ & in bilinear transformation

In this figure observe that the relationship is highly nonlinear at higher values of . The entire
frequency range (- to ) is mapped into to . This nonlinearity is called warping effect.

23. Write any two properties of Chebyshev Filters.

1. Chebyshev filters are all pole filters.
2. Poles of the chebyshev filters lie on the ellipse.
3. The ripples in magnitude have strong effect on the locations of poles

24. Why do we go for analog approximation to design a digital filter?

These are effective filter approximation techniques available in analog domain. Using
transformation methods a stable analog filter can be converted to stable digital filters. Hence it
becomes easier to design IIR filters from analog. But such effective approximations are not
available in discrete domain

25. List the various forms of realization of IIR system.

1. Direct Form I
2. Direct Form II
3. Cascade Realization
4. Parallel Form Realization

26. Define bilinear transformation.

The bilinear transformation is a mapping that transforms the left half of s plane into
the unit circle in the z plane only once, thus avoiding aliasing of frequency
components

Mapping:
1. LHS of s-plane is mapped inside the unit circle
2. RHS of s-plane is mapped outside the unit circle
3. Imaginary axis is mapped over the unit circle.

27.What are the properties of the bilinear transformation?

The mapping for the bilinear transformation is a one to one mapping; that is for every point z
, there is exactly one corresponding points s and vice versa
The j-axis maps on to the unit circle |z| = 1, the left half of the s plane maps to the interior of
the unit circle |z| = 1 and right half of the s plane maps on to the exterior of the unit circle |z| = 1.

28. How many number of additions , multiplications and memory location are
required to realize a system H(z) having M zeros and N poles in (a) direct form I
realization (b) direct form II realization
The direct form I realization requires M+N+1 multiplication, M+N additions and M+N+1
memory locations.
The direct form II realization requires M+N+1 multiplication, M+N additions and maximum
of (M,N) memory locations.

29. Mention advantage of direct form II and cascade structures?

1. Direct form II structures requires less number of storage allocations
2. Cascade structures are easy to implement, since second order section are simply cascaded.

30. Mention two transformations to digitize an analog filter.

1. Bilinear Transformation
2. Impulse invariant transformation

31.Write down the expression for the transfer function of the first orderbutterworth analog
filter having Low pass behavior.
The transfer function of the normalized lowpass butterworth filter can be expressedHan(s) =
1/(s+1)

32. Difference between analog filter and digital filter

33.What are the steps required to design an IIR digital filters.

It requires three steps
1. Map the desired digital filter specification into those for an equivalent analog filter
2. Derive the analog transfer for the analog prototype
3. Transform the transfer function of the analog prototype into an equivalent digital filter transfer
function.

34.What are the advantages and disadvantages of digital filters?

Advantages
1. A digital filter is highly immune to noise and posses considerable parameter stability
2. Digital filter afford a wide variety of shapes for the amplitude and phase responses.
3. Digital filters can be operated over a wide range of frequencies.
4. The coefficients of digital filter can be programmed and altered any time to obtain the desired
characteristics
5. Multiple filtering is possible only in digital filter
6. There are no problems of input or output impedance matching with digital filter
Disadvantages:
The quantization error arises due to finite word length in the representation of
signals and parameters.

35.Write the expression for order of Butterworth filter?

36.Write the expression for the order of chebyshev filter?

37.Write the various frequency transformations in analog domain?

38.Write the steps in designing chebyshev filter?

1. Find the order of the filter.
2. Find the value of major and minor axis .
3. Calculate the poles.
4. Find the denominator function using the above poles.
5. The numerator polynomial value depends on the value of n.
6. If n is odd: put s=0 in the denominator polynomial.
7. If n is even put s=0 and divide it by

39.Write down the steps for designing a Butterworth filter?

1. From the given specifications find the order of the filter
2. Find the transfer function from the value of N
3. Find
4. Find the transfer function ha(s) for the above value of by su s by that
value.
40. State the equation for finding the poles in chebyshev filter

41. What is warping effect?

For smaller values of w there exist linear relationship between w and .but for larger values of w
the relationship is nonlinear. This introduces distortion in the frequency axis. This effect
compresses the magnitude and phase response. This effect is called warping effect

42. Write a note on pre warping.

The effect of the non linear compression at high frequencies can be compensated. When the
desired magnitude response is piecewise constant over frequency, this compression can be
compensated by introducing a suitable rescaling or pre warping the critical frequencies.

43.Why impulse invariant method is not preferred in the design of IIR filters other
than low pass filter?
In this method the mapping from s plane to z plane is many to one. Thus there is an infinite
number of poles that map to the same location in the z plane, producing an aliasing effect. It is
inappropriate in designing high pass filters. Therefore this method is not much preferred.

44. Give the bilinear transform equation between s plane and z plane
s=2/T (z-1/z+1)

45. By impulse invariant method obtain the digital filter transfer function and the
differential equation of the analog filter h(s) =1/s+1

46.What is meant by impulse invariant method?

In this method of digitizing an analog filter, the impulse response of the resulting digital filter is
a sampled version of the impulse response of the analog filter. For e.g. if the transfer function is
of the form, 1/s-p, then

47.What do you understand by backward difference?

One of the simplest methods of converting analog to digital filter is to approximate
the differential equation by an equivalent difference equation.

48.What are the properties of chebyshev filter?

1. The magnitude response of the chebyshev filter exhibits ripple either in the stopband or the
pass band.
2. The poles of this filter lies on the ellipse.

49. Give the Butterworth filter transfer function and its magnitude characteristics
for different orders of filter.
The transfer function of the Butterworth filter is given by

50. Give the magnitude function of Butterworth filter.

The magnitude function of Butterworth filter is

51. Give any properties of Butterworth Lowpass filters

The magnitude response of the Butterworth filter decreases monotonically as
the frequency _ increases from 0 to
The magnitude response of the Butterworth filter closely approximates the
ideal response as the order N increases. The poles of the Butterworth filter lies on a circle.

52. Give the equation for the order N, major, minor axis of an ellipse in case of
chebyshev filter?

53. Give the expression for poles and zeroes of a chebyshev type 2 filters.

54. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the following methods
1. Approximation of derivatives
2. Impulse invariant method
3. Bilinear transformation

55. List the Butterworth polynomial for various orders.

56. Differentiate Butterworth and Chebyshev filter.

57. What is the mapping procedure between s plane and z plane in the method of mapping
of differentials? What is its characteristics?
The mapping procedure between s plane to z-plane in the method of mapping of differentials is
given by
H(z)= H(s) | s = (1-z-1)/T
The above mapping has the following characteristics
The left half of the s plane maps inside a circle in the z plane
The left half of the s plane maps outside a circle in the z plane
The j_ axis maps on to the perimeter of the circle

58. Define backward difference equation.

One of the simplest methods for converting an analog filter into a digital filter is to approximate
the differential equation by an equivalent difference equation
d/dt y(t) | t =nT = y(nT)-y(nT-T)/T
= y(n)-y(n-1)/T
The above equation is known as backward difference equation.

59. Name the method to convert analog filter transfer function to digital transfer
function.
The method to convert analog filter transfer function to digital transfer function is
Bilinear Transformation
Impulse Invariance Method Approximation of Derivatives

60. Use the backward difference for the derivative to convert the analog low pass
filter with system function H(s) = 1/ s+8
Ans: H(z)= 1/ (9-z-1)