UNIT I

IIR FILTER
1. What is a digital filter?
A digital filter is a device that eliminates noise and extracts the signal of interest from other
signals.

2. Analog filters are composed of which parameters?

* pass band
* stop band
* Cut-off frequency

3. Define pass band.

It passes certain range of frequencies. In this, attenuation is zero.

4. Define stop band.

It suppresses certain range of frequencies. In this, attenuation is infinity.

5. What is mean by cut-off frequency?

This is the frequency which separates pass band and stop band.

6. What is the difference between analog and digital filters?

Analog filters are designed using analog components (R,L,C) while digital filters are
implemented using difference equation and implemented using software.

7. What are the basic types of analog filters?

* Low pass filter - LPF
* High pass filter – HPF
* Band pass filter - BPF
* Band stop filter – BSF

9. Define IIR filter.

The filters designed by considering all the infinite samples of impulse
response are called IIR filter.

10. What is the condition for digital filter to be realize?

The impulse response of filter should be causal, h(n) = 0 for n<0.

11. Why ideal frequency selective filters are not realizable?

Ideal frequency selective filters are not realizable because they are non- causal. That is, its
impulse response is present for negative values of ‘n’ also.

12. For IIR filter realization what is required?

Present, past, future samples of input and past values of output are required.

13. Why IIR systems are called recursive systems?

Because the feedback connection is present from output side to input
14. Which types of structures are used to realize IIR systems?
* Direct form structure
* Cascade form structure
* Parallel form structure

15. Why direct form-II structure is preferred most and why?

The numbers of delay elements are reduced in direct form-II structure compared to direct form-I
structure. That means the memory locations are reduced in direct form-II structure.

16. Why direct form-I and direct form-II are called as direct form structures?
The direct form-I and direct form-II structures are obtained directly from the corresponding
transfer function without any rearrangements. So these structures are called as direct form
structures.

17. What is advantage of direct form structure?

Implementation of direct form is very easy.

18. Give the disadvantage of direct form structure?

Both direct form structures are sensitive to the effects of quantization errors in the coefficients.
So practically not preferred

19. What is the use of transpose operation?

If two digital structures have the same transfer function then they are called as equivalent
structures. By using the transpose operation, we can obtain equivalent structure from a given
realization structure.

20. What is transposition or flow graph reversal theorem?

If we reverse the directions of all branch transmittances and interchange input and output in the
flow graph then the system transfer function remains unchanged. * Parallel form structure

21. Why direct form-II structure is preferred most and why?

The numbers of delay elements are reduced in direct form-II structure compared to direct form-I
structure. That means the memory locations are reduced in direct form-II structure.

22. Why direct form-I and direct form-II are called as direct form structures?
The direct form-I and direct form-II structures are obtained directly from the corresponding
transfer function without any rearrangements. So these structures are called as direct form
structures.

23. What is advantage of direct form structure?

Implementation of direct form is very easy.

24. Give the disadvantage of direct form structure?

Both direct form structures are sensitive to the effects of quantization errors in the coefficients.
So practically not preferred
25. What is the use of transpose operation?
If two digital structures have the same transfer function then they are called as equivalent
structures. By using the transpose operation, we can obtain equivalent structure from a given
realization structure.

26. What is transposition or flow graph reversal theorem?

If we reverse the directions of all branch transmittances and interchange input and output in the
flow graph then the system transfer function remains unchanged.

27. Howa transposed structure is obtained?

* Reverse all signal flow graph directions.
* Change branching nodes into adders and vice-versa.
* Interchange input and output.

28. Why feed back is required in IIR systems?

It is required to generate infinitely long impulse response in IIR systems.

29. Write the expression for order of Butterworth filter?

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

31. Write the various frequency transformations in analog domain?

32. 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.
If n is odd: put s=0 in the denominator polynomial. If n is even put s=0 and divide it by (1+e2) ½

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

34. State the equation for finding the poles in chebyshev filter.
35. State the steps to design digital IIR filter using bilinear method.

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

37. 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.

38. 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
H (z) =1/1-e-pTz-1

39. 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.
d/dt(y(t)/t=nT=(y(nT)-y(nT-T))/T

40. What are the properties of chebyshev filter?

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

41. Give the Butterworth filter transfer function and its magnitude characteristics for
Different orders of filter.

42. 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 (IIM)
3. Bilinear transformation (BLT)

43. Write down bilinear transformation.

s=2/T (z-1/z+1)

44. What is a disadvantage of BLT method?

The mapping is non-linear and because of this, frequency warping effect takes place.
45. Differentiate Butterworth and Chebyshev filter.
Butterworth damping factor 1.44 and chebyshev is 1.06
Butterworth is flat response .but chebyshev is damped response.

46. How analog poles are mapped to digital poles in impulse invariant transformation?
In impulse invariant transformation the mapping of analog to digital poles are as follows, the
analog poles on the left half of s-plane are mapped into the interior of unit circle in z-plane. The
analog poles on the imaginary axis of s-plane are mapped into the unit circle in the z-plane. The
analog poles on the right half of s-plane are mapped into the exterior of unit circle in z-plane.

47. What is the importance of poles in filter design?

The stability of a filter is related to the location of the poles. For a stable analog filter the poles
should lie on the left half of s-plane. For a stable digital filter the poles should lie inside the unit
circle in the z-plane.

48. Why an impulse invariant transformation is not considered to be one-to-one?

In impulse invariant transformation any strip of width 2π/T in the s-plane for values of s-plane in
the range (2k-1)/T ≤ Ω ≤ (2k-1) π/T is mapped into the entire z-plane. The left half of each strip
in s-plane is mapped into the interior of unit circle in z-plane, right half of each strip in s-plane is
mapped into the exterior of unit circle in z-plane and the imaginary axis of each strip in s-plane is
mapped on the unit circle in z-plane. Hence the impulse invariant transformation is many-to-one.

49. What is Bilinear transformation?

The bilinear transformation is conformal mapping that transforms the s-plane to z-plane. In this
mapping the imaginary axis of s-plane is mapped into the unit circle in z-plane, The left half of s-
plane is mapped into interior of unit circle in z-plane and the right half of s-plane is mapped into
exterior of unit circle in z-plane. The Bilinear mapping is a one-to-one mapping and it is
accomplished when

50. How the order of the filter affects the frequency response of Butterworth filter.
The magnitude response of butterworth filter is shown in figure, from which it can be observed
that the magnitude response approaches the ideal response as the order of the filter is increased.

51. Write the properties of Chebyshev type –1 filter.

The magnitude response is equiripple in the pass band and monotonic in the stop band.
The chebyshev type-1 filters are all pole designs.
The normalized magnitude function has a value of at the cutoff frequency Ω c.
The magnitude response approaches the ideal response as the value of N increases.

Part-B

1. Describe the impulse invariance and bilinear transformation methods used for designing
digital IIR filters. (11)
2. For the given specifications design an analog Butterworth filter,
0.9 ≤ H(jΩ) ≤ 1 for 0 ≤ Ω ≤ 0.2π
H(jΩ) ≤ 0.2 for 0.4π ≤ Ω π (11)
3. Design a digital Butterworth filter satisfying the constraints
0.707 ≤ H(ejω) ≤ 1 for 0 ≤ ω ≤ π/2
H(ejω) ≤ 0.2 for 3π ≤ ω ≤ π
With T = 1 sec using Bilinear transformation. (11)
4. Design a chebyshev filter for the following specification using impulse invariance
method. 0.8 ≤ H(ejω) ≤ 1 for 0 ≤ ω ≤ 0.2π
H(ejω) ≤ 0.2 for 0.6π ≤ ω ≤ π (11)
5. Determine a cascade and parallel realisation of the system characterised by the transfer
function which is expressed as under: H(z)=[2(z+2)]/[z(z-0.1)(z+0.5)(z+0.4)] (11)

6 .Obtain the direct form I and direct form II realisations for third order IIR transfer function
which is expressed as below: H(z)=(0.28z2+0.319z+0.04)/(0.5z3+0.3z2+0.17z-0.2) (11)
7.Realize the system given by difference equation
y(n)=0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.252x(n-2) (11)
8. Using bilinear transformation obtain H(Z) if H(s)=1/(s+1)2 (11)
9. Convert the analog filter into a digital filter whose system function is
H(S)=S+0.2/(S+a)2+9
Use impulse invariant technique. Assume T =1s (11)
10. An analog filter has a transfer function H(s) = (10 / s2+7s+10). Design a digital filter
equivalent to this impulse invariant method. (11)

UNIT II

1. What are FIR and IIR systems?

The impulse response of a system consist of infinite number of samples are called IIR
system & the impulse response of a system consist of finite number of samples are called FIR
system.

2. What are the properties of FIR filters?

 FIR filter is always stable.
 A realizable filter can always be obtained.
 FIR filter has a linear phase response.

3. Define linear phase shift filter

For a filter to have linear phase the phase response θ(w) α w is the angular frequency.
The linear phase filter does not alter the shape of the signal. The necessary and sufficient
condition for a filter to have linear phase is given by,
h(n) = ± h(N-1-n); 0 ≤ n ≤ N-1

4. Define sampling process.

Sampling is a process of converting Ct signal into Dt signal.
5. Mention the types of sampling.
Up sampling & Down sampling

6. Define truncation.
Truncating the sequence by multiplying with window function to get the finite value

7. How phase distortion and delay distortion are introduced?

The phase distortion is introduced when the phase characteristics of a filter is nonlinear with in
the desired frequency band. The delay distortion is introduced when the delay is not constant
with in the desired frequency band.

8. What is mean by FIR filler?

The filter designed by selecting finite number of samples of impulse response (h(n) obtained
from inverse fourier transform of desired frequency response H(w)) are called FIR filters

9. Write the steps involved in FlR filler design

Choose the desired frequency response Hd(w), Take the inverse Fourier transform and obtain
Hd(n) Convert the infinite duration sequence Hd(n) to h(n) Take Z transform of h(n) to get
H(Z)

10. What are advantages of FIR filter?

Linear phase FIR filter can be easily designed. By Efficient realization of FIR filter exists as both
recursive and non-recursive structures. FIR filter realized non-recursively stable. The round off
noise can be made small in non-recursive realization of FIR filter.

11. What are disadvantages of FIR FILTER?

The duration of impulse response should be large to realize sharp cutoff filters. The non-integral
delay can lead to problems in some signal processing applications.

12. What is the necessary and sufficient condition for the linear phase characteristic of FlR
filter?
The phase function should be a linear function of w, which in turn requires constant group
delay and phase delay.

13. List the well-known design technique for linear phase FIR filter design?
Fourier series method and window method
Frequency sampling method, Optimal filters design method.

14. What is the reason that FlR filter is always stable?

FIR filter is always stable because all its poles are at the origin.

15 What condition for the FlR sequence h(n) are to be imposed II order that this jilter
can be called a liner phase filter?
The conditions are
(i) Symmetric condition h(n)=h(N-1-n)
(ii) Antisymmetric condition h(n)=-h(N-1-n)
16. Under what conditions a finite duration sequence h(u) will yield constant group delay
its frequency response characteristics and not the phase delay?
If the impulse response is anti-symmetrical, satisfying the condition
H(n)=-h(N-1-n)
The frequency response of FIR filter will have constant group delay and not the phase delay

17. State the condition for a digital filter to be causal and stable?
A digital filler is causal if its impulse response h(n)=O for n<O.
A digital filter is stable if its impulse response is absolutely summable .i.e,
00

n=-oo

18. What are the properties of FIR filler?

I .FIR filter is always stable.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.

19. When cascade from realization is preferred in FIR filters?

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

20. What are the disadvantages of Fourier series method?

In designing FIR filter using Fourier series method the infinite duration impulse
response is truncated nt n= ± (N-1/2).0irect truncation of the series will lead to fixed
percentage overshoots and undershoots before and after an approximated discontinuity in
the frequency response .

21. Whal is Gibbs phenomenon?

One possible way of finding an FIR filter that approximates H(0) would be to truncate
the infinite Fourier series at n= ± (N-1 /2).Abrupt truncation of the series will lead to
oscillation both in pass band and is stop band .This phenomenon is known as Gibbs
phenomenon.

22. What are the desirable characteristics of the windows?

The desirable characteristics of the window are
I .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 sides lobes of the frequency response should decrease in energy rapidly as so tends to it.
24. What is the necessary and sufficient condition for linear phase characteristics in FIR
filter?
The necessary and sufficient condition for linear phase characteristics 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 die sequence.

25. What are the advantages of Kaiser Widow?

I. It provides flexibility for the designer to select the side lobe level and N.
2. 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 rectangle window.

26. 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 defined as OFT
coefficients. The filter coefficients are then determined as die IDFT of this set of samples.

27. 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.

28. What is meant by autocorrelation?

The autocorrelation of a sequence is the correlation of a sequence with its shifted
Version, and this indicates how fast the signal changes.

29. Define white noise?

A stationary random process is said to be white noise if its power density spectrum is
constant. Hence the white noise has flat frequency response spectrum.

30. What do you understand by a fixed-point number?

In fixed point arithmetic the position of the binary point is fixed. The bit to the right left
represent the integer part. For example, the binary number OJ.I JOO has the value 1.75 in
decimal

31. What are Gibbs oscillations?

One possible way of finding an FIR filter that approximates H(0) would be to truncate
the infinite Fourier series at n= ± (N-1 /2).Abrupt truncation of the series will lead to
oscillation both in pass band and is stop band .This phenomenon is known as Gibbs
phenomenon.
32. What are the design techniques for designing FIR filters?
There are 3 methods,
1. Window method
2. Frequency sampling method
3. Optimal design method.

33. What is the principle of designing of 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

34. Write the procedure for designing FIR filter using frequency-sampling method.
 Choose the desired (ideal) frequency response Hd(w).
 Take N-samples of Hd(w) to generate the sequence
 Take inverse DFT of to get the impulse response h(n).
 The transfer function H (z) of the filter is obtained by taking z-transform of impulse
response.

35. What are the drawback in FIR filter design using windows and frequency sampling
method? How it is overcome?
The FIR filter design using windows and frequency sampling method does not have
Precise control over the critical frequencies such as wp and ws. This drawback can be overcome
by designing FIR filter using Chebyshev approximation technique. In this technique an error
function is used to approximate the ideal frequency response, in order to satisfy the desired
specifications.

36. Write the characteristic features of rectangular window.

 The main lobe width is equal to 4π/N.
 The maximum side lobe magnitude is –13dB.
 The side lobe magnitude does not decrease significantly with increasing w.

37. List the features of FIR filter designed using rectangular window.
The width of the transition region is related to the width of the main lobe of window
spectrum. Gibb’s oscillations are noticed in the pass band and stop band. The attenuation in the
stop band is constant and cannot be varied.

38. Why Gibb’s oscillations are developed in rectangular window and how it can be
eliminated or reduced?
The Gibb’s oscillations in rectangular window are due to the sharp transitions from 1 to 0
at the edges of window sequence. These oscillations can be eliminated or reduced by replacing
the sharp transition by gradual transition. This is the motivation for development of triangular
and cosine windows.
39. List the characteristics of FIR filters designed using windows.
The width of the transition band depends on the type of window. The width of the
transition band can be made narrow by increasing the value of N where N is the length of the
window sequence. The attenuation in the stop band is fixed for a given window, except in case of
Kaiser Window where it is variable.

40. What are the conditions to be satisfied for constant phase delay in linear phase FIR
filters?
The conditions for constant phase delay are Phase delay, α = (N-1)/2 (i.e., phase delay is
constant) Impulse response, h (n) = -h (N-1-n) (i.e., impulse response is antisymmetric)

41. How constant group delay & phase delay is achieved in linear phase FIR filters?
The following conditions have to be satisfied to achieve constant group delay & phase delay.
Phase delay, α = (N-1)/2 (i.e., phase delay is constant) Group delay, β = π/2 (i.e., group delay is
constant) Impulse response, h(n) = -h(N-1-n) (i.e., impulse response is antisymmetric)

42. What are the possible types of impulse response for linear phase FIR filters?
There are four types of impulse response for linear phase FIR filters
 Symmetric impulse response when N is odd.
 Symmetric impulse response when N is even.
 Antisymmetric impulse response when N is odd.
 Antisymmetric impulse response when N is even.

43. Draw the direct form realization of FIR system.

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

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

47. State the equations used to convert the FIR filter coefficients to the lattice filter
Coefficient.

48. What is transposition theorem & transposed structure?

The transpose of a structure is defined by the following operations.
• Reverse the directions of all branches in the signal flow graph
• Interchange the input and outputs.
• Reverse the roles of all nodes in the flow graph.
• Summing points become branching points.
• Branching points become summing points.
According to transposition theorem if we reverse the directions of all branch transmittance and
interchange the input and output in the flow graph, the system function remains unchanged.
49. Draw the M stage lattice filter.

50. What is meant by fixed point number?

In fixed point number the position of a binary point is fixed. The bit to the right represent the
fractional part and those to the left is integer part.

51. What are the different types of fixed point arithmetic?

Depending on the negative numbers are represented there are three forms of fixed point
arithmetic. They are sign magnitude, 1’s complement, 2’s complement

PART-B

1. (a) Obtain the cascade and parallel realization of the system described by
y(n) = -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2) (7)
(b) Discuss about any three window functions used in the design of FIR filters. (4)
2. Determine the direct form II and parallel form realization for the following system.
y(n) = -0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.252x(n-2) (11)
3. 4. (a) Write the expressions for the Hamming, Hanning, Bartlett and Kaiser windows.
(4)
(b) Explain the design of FIR filters using windows. (7)
5. Design an ideal high pass filter with
Hd(ejω) = 1 for π/4 ≤ ω ≤ π
= 0 for ω ≤ π/4
Using Hanning window for N=11. (11)
6. Design an ideal high pass filter with
Hd(ejω) = 1 for π/4 ≤ ω ≤ π
= 0 for ω ≤ π/4
Using Hamming window for N=11. (11)
7. Using a rectangular window technique design a lowpass filter with pass band gain of
unity, cutoff frequency of 1000 Hz and working at a sampling frequency of 5kHZ. The
length of the impulse response should be 7. (11)
8. Design an ideal Hilbert transformer having frequency response
H(ejω) = j for -π ≤ ω ≤ 0
= -j for 0 ≤ ω ≤ π
Using blackman window for N=11.Plot the frequency response. (11)
9. Design an ideal Hilbert transformer having frequency response
H(ejω) = j for -π ≤ ω ≤ 0
= -j for 0 ≤ ω ≤ π
Using blackman window for N=11.Plot the frequency response. (11)

Unit – III
1. What are all the blocks are used to represent the CT signals by its samples?
* Sampler
* Quantizer

2. Define sampling process.

Sampling is a process of converting Ct signal into Dt signal.

3. Mention the types of sampling.

* Up sampling
* Down sampling

4. What is meant by quantizer?

It is a process of converting discrete time continuous amplitude into discrete time discrete
amplitude.

5. Define system function?

The ratio between z transform of output signal y(z) to z transform of input signal x(z) is called
system function of the particular system.

6. List out the types of quantization process.

Truncation & Rounding

7. Define truncation.
Truncating the sequence by multiplying with window function to get the finite value

8. What is meant by limit cycle oscillations?

In fixed point addition, overflow occurs due to excess of results bit, which are stored at the
registers. Due to this overflow, oscillation will occur in the system. Thus oscillation is called as
an overflow limit cycle oscillation.
23. How would you relate the steady-state noise power due to quantization and the b bits
representing the binary sequence?

Steady state noise power

Where b is the number of bits excluding sign bit.

24. What are the two kinds of limit cycle behavior in DSP?

1. Zero input limit cycle oscillations

2. Overflow limit cycle oscillations

25.What is meant by autocorrelation?

The autocorrelation of a sequence is the correlation of a sequence with its shifted version, and
this indicates how fast the signal changes.

26. What do finite word length effects mean?

The effects due to finite precision representation of numbers in a digital system are called finite
word length effects.

27. List some of the finite word length effects in digital filters.

 Errors due to quantization of input data.

 Errors due to quantization of filter co-efficient
 Errors due to rounding the product in multiplications
 Limit cycles due to product quantization and overflow in addition.

28. What are the different formats of fixed-point representation?

a. Sign magnitude format

b. One’s Complement format
c. Two’s Complement format.
In all the three formats, the positive number is same but they differ only in representing
negative numbers.
29. Explain the floating-point representation of binary number.

The floating-point number will have a mantissa part. In a given word size the bits allotted for
mantissa and exponent are fixed. The mantissa is used to represent a binary fraction number and
the exponent is a positive or negative binary integer. The value of the exponent can be
adjusted to move the position of binary point in mantissa. Hence this representation is called
floating point.

30. What are the types of arithmetic used in digital computers?

The floating point arithmetic and two’s complement arithmetic are the two types of
arithmetic employed in digital systems.

31. What are the two types of quantization employed in digital system?
The two types of quantization in digital system are Truncation and Rounding.

32. What is truncation?

The truncation is the process of reducing the size of binary number by discarding all bits less
significant than the least significant bit that is retained. In truncation of a binary number of
b bits all the less significant bits beyond bth bit are discarded.

33. What is rounding?

Rounding is the process of reducing the size of a binary number to finite word sizes of b-bits
such that, the rounded b-bit number is closest to the original unquantized number.

34. Explain the process of upward rounding?

In upward rounding of a number of b-bits, first the number is truncated to b-bits by
retaining the most significant b-bits. If the bit next to the least significant bit that is
retained is zero, then zero is added to the least significant bit of the truncated number. If the
bit next to the least significant bit that is retained is one then one is added to the least
significant bit of the truncated number.

35. How the digital filter is affected by quantization of filter coefficients?

The quantization of the filter coefficients will modify the value of poles & zeros and so
the location of poles and zeros will be shifted from the desired location. This will create
deviations in the frequency response of the system. Hence the resultant filter will have a
frequency response different from that of the filter with unquantized coefficients.

36. How the sensitivity of frequency response to quantization of filter coefficients is

minimized?
The sensitivity of the filter frequency response to quantization of the filter coefficients is
minimized by realizing the filter having a large number of poles and zeros as an
interconnection of second order sections. Hence the filter can be realized in cascade or
parallel form, in which the basic buildings blocks are first order and second order sections.
37. What is meant by product quantization error?
In digital computations, the output of multipliers i.e., the product are quantized to finite
word length in order to store them in registers and to be used in subsequent calculations.
The error due to the quantization of the output of multiplier is referred to as product
quantization error.

38. Why rounding is preferred for quantizing the product?

In digital system rounding due to the following desirable characteristic of rounding
performs the product quantization
1. The rounding error is independent of the type of arithmetic
2. The mean value of rounding error signal is zero.
3. The variance of the rounding error signal is least.

39. Define noise transfer function (NTF)?

The Noise Transfer Function is defined as the transfer function from the noise source to
the filter output. The NTF depends on the structure of the digital networks.

40. What are limit cycles?

In recursive systems when the input is zero or some nonzero constant value, the
nonlinearities die to finite precision arithmetic operations may cause periodic oscillations
in the output. These oscillations are called limit cycles.

41. What are the two types of limit cycles?

The two types of limit cycles are zero input limit cycles and overflow limit cycles.

42. What is zero input limit cycles?

In recursive system, the product quantization may create periodic oscillations in the
output. These oscillations are called limit cycles. If the system output enters a limit
cycles, it will continue to remain in limit cycles even when the input is made zero. Hence
these limit cycles are also called zero input limit cycles.

43. How overflow limit cycles can be eliminated?

The overflow limit cycles can be eliminated either by using saturation arithmetic or by
scaling the input signal to the adder.

44. What is the drawback in saturation arithmetic?

The saturation arithmetic introduces nonlinearity in the adder which creates signal
distortion.

45. What are the errors generated by A/D process?

The A/D process generates two types of errors. They are quantization error and
saturation error. The quantization error is due to representation of the sampled signal by
a fixed number of digital levels. The saturation errors occur when the analog signal
exceeds the dynamic range of A/D converter.
 46. What is quantization step size?
In digital systems, the numbers are represented in binary. With b-bit binary we
can generate 2b different binary codes. Any range of analog value to be represented
in binary should be divided into 2b levels with equal increment. The 2b levels are called
quantization levels and the increment in each level is called quantization step size. If R is
the range of analog signal then, Quantization step size, q = R/2b

47. Why errors are created in A/D process?

In A/D process the analog signals are sampled and converted to binary. The
sampled analog signal will have infinite precision. In binary representation of b- bits we
have different values with finite precision. The binary values are called quantization
levels. Hence the samples of analog are quantized in order to fit into any one of the
quantized levels. This quantization process introduces errors in the signal.

48. What is saturation arithmetic?

In saturation arithmetic when the result of an arithmetic operation exceeds the
dynamic range of number system, then the result is set to maximum or minimum possible
value. If the upper limit is exceeded then the result is set to maximum possible value. If
the lower limit is exceeded then the r4esult is set to minimum possible value.

49. What is overflow limit cycle?

In fixed point addition the overflow occurs when the sum exceeds the finite word
length of the register used to store the sum. The overflow in addition may lead to
oscillations in the output which is called overflow limit cycles.

50. How overflow limit cycles can be eliminated?

The overflow limit cycles can be eliminated either by using saturation arithmetic
or by scaling the input signal to the adder.

51. What is the drawback in saturation arithmetic?

The saturation arithmetic introduces nonlinearity in the adder which creates signal
distortion.

PART-B

1.Explain zero input limit cycle oscillations (11)

2. Describe the errors introduced by quantization. Explain the impact of quantization
of filter coefficients on the location of poles. (11)
3. Write a brief note on:
i) Input quantization (5)
ii) Limit cycles (6)
4. Discuss in detail the various quantization effects in the design of digital filters. (11)
5. Find the effect of co-efficient quantization on pole locations of the given second order
IIR system, when it is realized in direct form I and in cascade form. Assume a word
length of 4 bits through truncation. (11)
H(z) = 1 / (1 – 0.9 z + 0.2 z )
-1 -1

6. Express the decimal values -6/8 and 9/8 in

(i) Sign magnitude form (3)
(ii) One’s complement form (4)
(iii) Two’s complement form (4)
7. Derive the quantization input nose power and determine the signal to noise ratio of
the system (11)
8. Derive the truncation error and round off error noise power and compare both errors
(11)
9. Explain product quantization error and coefficient quantization error with examples
(11)
10. Derive the scaling factor So that prevents the overflow limit cycle oscillations in a
second order IIR system (11)

UNIT-IV

1. What is the energy density spectrum?

The quantity |Xa(F)|2 represent the distribution of signal energy as a function of
frequency and hence it is called the energy density spectrum of the signal that is
Sxx(F)=|X(F)|2
∞
Sxx(f)= |∑ x(n)e-j2ПKf |2
n= -∞

2. What is power density spectrum?

Let x(t) be a stationary random process. The statistical autocorrelation function
for this signal is
γxx(τ)=E[x*(t)x(t+τ)]
The Fourier transform of the autocorrelation function of a stationary random process
gives their power density spectrum
γxx(f)=F(γxx(τ))
∞
=∫ γxx(τ) e-j2Пf τ dτ
-∞
3. Using indirect method, how to find the energy density spectrum?
It requires two steps:
First, the autocorrelation rxx(k) is computed from x(n)
The Fourier transform of the autocorrelation is computed.
∞
r xx(k)= ∑x*(n)x(n+k)
n= -∞
∞
Sxx(f)= ∑ r xx(k)e-j2ПKf
n= -∞

4. What is three types of non parametric methods of power spectrum estimation?

i). Bartlett methods (Averaging periodogram)
ii) Welch method (modified Averaging periodogram)
iii) Blackman and tukey methods (smoothing periodogram)

5. What are nonparametric methods?

These methods make no assumption about how the data were generated and
hence are called nonparametric methods.

6. Define Periodogram.
Schuster defined the periodogram as a method to discover the frequencies of the
hidden harmonics signal. The estimate Pxx (f) can also be expressed as
N-1
Pxx(f)=1/N | ∑ x(n) e-j2Пfn |2 =1/N|X(f)|2
n= 0
Where X(f) is the Fourier transform of the sample sequence x(n). This well-
known form of the power density spectrum estimate is called the Periodogram.

7. What is a Bartlett method?

In this method to reduce the variance of the Periodogram,
Three steps
i) First divide the N point sequence x (n) into k non overlap subsequence of length M
ii) Find the periodogram for each sub sequence.
iii) Calculate the average periodogram of k subsequence.

8. What is the advantage of Bartlett window methods?

The effect of reducing the length of data from N point to M=N/K, result in a window
whose spectral width has been increased by a factor of K. consequently, the frequency
resolution has been reduced by a factor K in return in resolution we reduced the
variance.

9. What is the difference between Bartlett and welch’s methods?

i) First difference is that welch’s method allows overlapping of data sequence. The
overlap 50% or 75%.
ii) The second difference is the data with in a sequence are windowed prior to
computing the periodogram.

10. What is a Blackman and turkey method?

Blackman and turkey proposed a method in which the sampled autocorrelation
sequence is windowed first and then Fourier transformed.

11. Define quality of nonparametric methods?

The ratio of it’s the square of the mean of power spectrum estimate to variance.
Qa=E[Pxx(f)]2/var[Pxx(f)]

12. Define variability.

The reciprocal of this quantity called the variability.

13. What is the quality of three power spectrum estimate?

Estimate Quality factor
Bartlett 1.11N∆f
Welch 1.39N∆f
Blackman- 2.34 N∆f
tukey

14. Determine the frequency resolution of Bartlett, Welch, Blackman turkey methods
of power spectrum estimate. For quality factor 10.assume that overlaps in Welch
method is 50% and length of the sample sequence is 1000.
Quality factor=10
Length of sample sequence (N) =1000
Overlap in Welch method=50%
Bartlett method
QB=1.11N∆f
∆f= (Qbart/1.11N)=0.009
Welch method
Qw=1.39N∆f
∆f=0.0072
Blackman-turkey
QBT=2.34 N ∆f
∆f=0.0042

15. What is the advantage of Welch methods?

i) The Welch periodogram is reasonably computationally efficient due to the use of FFT
algorithms.
ii) The Welch method the variance of the random process is reduced compared to basic
periodogram and Bartlett methods.
iii) Welch method allows all the windowing techniques.

16. Limitation of non parametric methods for power spectrum estimation? (What are
the disadvantages of non-parametric methods of power spectral estimation?)
i) It requires long data sequence to obtain the necessary frequency resolution.
ii) Spectral leakage effect because of windowing.
iii)The assumption of the autocorrelation estimator x(m) to be zero for m≥N. This
assumption limits the frequency resolution and quality of power spectrum estimate.
iv) Assumption that the data are periodic with period N. These assumptions may not be
realistic.

17. Define mean.

The procedure of determining the average weight of a group of objects by summing
their individual weights and dividing by the total number of objects gives the average
value of x.
Mathematically the discrete sample mean can be described
_ n
X =1/n ∑xi
i=1
For the continuous case that mean value of the random variable,X is defined as
∞
X =E[X] =∫xfX(x) dx
-∞
where E[X] is read ``the expected value of X''. Other names for the same mean value x
or the expected value E[X] are average value and statistical average.

18. What are the two properties of power spectrum estimator?

i) Bias
ii) Variance
19. Define the bias of estimator?
Bias of estimator is defined as the true value of the parameter minus the expected
value of the estimator.

20. Define unbiased estimator.

An unbiased estimator is one for which the bias is 0. This then means that the expected
value of the estimator is the true value so that the probability density is symmetrical
then its center would be at the true value.

21. What is consistent estimator?

An estimator is said to be consistent if as the number of observation becomes lager, the
bias and the variance both tends to zero.
If the bias and variance both tend to zero as the limit tends to infinity or the number of
observations become large, the estimator is said to be consistent.

22. What is Variance?

The variance of an estimator effectively measures the width of the probability density
and is defined as
σmx=E[(mx)2]-[ E[(mx)]2
In using estimates the mean value estimate of mx, for a Gaussian random process is the
sample mean. A good estimator should have a small variance in addition to having a
small bias suggesting that the probability density function is concentrated about its
mean value. This says that as the number of observations N increase, the variance of the
sample mean decreases, and since the bias is zero, the sample mean is a consistent
estimator.

23. What are deterministic and random signals with an example?

Deterministic signals are functions that are completely specified in time. The nature
and amplitude of such signal at any time can be predicted. The Patten of the signal is a
regular and can be characteristics mathematically.
Examples
x (t)=ct this is the ramp whose amplitude of this signal increases linearly with time and
slope is c.
Random Signal (Non Deterministic Signal)
A non deterministic signal is one whose occurrence is random in nature and its Patten is
quite irregular.
24. Define the terms: i) auto correlation ii) cross correlation
Correlation gives a measure of similarity between two data sequences.
Auto correlation
For sequence x(n) the auto correction function r xx(k) is defined as
∞
r xx(k)= ∑x*(n)x(n+k) k=0,±1,±2,.....
n= -∞
or equivalently,
∞
r xx(k)= ∑x*(n)x(n-k) k=0,±1,±2,.....
n= -∞

Cross correlation
For two sequence x(n) and y(n),the cross correction function r xy(k) is defined as
∞
r xy(k)= ∑x*(n)y(n+k) k=0,±1,±2,.....
n= -∞
or equivalently,
∞
r xy(k)= ∑x*(n)y(n-k) k=0,±1,±2,.....
n= -∞
25. Define auto covariance.
Related to the autocorrelation function is the auto covariance function, which is
defined as
Cxx (t1, t2) =E [xt1-m (t1)] [xt2-m (t2)]
=γxx (t1, t2)-m(t1)m(t2)
Where m (t1) =E [xt1] and m (t2) =E [xt2] are the mean of x1 and x2 respectively. When
the process is stationary
Cxx(t1,t2)=Cxx(t1-t2)=Cxx(τ)= γxx (τ)-mx2
Where τ = t1-t2

26. What is zero padding? Does zero padding improve the frequency resolution in
Spectral estimate?
Zero padding is increase the length of sequence by adding zero to the given sequence.
Note that zero padding does not change the resolution but it does have the effect of
interpolating the spectrum pxx(f)

27. Define cross power spectral density.

The definition of the power density spectrum can be extended to two jointly
stationary random process x(t) and y(t),which have a cross correction function γxy(τ)
The fourier transform of the γxy(τ) is
∞
γxy(F) =∫ γxy(τ) e-j2ПF τ d τ
-∞
Which is called the cross power density spectrum.

28.Define white noise?

A stationary random process is said to be white noise if its power density
Spectrum is constant. Hence the white noise has flat frequency response spectrum.
SX(w) = σx, -π ≤ wπ

29. What do you understand by a fixed-point number?

In fixed point arithmetic the position of the binary point is fixed. The bit to the right represents
the fractional part of the number & those to the left represent the integer part. For example, the
binary number 01.1100 has the value 1.75 in decimal.

30. What is the objective of spectrum estimation?

The main objective of spectrum estimation is the determination of the power spectral density of a
random process. The estimated PSD provides information about the structure of the random
process which can be used for modeling, prediction or filtering of the deserved process.

31. What is meant by block floating point representation? What are its advantages?
In block point arithmetic the set of signals to be handled is divided into blocks. Each block has
the same value for the exponent. The arithmetic operations within the block uses fixed point
arithmetic & only one exponent per block is stored thus saving memory. This representation of
numbers is more suitable in certain FFT flow graph & in digital audio applications.

32. What are the advantages of floating point arithmetic?

1. Large dynamic range
2. Over flow in floating point representation is unlike.

33. What are the three-quantization errors to finite word length registers in digital filters?
1. Input quantization error
2. Coefficient quantization error
3. Product quantization error

34. How the multiplication & addition are carried out in floating point arithmetic?
In floating point arithmetic, multiplication are carried out as follows, Let f1 = M1*2c1 and f2 =
M2*2c2. Then f3 = f1*f2 = (M1*M2) 2(c1+c2) that is, mantissa is multiplied using fixed-point
arithmetic and the exponents are added. The sum of two floating-point numbers is carried out by
shifting the bits of the mantissa of the smaller number to the right until the exponents of the two
numbers are equal and then adding the mantissas.

35. What do you understand by input quantization error?

In digital signal processing, the continuous time input signals are converted into digital using a b-
bit ACD. The representation of continuous signal amplitude by a fixed digit produce an error,
which is known as input quantization error.

36. Draw the block diagram Analog domain signal processing?

37. What are the applications of Multi rate signal processing?

38. Define Decimation.

Decimation is a process in which sampling rate is reduced. It is also called as down sampling.

39. Define Aliasing.

The spectrum obtained after decimation overlaps the original spectrum. This overlaps causes
aliasing.

40. Name the two different blocks of interpolator.

1. upsampler
2.Anti-imaging filter

41. What is sample rate conversion?

It is the process of converting signal from one sampling rate to another, while changing the
information carried by the signal as little as possible.

42. Express the fraction 7/8 and -7/8 in sign magnitude, 2’s complement and 1’s
complement.
Fraction (7/8) = (0.111)2 in sign magnitude, 2’s complement and 1’s complement.
Fraction (-7/8)= (1.111)2 in sign magnitude
(1.000)2 1’s complement.
(1.001)2 2’s complement.
43. Give the expression for signal to quantization noise ratio and calculate the
improvement with an increase of 2 bit to the existing bit.
SNR=6.02b+10.79+10log10 σx 2
With an increase of 2 bits, increase in SNR is approximately 12 dB

44. Explain briefly the need for scaling in the digital filter implementation.
To prevent overflow, the signal level at certain points in the digital filter must
be scaled so that no overflow occurs in the adder.

45. What are the different types of signal representations?

a. Graphical representation
b. Functional representations
c. Tabular representation
d. Sequence representation.

46. Check the linearity and stability of g(n),

 since square root is nonlinear, the system is nonlinear.
 as long as x (n) is bounded; its square root is bounded. Hence this system is stable.

47. What is a decimation-in-frequency algorithm?

In this the output sequence X (K) is divided into two N/2 point sequences and each N/2 point
sequences are in turn divided into two N/4 point sequences.

48. What are the various applications of speech signals?

They are Speech and audio (speech recognition, speech synthesis, text to Speech, digital
audio, equalization)

49. Describe briefly the different methods of power spectral estimation?

 Bartlett method
 Welch method
 Blackman-Tukey method

50. What are nonparametric methods?

These methods make no assumption about how the data were generated and
hence are called nonparametric methods.

PART - B

1. Explain how DFT and FFT are useful in power spectral estimation. (11)

2. Explain Power spectrum estimation using the Bartlett window. (11)

3. Obtain the mean and variance of the averaging modified period gram estimate. (11)

4. How is the Blackman and Turkey method used in smoothing the Periodogram? (11)

5. Derive the mean and variance of the power spectral estimate of the Blackman and Turkey
method. (11)

6. What are the limitations of non-parametric methods in spectral estimation? (8)

7. How the parametric methods overcome the limitations of the non-

parametric methods? (11)

UNIT V - DIGITAL SIGNAL PROCESSOR

1. Write short notes on general purpose DSP processors

General-purpose digital signal processors are basically high speed microprocessors with hard
ware architecture and instruction set optimized for DSP operations. These processors make
extensive use of parallelism, Harvard architecture, pipelining and dedicated hardware whenever
possible to perform time consuming operations.

2. Write notes on special purpose DSP processors.

There are two types of special; purpose hardware.
(i) Hardware designed for efficient execution of specific DSP algorithms such as digital filter,
FFT.
(ii) Hardware designed for specific applications, for example telecommunication, digital audio.

3. Briefly explain about Harvard architecture.

The principal feature of Harvard architecture is that the program and the data memories lie in
two separate spaces, permitting full overlap of instruction fetch and execution.
Typically these types of instructions would involve their distinct type.
1. Instruction fetch
2. Instruction decode
3. Instruction execute

4. Briefly explain about multiplier accumulator.

The way to implement the correlation and convolution is array multiplication Method. For
getting down these operations we need the help of adders and multipliers. The combination of
these accumulator and multiplier is called as multiplier accumulator.

5. What are the types of MAC is available?

There are two types MAC’S available
1. Dedicated & integrated
2. Separate multiplier and integrated unit
6. What is meant by pipeline technique?
The pipeline technique is used to allow overall instruction executions to overlap. That is where
all four phases operate in parallel. By adapting this technique, execution speed is increased.

7. What are four phases available in pipeline technique?

The four phases are
(i) Fetch
(ii) Decode
(iii)Read
(iv) Execution

8. In a non-pipeline machine, the instruction fetch, decode and execute take 30 ns, 45 ns
and 25 ns respectively. Determine the increase in throughput if the instruction were
pipelined.
Assume a 5ns pipeline overhead in each stage and ignore other delays.
The average instruction time is = 30 ns+45 ns + 25 ns = 100 ns
Each instruction has been completed in three cycles = 45 ns * 3 = 135ns
Throughput of the machine =
The average instruction time/Number of M/C per instruction
= 100/135 = 0.7407
But in the case of pipeline machine, the clock speed is determined by the speed of the
slowest stage plus overheads.
In our case is = 45 ns + 5 ns =50 ns
The respective throughput is = 100/50 = 2.00
The amount of speed up the operation is = 135/50 = 2.7 times

9. Assume a memory access time of 150 ns, multiplication time of 100 ns, addition time of
100 ns and overhead of 10 ns at each pipe stage. Determine the throughput of MAC
After getting successive addition and multiplications
The total time delay is 150 + 100 + 100 + 5 = 355 ns
System throughput is = 1/355 ns.

10.Write down the name of the addressing modes.

Direct addressing.
Indirect addressing.
Bit-reversed addressing.
Immediate addressing.
i. Short immediate addressing.
ii. Long immediate addressing.
Circular addressing.

11. What are the instructions used for block transfer in C5X Processors?
The BLDD, BLDP and BLPD instructions use the BMAR to point at the source or destination
space of a block move. The MADD and MADS also use the BMAR to address an operand in
program memory for a multiply accumulator operation
12. Briefly explain about the dedicated register addressing modes.
The dedicated-registered addressing mode operates like the long immediate addressing modes,
except that the address comes from one of two special-purpose memory mapped registers in the
CPU: the block move address register (BMAR) and the dynamic bit manipulation register
(DBMR).
The advantage of this addressing mode is that the address of the block of memory to be acted
upon can be changed during execution of the program.

13. Briefly explain about bit-reversed addressing mode?

In the bit-reversed addressing mode, INDX specifies one-half the size of the FFT. The value
contained in the current AR must be equal to 2n-1, where n is an integer, and the FFT size is 2n.
An auxiliary register points to the physical location of a data value. When we add INDX t the
current AR using bit reversed addressing, addresses are generated in a bit reversed fashion.
Assume that the auxiliary registers are eight bits long, that AR2 represents the base address of
the data in memory (0110 00002), and that INDX contains the value 0000 10002.

14. Briefly explain about circular addressing mode.

Many algorithms such as convolution, correlation, and finite impulse response (FIR) filters can
use circular buffers in memory to implement a sliding window; which contains the most recent
data to be processed. The ‘C5x supports two concurrent circular buffer operating via the ARs.
The following five memory-mapped registers control the circular buffer operation.
1. CBSR1- Circular buffer 1 start register.
2. CBSR2- Circular buffer 2 start Register,
3. CBER1- Circular buffer 1 end register
4. CBER2- Circular buffer 2 end register
5. CBCR - Circular buffer control register.

15. Write the name of various part of C5X hardware.

1. Central arithmetic logic unit (CALU)
2. Parallel logic unit (PLU)
3. Auxiliary register arithmetic unit (ARAU)
4. Memory-mapped registers.
5. Program controller.

16. Write short notes about arithmetic logic unit and accumulator.
The 32-bit general-purpose ALU and ACC implement a wide range of arithmetic and logical
functions, the majority of which execute in a single clock cycle. Once an operation is performed
in the ALU, the result is transferred to the ACC, where additional operations, such as shifting,
can occur. Data that is input to the ALU can be scaled by the prescaler.
The following steps occur in the implementation of a typical ALU instruction:
1. Data is fetched from memory on the data bus,
2. Data is passed through the prescaler and the ALU, where the arithmetic is performed,
and
3. The result is moved into the ACC.
The ALU operates on 16-bit words taken from data memory or derived from immediate
instructions. In addition to the usual arithmetic instructions, the ALU can perform Boolean
operations, thereby facilitating the bit manipulation ability required of high-speed controller.
One input to the ALU is always supplied by the ACC. The other input can be transferred from
the PREG of the multiplier, the ACCB, or the output of the prescaler. After the ALU has
performed the arithmetic or logical operation, the result is stored in the ACC.

17. Write short notes about parallel logic unit.

The parallel logic unit (PLU) can directly set, clear, tests, or toggle multiple bits in control/status
register for any data memory location. The PLU provides a direct logic operation path to data
memory values without affecting the contents of the ACC or the PREG.

18. What is meant by auxiliary register file?

The auxiliary register file contains eight memory-mapped auxiliary registers (AR0-AR7), which
can be used for indirect addressing of the data memory or for temporary data storage.
Indirect auxiliary register addressing allows placement of the data memory address of an
instruction operand into one of the AR. The ARs are pointed to by a 3-bit auxiliary register
pointer (ARP) that is loaded with a value from 0-7, designating AR0-AR7, respectively.

19. Write short notes about circular registers in C5X.

The ‘C5x devices support two concurrent circular buffers operating in conjunction with user-
specified auxiliary register. Two 16-bit circular buffer start registers (CBSR1 and CBSR2)
indicate the address where the circular buffer starts. Two 16-bit circular buffer end registers
(CBER1 and CBER2) indicate the address where the circular buffer ends. The 16-bit circular
buffer control register (CBCR) controls the operation of these circular buffers and identifies the
auxiliary registers to be used.

20. List the on-chip peripherals in 5X.

The C5X DSP on-chip peripherals available are as follows:
1. Clock Generator
2. Hardware Timer
3. Software-Programmable Wait-State Generators
4. Parallel I/O Ports
5. Host Port Interface (HPI)
6. Serial Port
7. Buffered Serial Port (BSP)
8. Time-Division Multiplexed (TDM) Serial Port
9. User-Maskable Interrupts

21. What are the different buses of TMS320C5X?

The C5X architecture has four buses and their functions are as follows:
Program bus (PB)
Program address bus (PAB)
Data read bus (DB)
Data read address bus (DAB)
 22. What is the function Program bus (PB)?
It carries the instruction code and immediate operands from program memory space to
the CPU.

23. What is the function of PAB?

Program address bus (PAB):
It provides addresses to program memory space for both reads and writes.

24. What is the function of DB?

Data read bus (DB):
It interconnects various elements of the CPU to data memory space.

25. What is the function of DAB?

Data read address bus (DAB):
It provides the address to access the data memory space.

26. What are the various addressing modes of TMS processor?

 Immediate. Register
 Register indirect
 Indexed

27. What are the various number representations in digital computer?

 Fixed point
 Floating point
 Block floating point

28. What are the factors that influence selection of DSPs?

* Architectura lfeatures
* Execution speed
* Type of arithmetic
* Word length

29. What are the classification digital signal processors?

The digital signal processors are classified as
(i) General purpose digital signal processors.
(ii) Special purpose digital signal processors.

30. What are the applications of PDSPs?

Digital cell phones, automated inspection, voicemail, motor control, video conferencing,
noise cancellation, medical imaging, speech synthesis, satellite communication etc.
31. Give some examples for fixed point DSPs.
TM32OC50, TMS320C54, TMS320C55, ADSP-219x, ADSP-219xx.

32. Give some example for floating point DSPs?

TMS320C3x, TMS320C67x, ADSP-21xxx

33. What is pipelining?

Pipelining a processor means breaking down its instruction into a series of discrete pipeline
stages which can be completed in sequence by specialized hardware.

34. What is pipeline depth?

The number of pipeline stages is referred to as the pipeline depth.

35. What are the advantages of VLIW architecture?

Advantages of VLIW architecture
 Increased performance
 Better compiler targets
 Potentially easier to program
 Potentially scalable
 Can add more execution units; allow more instructions to be packed into the VLIW
instruction.

36. What are the disadvantages of VLIW architecture?

Disadvantages of VLIW architecture
 New kind of programmer/compiler complexity
 Program must keep track of instruction scheduling
 Increased memory use
 High power consumption

37. What is the pipeline depth of TMS320C50 and TMS320C54x?

 TMS320C50 – 4
 TMS320C54x – 6

38. What are the different stages in pipelining?

 The fetch phase
 The decode phase
 Memory read phase
 The execute phase

39. List the various registers used with ARAU.

Eight auxiliary registers (AR0 – AR7)
Auxiliary register pointer (ARP) Unsigned 16-bit ALU

40. What are the control processing units of ‘C5x?

The central processing unit consists of the following elements:
 Central arithmetic logic unit (CALU)
  Parallel logic unit (PLU)
 Auxiliary register arithmetic unit (ARAU)
 Memory mapped registers
 Program controller

41. What is the function of parallel logic unit?

The parallel logic unit is a second logic unit that executes logic operations of data without
affecting the contents of accumulator.

42. List the on chip peripherals in ‘C5x.

The on-chip peripherals interfaces connected to the ‘C5x CPU include
 Clock generator
 Hardware timer
 Software programmable wait state generators
 General purpose I/O pins

43. What are the arithmetic instructions of ‘C5x?

ADD, ADDB, ADDC, SUB, SUBB, MPY, MPYU

44. What are the shift instructions?

ROR, ROL, ROLB, RORB, BSAR.

45. What are the general purpose I/O pins?

 Branch control input (BIO)
 External flag (XF)

46. What are the logical instructions of ‘C5x?

AND, ANDB, OR, ORB, XOR, XORB

47. What are load/store instructions?

LACB, LACC, LACL, LAMM, LAR, SACB, SACH, SACL, SAR, SAMM.

48. Mention the addressing modes available in TMS320C5X processor?

1. Direct addressing mode
2. Indirect addressing mode
3. Circular addressing mode
4. Immediate addressing
5. Register addressing
6. Memory mapped register addressing

49. Give the features of DSPs?

* Architectural features
* Execution speed
* Type of arithmetic
* Word length
50. What are the various ports are available in DSP Processor?
 Parallel I/O ports
 Serial port interface
 Buffered serial port
 Time-division multiplexed (TDM) serial port
 Host port interface
 User unmask able interrupts

51. What is function of NOP instruction?

* NOP- No operation
* Perform no operation.

52. What is function of ZAC instruction?

ZAC – Zero accumulator
Clear the contents of accumulator to zero.

53. Give the function of BIT instruction.

BIT – Test bit
Copy the specified bit of the data memory value to the TC bit in ST1.

54. Mention the function of B instruction.

B – Branch conditionally-Branch to the specified program memory address. Modify the current
AR and ARP as specified.

55. What is use of ADD instruction?

ADD – Add to accumulator with shift.
Add the content of addressed data memory location or an immediate value of accumulator, if a
shift is specified, left-shift the data before adds. During shifting, low- order bits are Zero-filled,
and high-order bits are sign extended if SXM=1.

56. Give the advantages of DSPs?

Architectural features, Execution speed, Type of arithmetic, Word length

57. Give the applications of DSP Processors?

Digital cell phones, automated inspection, voicemail, motor control, video conferencing,
noise cancellation, medical imaging, speech synthesis, satellite communication etc.

Part-B
1. Explain in detail about the applications of PDSP (11)

2. Explain briefly (11)

(i) Von Neumann architecture

(ii) Harvard architecture

 (iii)VLIW architecture

3. Explain in detail about (11)

(i) MAC unit

(ii) Pipelining

4. Draw and explain the architecture of TMS 320C5x processor (11)

5. Explain in detail about the Addressing modes of TMS 320C50 (11)

6. Explain in detail about the Addressing modes of TMS 320C6X (11)