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UNIT I

1. Define DSP.

DSP - Digital Signal Processing.

It is defined as changing or analyzing information which is measured as discrete time sequences.

* signal in

* Analog to Digital converter

* Digital Signal processor

* Digital to Analog converter

* signal out

* Veracity

* Simplicity

* Repeatability

* Telecommunication – spread spectrum, data communication

* Biomedical – ECG analysis, Scanners.

* Speech/audio – speech recognition

* Military – SONAR, RADAR

5. Define Signal.

Signal is a physical quantity that varies with respect to time , space or any other independent

variable.

(Or)

It is a mathematical representation of the system

Eg: y(t) = t. and x(t)= sin t.

6. Define system.

A set of components that are connected together to perform the particular task. E.g.; Filters

(i) Discrete time signal

(ii) Continuous time signal

Discrete time signals are defined only at discrete times, and for these signals, the independent

variable takes on only a discrete set of values.

Classification of discrete time signal:

1. Periodic and A periodic signal

2. Even and Odd signal

EE T65 DIGITAL SIGNAL PROCESSING

Continuous time signals are defined for a continuous of values of the Independent variable. In

the case of continuous time signals the independent Variable is continuous.

For example:

(i) A speech signal as a function of time

(ii) Atmospheric pressure as a function of altitude

Classification of continuous time signal:

(i) Periodic and A periodic signal

(ii) Even and Odd signal

Discrete time Unit impulse is

defined as δ[n]={1, n=0; }

{ 0, otherwise.}

Unit impulse is also known as unit sample. Discrete time unit step signal is defined by

U[n]={0,n=0}

{1,n>= 0}

A discrete time signal is said to be even when, x[-n]=x[n].

The continuous time signal is said to be even when, x(-t)= x(t)

For example, Cosine wave is an even signal. The discrete time signal is said to be odd when x[-

n]= -x[n]

The continuous time signal is said to be odd when x(-t)= -x(t)

Odd signals are also known as nonsymmetrical signal. Sine wave signal is an odd signal.

A signal is said to be energy signal if it have finite energy and zero power. A signal is said to be

power signal if it have infinite energy and finite power. If the above two conditions are not

satisfied then the signal is said to be Neither energy nor power signal

The analog signal is a continuous function of independent variables. The analog Signal is defined

for every instant of independent variable and so magnitude of Independent variable is continuous

in the specified range. Here both the independent Variable and magnitude are continuous.

The digital signal is same as discrete signal except that the magnitude of signal is Quantized

a. Graphical representation

b. Functional representations

c. Tabular representation

d. Sequence representation.

EE T65 DIGITAL SIGNAL PROCESSING

If the discrete time signal repeated after equal samples of time then it is called periodic signal.

When the discrete time signal x[n] satisfies the condition x[n+N]=x(n), then it is called periodic

signal with fundamental period N samples.

If x(n) * x(n+N) then it is called non periodic signals.

The types of discrete time signals are

* Energy and power signals

* Periodic and A periodic signals

* Symmetric (Even) and Ant symmetric (Odd) signals

If E is finite i.e. 0<E<α , then x (n) is called energy signal.

If P is finite i.e. 0<P<α , then the signal x(n) is called a power signal.

The sampling frequency must be at least twice the maximum frequency present in the signal.

That is Fs = > 2fm Where, Fs = sampling frequency, Fm = maximum frequency

It is the minimumrate at which a signal can be sampled and still reconstructed fromits samples.

Nyquist rate is always equal to 2fm.

The superimposition of high frequency behavior on to the low frequency behavior is referred as

aliasing. This effect is also referred as folding.

To avoid the aliasing effect the sampling frequency must be twice the maximum frequency

present in the signal.

It is also referred as up sampling. that is , increasing the sampling rate.

A filter that is used to reject high frequency signals before it is sampled to remove the aliasing of

unwanted high frequency signals is called an ant aliasing filter.

* zero order hold

* first order hold

EE T65 DIGITAL SIGNAL PROCESSING

Sampling rate = number of samples / second.

The output of the system y(n) is obtained for the unit step input u(n) then it is said to be step

response of the system.

The Transfer function of DT system is defined as the ratio of Z transform of the system output to

the input. That is , H(z)=Y(z)/X(z).

The impulse response is the output produced by DT system when unit impulse is applied at the

input. The impulse response is denoted by h(n). The impulse response h(n) is obtained by taking

inverse Z transform from the transfer function H(z).

The causal system generates the output depending upon present &past inputs only. A causal

system is non anticipatory.

A linear system should satisfy superposition principle. A linear system should satisfy

F[ax1(n)+bx2(n)] = a y 1(n)+by2(n)

Where, y1(n)=F[x1(n)] & y2(n)=F[x2(n)]

1 .A system is time invariant if the behavior and characteristics of the system are fixed over time.

2. A system is time invariant if a time shift in the input signal results in an identical time shift in

the output signal.

3. For example, a time invariant system should produce y (t-t0) as the output

Continuous time signal: A signal x(t) is said to be continuous if it is defined for all time t.

Continuous time signal arise naturally when a physical waveform such as acoustics wave or

light wave is converted into a electrical signal. This is effected by means of transducer.(e.g.)

microphone, photocell.

Discrete time signal: A discrete time signal is defined only at discrete instants of time. The

independent variable has discrete values only, which are uniformly spaced. A discrete time

signal is often derived from the continuous time signal by sampling it at a uniform rate.

Continuous-time and discrete time signals, Even and odd signals, Periodic signals and non-

periodic signals Deterministic signal and Random signal Energy and Power signal

EE T65 DIGITAL SIGNAL PROCESSING

Continuous time and discrete time systems

Linear and Non-linear systems

Causal and Non-causal systems

Static and Dynamic systems

Time varying and time in-varying systems

Distributive parameters and Lumped parameters systems

Stable and Un-stable systems.

Even signal: continuous time signal x(t) is said to be even if it satisfies the condition

x(t)=x(-t) for all values of t.

Odd signal: he signal x(t) is said to be odd if it satisfies the condition x(-t)=-x(t) for all t. In

other words even signal is symmetric about the time origin or the vertical axis, but odd signals

are anti-symmetric about the vertical axis.

Deterministic Signal: deterministic signal is a signal about which there is no certainty with

respect to its value at any time. Accordingly we find that deterministic signals may be modeled

as completely specified functions of time.

Random signal: random signal is a signal about which there is uncertainty before its actual

occurrence. Such signal may be viewed as group of signals with each signal in the ensemble

having different wave forms

Energy signal: signal is referred as an energy signal, if and only if the total energy of the

signal satisfies the condition 0<E<∞. The total energy of the continuous time signal x (t) is

given as

E=limT→∞∫x2 (t)dt, integration limit from –T/2 to +T/2

Power signal: signal is said to be powered signal if it satisfies the condition 0<P<∞. The

average power of a continuous time signal is given by

P=limT→∞1/T∫x2(t)dt, integration limit is from-T/2 to +T/2.

Operations performed on dependent variables:

Amplitude scaling: y (t) =cx (t), where c is the scaling factor, x(t) is the continuous time

signal.

Addition: y (t)=x1(t)+x2(t)

Multiplication y (t)=x1(t)x2(t)

Differentiation: y (t)=d/dt x(t)

Integration (t) =∫x(t)dt

Operations performed on independent variables

Time shifting Amplitude

scaling Time reversal

EE T65 DIGITAL SIGNAL PROCESSING

The elementary signals serve as a building block for the construction of more complex

signals. They are also important in their own right, in that they may be used to model many

physical signals that occur in nature.There are five elementary signals. They are as follows

Unit step function

Unit impulse function

Ramp function

Exponential function

Sinusoidal function

Stability: A system is said to be stable if the input x(t) satisfies the condition(t)│≤Mx<∞ and the

output satisfies the condition │y(t)│≤My<∞ for all t.

Memory: A system is said to be memory if the output signal depends on the present and the past

inputs.

Invertibility: A system is said to be invertible if the input of the system can be recovered from

the system output.

Time invariance: A system is said to be time invariant if a time delay or advance of the input

signal leads to an identical time shift in the output signal.

Linearity: A system is said to be linear if it satisfies the super position principle

i.e.) R (ax1(t)+bx2(t))=ax1(t)+bx2(t)

A system is said to be memory system if its output signal at any time depends on the past

values of the input signal. Circuit with inductors capacitors are examples of memory system.

A system is said to be memory less system if the output at any time depends on the present

values of the input signal. An electronic circuit with resistors is an example for memory less

system.

A system is said to be invertible system if the input of the system can be recovered from the

system output. The set of operations needed to recover the input as the second system

connected in cascade with the given system such that the output signal of the second system is

equal to the input signal applied to the system.

H-1{y(t)}=H-1{H{x(t)}}.

44. What are time invariant systems?

A system is said to be time invariant system if a time delay or advance of the input signal leads

to an identical shift in the output signal. This implies that a time invariant system responds

identically no matter when the input signal is applied. It also satisfies the condition

R{x(n-k)}=y(n-k).

EE T65 DIGITAL SIGNAL PROCESSING

45. Is a discrete time signal described by the input output relation y[n]= rnx[n] time

invariant.

R{x[n-k]}=R(x[n]) / x[n]→x[n-k]

y[n-k]=y[n] / n→n-k

=rn-kx [n-k] -------------------(2)

Equations (1)≠Equation(2)

Hence the signal is time variant.

46. Show that the discrete time system described by the input-output relationship y[n]

=nx[n] is linear?

L.H.S:R{ a1x1[n]+b1x2[n] }=R{x[n]} /x[n] → a1x1[n]+b1x2[n]

= a1 nx1[n]+b1 nx2[n] -------------------(1)

R.H.S: a1 y1[n]+b1 y2[n]= a1 nx1[n]+b1 nx2[n] --------------------(2)

Equation(1)=Equation(2)

Hence the system is linear

47. What are the steps involved in digital signal processing?

Converting the analog signal to digital signal, this is performed by A/D converter

Processing Digital signal by digital system.

Converting the digital signal to analog signal, this is performed by D/A converter.

Speech processing – Speech compression & decompression for voice storage system

Communication – Elimination of noise by filtering and echo cancellation.

Bio-Medical – Spectrum analysis of ECG, EEG etc.

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

* Up sampling

* Down sampling

PART-B

1.For each of the following systems, determine whether the system is static stable, causal, linear

and time invariant

EE T65 DIGITAL SIGNAL PROCESSING

a. y(n) = e x(n)

b.y(n) = ax(n) +b

c. y(n) = Σnk=n0 x(k)

d.y(n) = Σn+1k= -∞ x(k)

e. y(n) = n x2(n)

f. y(n) = x(-n+2)

g.y(n) = nx(n)

h.y(n) = x(n) +C

i. y(n) = x(n) – x(n-1)

j. y(n) = x(-n)

k.y(n) = Δ x(n) where Δ x(n) = [x(n+1) – x(n)]

l. y(n) = g(n) x(n)

m. y(n) = x(n2)

n.y(n) = x2(n)

o.y(n) = cos x(n)

p.y(n) = x(n) cos ω0n

2.Compute the linear convolution of h(n) = {1,2,1} and x(n) ={1,-3,0,2,2}

3.Explain the concept of Energy and Power signals and determine whether the following are

energy or power signals

a. x(n) = (1/3)n u(n)

b.x(n) = sin (π / 4)n

4.The unit sample response h(n) of a system is represented by

h (n) = n2u(n+1) – 3 u(n) +2n u(n-1) for -5≤ n ≤5. Plot the unit sample response.

5.State and prove sampling theorem. How do you recover continuous signals from its samples?

Discuss the various parameters involved in sampling and reconstruction.

6.What is the input x(n) that will generate an output sequence

y(n) = {1,5,10,11,8,4,1} for a system with impulse response h(n) = {1,2,1}

7.Check whether the system defined by h(n) = [5 (1/2)n +4(1/3)n] u(n) is stable?

8.Explain the analog to digital conversion process and reconstruction of analog signal from

digital signal.

9.What are the advantages and disadvantages of digital signal processing compared with analog

signal processing?

10. Classify and explain different types of signals.

11. Explain the various elementary discrete time signals.

12. Explain the different types of mathematical operations that can be performed on a discrete

time signal.

13. Explain the different types of representation of discrete time signals.

14. Determine whether the systems having the following impulse responses are causal and stable

a. h(n) = 2n u(-n)

b.h(n) = sin nπ / 2

c. h(n) = sin nπ + δ (n)

d.h(n) = e2n u(n-1)

15. The impulse response of a linear time invariant system is h (n) = {1, 2, 1,-1}. Determine the

response of the system to the input signal x (n) = {1, 2, 3, 1}.

EE T65 DIGITAL SIGNAL PROCESSING

UNIT II

1. Define z transform?

The Z transform of a discrete time signal x(n) is defined as,

The region of convergence (ROC) is defined as the set of all values of z for Which X(z)

converges.

For causal system the roc is exterior to the circle of radius r. For anti causal system it is interior

to the circle of radius r.

4. Explain about the roc of causal and anti causal finite sequences

For causal system the roc is entire z plane except z=0. For anti causal system it is entire z plane

except z=α .

a. The roc is a ring or disk in the z plane centered at the origin.

b. The roc cannot contain any pole.

c. The roc must be a connected region

d. The roc of an LTI stable system contains the unit circle.

If z{x1(n)}=X1(z) and z{X2(n)}=x2(z) then,

z{ax1(n)+bx2(n)}=aX1(z)+bX2(z), where a&b are constants.

If z{x(n)}=X(z) then z{x(n-k)}=z-kX(z)

If z{x(n)}=X(z) then z{an x(n)}=X(a-1 z)

If z{x(n)}=X(z) then z{x(-n)}=X(z-1)

If z{x(n)}=X(z) & z{h(n)}=H(z) then, z{x(n)*h(n)}=X(z)H(z)

The ratio between z transform of out put signal y(z) to z transform of input signal x(z) is called

system function of the particular system.

EE T65 DIGITAL SIGNAL PROCESSING

All the poles of the system are with in the unit circle.

The sum of impulse response for all values of n is bounded.

Z-transform is used for analysis the both periodic and a periodic signals.

Z[ δ(n) ] = 1

The zeros of the system H(z) are the values of z for which H(z) = 0.

The poles of the system H(z) are the values of z for which H(z) = α.

Z [ A δ (n-m) ] =1.

The convolution property states that the convolution of two sequences in time domain is

equivalent to multiplication of their Z transforms.

20. What are the different methods to calculate the inverse Z transform?

i. Long division method

ii. Partial fraction expansion method

iii. Residue method

iv. Convolution method

The function of discrete time systems is to process a given input sequence to generate output

sequence. In practical discrete time systems, all signals are digital signals, and operations on

such signals also lead to digital signals. Such discrete time systems are called digital filter.

Linear & Non linear system

Causal & Non Causal system

Stable & Un stable system

Static & Dynamic systems

A system is said to be linear system if it satisfies Super position principle. Let us consider

x1(n) & x2(n) be the two input sequences & y1(n) & y2(n) are the responses respectively,

T[ax1(n) + bx2(n)] = a y1(n) + by2(n)

EE T65 DIGITAL SIGNAL PROCESSING

When the output of the system depends only upon the present input sample, then it is called

static system, otherwise if the system depends past values of input then it is called dynamic

system

When the output of the system depends only upon the present and past input sample, then it

is called causal system, otherwise if the system depends on future values of input then it is called

non-causal system

If y(n) is the response to an input x(n), then the response to an input

X(n) = x(n-n0) then y(n) = y(n-n0)

When the system satisfies above condition then it is said to shift in variant, otherwise it is

variant.

Impulse signal (n):

The impulse signal is defined as a signal having unit magnitude at n = 0 and

zero for other values of n.

(n) = 1; n = 0

0; n 0

Unit step signal u(n):

The unit step signal is defined as a signal having unit magnitude for all

values of n 0

u(n) = 1; n 0

0; n 0

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.

28. What are the basic elements used to construct the block diagram of discrete time

system?

The basic elements used to construct the block diagram of discrete time Systems are Adder,

Constant multiplier &Unit delay element.

The values of z for which z – transform converges is called region of convergence (ROC). The z-

transform has an infinite power series; hence it is necessary to mention the ROC along with z-

transform.

EE T65 DIGITAL SIGNAL PROCESSING

Linearity

Time Shifting

Frequency shift or Frequency translation

Time reversal

Partial fraction expansion

Power series expansion

Contour integration (Residue method)

A continuous time signal can be represented in its samples and recovered back if the

sampling frequency Fs 2B. Here ‘Fs’ is the sampling frequency and ‘B’ is the maximum

frequency present in the signal.

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.

1. Commutative property x(n) * h(n) = h(n) * x(n)

2. Associative property [x(n) * h1(n)]*h2(n) = x(n)*[h1(n) * h2(n)]

3. Distributive property x(n) *[ h1(n)+h2(n)] = [x(n)*h1(n)]+[x(n) * h2(n)]

Let us consider the discrete time signal x(n).Its DTFT is denoted as X(w).It is given as

X(w)= x(n)e

-jwn

The conditions are

• If x(n)is absolutely summable then |x(n)|<

• If x(n) is not absolutely summable then it should have finite energy for DTFT to exit.

The DTFT of unit sample is 1 for all values of w.

Periodicity

Linearity

Time shift

Frequency shift

EE T65 DIGITAL SIGNAL PROCESSING

Scaling

Differentiation in frequency domain

Time reversal

Convolution

Multiplication in time domain

Parseval’s theorem

DFT is defined as X(w)= x(n)e-jwn.

Here x(n) is the discrete time sequence

X(w) is the fourier transform ofx(n).

The method of appending zero in the given sequence is called as Zero padding.

A Sequence is said to be circularly even if it is symmetric about the point zero on the circle. x(N-

n)=x(n),1<=n<=N-1.

A Sequence is said to be circularly odd if it is anti-symmetric about point x(0) on the circle

A circularly folded sequence is represented as x ((-n)) N. It is obtained by plotting x (n) in

clockwise direction along the circle.

This property states that multiplication of two DFT is equal to circular convolution of their

sequence in time domain.

Consider the complex valued sequences x(n) and y(n).If

x(n)y*(n)=1/N X(k)Y*(k)

The Z transform of a discrete time signal x(n) is denoted by X(z) and is given by X(z)= x(n)Z-n.

46. Define N point DFT.

The DFT of discrete sequence x(n) is denoted by X(K). It is given by,

Here k=0,1,2…N-1

Since this summation is taken for N points, it is called as N-point DFT.

The DFT of unit impulse δ(n) is unity.

EE T65 DIGITAL SIGNAL PROCESSING

Linearity, P er i o di ci t y , C i r c u l a r s y m m e t r y , s y m m e t r y , T i m e s h i f t , F r e q u e n c y

shift, complex conjugate, convolution, correlation and Parseval’s theorem.

DFT of linear combination of two or more signals is equal to the sum of linear combination of

DFT of individual signal.

DFT of a finite length sequence results in a periodic sequence.

PART-B

1. Determine the Z-transform and ROC of

b. x(n) = n2an u(n)

c. x(n) = -1/3 (-1/4)n u(n) – 4/3 (2)n u(-n-1)

d. x(n) = an u(n) + bn u(n) + cn u(-n-1) , |a | < | b| < |c|

e. x(n) = cos ωn u(n)

f. x(n) = sin ω0n . u(n)

g. x(n) = an u(n)

h. x(n) = [ 3 (2n) – 4 (3n)] u(n)

2. Find the inverse Z-transform of

a. X(z) = z (z+1) / (z-0.5)3

b. X(z) = 1+3z-1 / 1 + 3z-1 + 2z-2

c. H(z) = 1 / [1 - 3z-1 + 0.5z-2] |z | > 1

2

d. X(z) = [z (z - 4z +5)] / [(z-3) (z-2) ( z-1)] for ROC |2 | < | z| < |3|, |z| > 3, |z|< 1

3. Determine the system function and pole zero pattern for the system described by difference

equation y (n) -0.6 y (n-1) +0.5 y (n-2) = x (n) – 0.7 x (n-2)

4. Determine the pole –zero plot for the system described by the difference equation

y (n) – 3/4 y (n-1) +1/8 y (n-2) = x(n) – x(n-1)

5. Explain the properties of Z-transform.

6. Perform the convolution of the following two sequences using Z-transforms.

x(n) = 0.2n u(n) and h(n) = (0.3)n u(n)

7.A causal LTI system has an impulse response h(n) for which the Z-transform is given by

H(z) = (1+z-1) / [(1 + 1/2z-1) (1 + 1/4z-1). What is the ROC of H (z)? Is the system stable?

8.Find the Z-transform X (z) of an input x (n) that will produce the output

y(n) = -1/3 (-1/4)n u(n) – 4/3 (2)n u(-n-1).Find the impulse response h (n) of the system.

9. Solve the difference equation y(n) -3y(n-1) – 4y(n-2) = 0, n ≥ 0 ,y(-1) = 5

10. Compute the response of the system

y(n) = 0.7 y(n-1)-0.12y(n-2) +x(n-1)+ x (n-2)to the input x(n) = n u(n)

11. What is ROC? Explain with an example.

12. A causal LTI IIR digital filter is characterized by a constant co-efficient difference equation

given by y(n) = x(n-1)-1.2x(n-2)+x(n-3)+1.3 y(n-1) – 1.04 y(n-2)+0.222y(n-3),obtain its transfer

function.

EE T65 DIGITAL SIGNAL PROCESSING

13. Determine the system function and impulse response of the system described by the

difference equation y(n) = x(n) +2x(n-1)- 4x(n-2) + x(n-3)

14. Solve the difference equation y(n) - 4y(n-1) - +4 y(n-2) = x(n) – x(n-1) with the initial

condition y(-1) = y(-2) = 1

15. Find the impulse response of the system described by the difference equation y(n) = 0.7 y(n-

1) -0.1 y(n-2) +2 x(n) – x(n-2)

16. Determine the z- transform and ROC of the signal x (n) = [3 (2n) – 4 (3n)] u(n).

17. State and prove convolution theorem in z-transform.

18. Given x(n) = δ(n) + 2 δ(n-1) and y(n) = 3 δ(n+1) + δ(n)- δ(n-1). Find x(n) * y(n) and

X(z).Y(z).

Unit – III

1. Define DFT of a discrete time sequence.

The DFT is used to convert a finite discrete time sequence x(n) to an N point frequency domain

sequence X(k).The N point DFT of a finite sequence x(n) of length L,(L<N) is defined as,

The Inverse DFT of the sequence of length N is defined as,

a. Periodicity

b. Linearity

c. Time reversal

d. Circular time shift e. Duality

f. Circular convolution

g. Symmetry

h. Circular symmetry

If x(k) is N-point DFT of a finite duration sequence x(n), then x(n+N) = x(n) for all n.

X(k+N) = X(k) for all k.

If X1(k) and X2(k) are N-point DFTs of finite duration sequences x1(n) and x2(n), then DFT [a

X1(n) + b X2(n)] = a X1(k) + b X2(k), a, b are constants.

If DFT[x(n)] =X(k), then DFT[x((-n))N] = DFT[x(N-n)] = X((-k))N = X(N-k)

EE T65 DIGITAL SIGNAL PROCESSING

Let x1(n) and x2(n) are finite duration sequences both of length n with DFTs x1(k) and x2(k).

If X3(k) = X1(k) X2(k), then the sequence X3(k) can be obtained by circular convolution.

DFT is used for analysis the both periodic and a periodic signals.

Let the sequence x (n) has a length L. If we want to find the N-point DFT (N>L) of the sequence

x(n), we have to add (N-L) zeros to the sequence x(n). This is known as Zero padding.

The uses of zero padding are

1) We can get better display of the frequency spectrum.

2) With zero padding the DFT can be used in linear filtering.

The direct evaluation of DFT requires N complex multiplications and N – N complex additions.

Thus for large values of N direct evaluation of the DFT is difficult. By using FFT algorithm the

number of complex computations can be reduced. So we use FFT

The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use of the

symmetry and periodicity properties of twiddle factor to effectively reduce the DFT computation

time. It is based on the fundamental principle of decomposing the mutation of DFT of a sequence

of length N into successively smaller DFTs.

12. How many multiplications and additions are required to compute N point DFT using

Radix-2 FFT?

The number of multiplications and additions required to compute N point DFT Using radix-2

FFT are N log2 N and N/2 log2 N respectively.

The FFT algorithm is most efficient in calculating N point DFT. If the number of output points N

can be expressed as a power of 2 that is N=2M, where M is an integer, then this algorithm is

known as radix-2 algorithm.

Decimation-In-Time algorithm is used to calculate the DFT of a N point sequence. The idea is to

break the N point sequence into two sequences, the DFTs of which can be combined to give the

DFT of the original N point sequence. This algorithm is called DIT because the sequence x(n) is

often spitted into smaller subsequences.

It is a popular form of the FFT algorithm. In this the output sequence X(k) is divided into smaller

and smaller sub-sequences , that is why the name Decimation In Frequency.

EE T65 DIGITAL SIGNAL PROCESSING

The applications of FFT algorithm includes

1) Linear filtering

2) Correlation

3) Spectrum analysis

The radix 2 DIT FFT is an efficient algorithm for computing DFT. The idea is to break N point

sequence in to two sequences, the DFT of which can be combined to give DFT of the original N-

point sequence. Initially the N point sequence is divided in to two N/2 point sequences, on the

basis of odd and even and the DFTs of them are evaluated and combined to give N-point

sequence.

DTFT output is continuous in time whereas DFT output is Discrete in time.

19. What are the differences and similarities between DIF and DIT algorithms?

Differences:

1) The input is bit reversed while the output is in natural order for DIT, whereas for DIF the

output is bit reversed while the input is in natural order.

2) The DIF butterfly is slightly different from the DIT butterfly, the difference being that the

complex multiplication takes place after the add-subtract operation in DIF.

Similarities:

Both algorithms require same number of operations to compute the DFT. Both algorithms can be

done in place and both need to perform bit reversal at some place during the computation.

1. The radix 2 DIFFFT is an efficient algorithm for computing DFT in this the output sequence

x(k) is divided in to smaller and smaller.

2. The idea is to break N point sequence in to two sequences ,x1(n) and x2(n) consisting of the

first N/2 points of x(n)and last N/2 points of x(n) respectively. Then we find N/2 point sequences

f(n) and g(nSimilarly).

3. The N/2 DFT s are divided and expressed in to the combination of N/4 point DFT’ s. This

process is continued until we left with 2-point DFT’s.

21. What are the differences between DIT and DIF algorithms?

* For DIT the input is bit reversed and the output is in natural order, and in DIF the input is in

natural order and output is bit reversed.

* In butterfly the phase factor is multiplied before the add and subtract operation but in DIF it is

multiplied after add-subtract operation.

An algorithm that uses the same location to store both the input and output sequence is called in-

place algorithm.

EE T65 DIGITAL SIGNAL PROCESSING

DIT – Time is decimated and input is bi reversed format output in natural order

DIF – Frequency is decimated and input is natural order output is bit reversed Format.

There are 3 stages are available for 8 Point DFT

25. How many multiplication terms are required for doing DFT by expressional?

Expression Method and FFT method

Expression –N2

FFT - N /2 logN

It is a finite duration discrete frequency sequence which is obtained by sampling one period of

Fourier transform. Sampling is done at ‘N ‘equally spaced points over the period extending from

ω = 0 to ω = 2π.

The DFT of unit impulse δ(n) is unity.

28. Why the result of circular and linear convolution is not same ?

Circular convolution contains same number of samples as that of x(n) and h(n), while in linear

convolution, number of samples in the result(N) are, N =L +M -1. Where, L = Number of

samples in x(n). M = Number of samples in h(n). That is why the result of linear and circular

convolution is not same.

29. How to obtain same result from linear and circular convolution?

* Calculate the value of ‘N’, that means number of samples contained in linear convolution.

* By doing zero padding make the length of every sequence equal to number of samples

contained in linear convolution.

* Perform the circular convolution. The result of linear and circular convolution will be same.

30. How will you perform linear convolution from circular convolution?

* Calculate the value of ‘N’, that means number of samples contained in linear convolution.

* By doing zero padding make the length of every sequence equal to number of samples

contained in linear convolution.

* Perform the circular convolution. The result of linear and circular convolution will be same.

31. What methods are used to do linear filtering of long data sequences?

* Overlap save method.

* Overlap add method.

EE T65 DIGITAL SIGNAL PROCESSING

For the computation of N-point DFT, N2 complex multiplication and N2 – N complex additions

are required. If the value of N is large then the number of computations will go into lakhs. This

proves inefficiency of direct DFT computation.

33. What is the way to reduce number of arithmetic operations during DFT computation?

Numbers of arithmetic operations involved in the computation of DFT are greatly reduced by

using different FFT algorithms as follows,

• Radix-2 FFT algorithm.

- Radix-2 Decimation In Time (DIT) algorithm.

- Radix-2 Decimation In Frequency (DIF) algorithm.

• Radix-4 FFT algorithm.

The two different factors are,

* Twiddle factor is periodic.

* Twiddle factor is symmetric.

Addition of one zero after each sample in x(n) is called up sampling process. Due to this process,

the entire DFT repeats one time.

Correlation is the basic process of doing linear filtering using FFT. The correlation is nothing but

the convolution with one of the sequence, folded. Thus, by folding the sequence h(n), we can

compute the linear filtering (convolution) using FFT.

FFT algorithms, for computing the DFT when the size N is a power of 2 and when it is a power

of 4

38. Compare the number of complex multiplications required for direct calculation and

FFT evolution of N – point DFT if N = 1024.

The number of complex multiplications required using direct computation is N2 = 10242

=1048576. The number of complex multiplications required using FFT is (N/2) log2N = (1024/2)

log21024= 5120.

"Bit reversal" is just what it sounds like: reversing the bits in a binary word from left to write.

Therefore the MSB's become LSB's and the LSB's become MSB's. The data ordering required by

radix-2 FFT's turns out to be in "bit reversed" order, so bit-reversed indexes are used to combine

FFT stages.

40. Determine the number of multiplication required in finding 64-point DFT using Radix-

2 FFT algorithm.

EE T65 DIGITAL SIGNAL PROCESSING

The number of multiplications required to compute N point DFT Using radix-2 FFT is N

log2 N Here N = 64, Hence Number of multiplication = 64×log2 (64) = 384.

DTFT output is continuous in time whereas DFT output is Discrete in time.

42. What is circular time shift of sequence?

Shifting the sequenceN in time domain by ‘1’ samples is equivalent to multiplying the

sequence in frequency domain by W kl

For the computation of N-point DFT, N2 complex multiplications and N[N-1] Complex

additions are required. If the value of N is large than the number of computations will go into

lakhs. This proves inefficiency of direct DFT computation.

44. What is the way to reduce number of arithmetic operations during DFT computation?

Number of arithmetic operations involved in the computation of DFT is greatly reduced by using

different FFT algorithms as follows.

1. Radix-2 FFT algorithms.

-Radix-2 Decimation in Time (DIT) algorithm.

- Radix-2 Decimation in Frequency (DIF) algorithm.

2. Radix-4 FFT algorithm.

45. What is the computational complexity using FFT algorithm?

1. Complex multiplications = N/2 log2N

2. Complex additions = N log2N

Correlation is the basic process of doing linear filtering using FFT. The correlation is nothing

but the convolution with one of the sequence, folded. Thus, by folding the sequence h (n), we

can compute the linear filtering using FFT.

Let the sequence x (n) has a length L. If we want to find the N point DFT (N>L) of the sequence

x(n). This is known as zero padding. The uses of padding a sequence with zeros are

(i)We can get ‘better display’ of the frequency spectrum.

(ii) With zero padding, the DFT can be used in linear filtering.

The inverse DFT of an N point sequence X(K) is defined as If we take complex conjugate and

multiply by N, we get The right hand side of the above equation is DFT of the sequence X*(K) and

may be computed using any FFT algorithm. The desired output sequence x (n) can be then

obtained by complex conjugating the DFT of the above equation and dividing by N to give

1. Linear filtering

2. Correlation

3. Spectrum analysis

EE T65 DIGITAL SIGNAL PROCESSING

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.

PART-B

1. Perform circular convolution of the sequence using DFT and IDFT technique

x1(n) = {2, 1,2,1} x2 (n) = {0,1,2,3}

2. Compute the DFT of the sequence x(n) = {1,1,1,1,1,1,0,0}

3. From the first principles obtain the signal flow graph for computing 8 – point DFT using

radix-2 DIF-FFT algorithm. An 8 point sequence is given by x(n)={2,2,2,2,1,1,1,1} compute its

8 point DFT of x(n) by radix-2 DIF-FFT

4. Explain any five properties of DFT.

5. Derive DIF – FFT algorithm. Draw its basic butterfly structure and compute the DFT x(n) = (-

1)n using radix 2 DIF – FFT algorithm.

6. Compute the DFT of the sequence x (n) = 1/3 δ (n) – 1/3 δ (n-1) +1/3 δ (n -2)

7. i) Compute the DFT of the sequence x (n) = (-1)n

ii) What are the differences and similarities between DIT – FFT and DIF – FFT algorithms?

8. Compute 4-point DFT of the sequence x (n) = (0, 1, 2, 3)

9. Explain the procedure for finding IDFT using FFT algorithm

10. Derive the decimation-in-frequency radix-2 FFT algorithm for evaluating DFT of the

discrete-time sequence and draw flow graph for 8-point DFT computation.

11. Explain the calculation of inverse DFT using FFT algorithm.

12. Compute the N point DFT of x{n) = an u(n) for cases |a| < 1 and |a| = 1

Unit – IV

1. What is a digital filter?

A digital filter is a device that eliminates noise and extracts the signal of interest from other

signals.

* pass band

* stop band

* Cut-off frequency

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

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

EE T65 DIGITAL SIGNAL PROCESSING

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

Analog filters are designed using analog components (R,L,C)while digital filters are

implemented using difference equation and implemented using software.

* Low pass filter - LPF

* High pass filter – HPF

* Band pass filter - BPF

* Band stop filter – BSF

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

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.

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

Because the feedback connection is present from output side to input

* Direct form structure

* Cascade form structure

* Parallel form structure

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.

14. 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.

Implementation of direct form is very easy.

EE T65 DIGITAL SIGNAL PROCESSING

Both direct form structures are sensitive to the effects of quantization errors in the coefficients.

So practically not preferred

17. 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.

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.

* Reverse all signal flow graph directions.

* Change branching nodes into adders and vice-versa.

* Interchange input and output.

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

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)1/2

EE T65 DIGITAL SIGNAL PROCESSING

For smaller values of w there exist linear relationship between w and Ω.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.

The effect of the nonlinear 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 prewar Ping the critical frequencies.

28. Give the bilinear transform equation between s plane and z plane

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

29. 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.

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.

The mapping is non-linear and because of this, frequency warping effect takes place.

Filter is a frequency selective device, which amplify particular range of frequencies and attenuate

particular range of frequencies.

33. What are the types of digital filter according to their impulse response?

The two digital filters are

IIR(Infinite impulse response)filter

FIR (Finite Impulse Response) filter.

1. The phase distortion is introduced when the phase characteristics of a filter is Nonlinear with

in the desired frequency band

EE T65 DIGITAL SIGNAL PROCESSING

2. The delay distortion is introduced when the delay is not constant with in the desired frequency

band

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

filter.

It is suitable only for designing of low pass and band pass IIR digital filters with relatively small

resonant frequencies.

It has smallest transition bandwidth and also it is more efficient.

It is used to design other filters like HPF, BPF and band reject filters from LPF.

FIR filter is always stable.

A realizable filter can always be obtained.

FIR filter has a linear phase response.

The bilinear transformation method overcomes the effect of aliasing that is caused due to the

analog frequency response containing components at or beyond the nyquist frequency. The

bilinear transform is a method of compressing the infinite, straight analog frequency axis to a

finite one long enough to wrap around the unit circle only once.

S = (2/T)[ (Z-1) / (Z+1)]

The philosophy of this technique is to transform an analog prototype filter into an IIR discrete

time filter whose impulse response [h(n)] is a sampled version of the analog filter’s impulse

response, multiplied by T. This procedure involves choosing the response of the digital filter as

an equi-spaced sampled version of the analog filter.

When bilinear transformation is applied, the discrete time frequency is related continuous time

frequency as, Ω = 2tan-1ΩT/2. This equation shows that frequency relationship is highly

nonlinear. It is also called frequency warping. This effect can be nullified by applying pre-

warping. The specifications of equivalent analog filter are obtained by following relationship,

Ω = 2/T tan ω/2. This is called pre-warping relationship.

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,

EE T65 DIGITAL SIGNAL PROCESSING

Advantages: 1. Many to one mapping. 2. Linear frequency relationship between analog and its

transformed digital frequency,

Disadvantage: Aliasing

45. State the condition for a digital filter to be causal and stable

The response of the causal system to an input does not depend on future values of that input, but

depends only on the present and/or past values of the input.

A filter is said to be stable, bounded-input bounded output stable, if every bounded input

produces a bounded output. A bounded signal has amplitude that remains finite.

46. Mention any two procedures for digitizing the transfer function of an analog filter.

1. Impulse Invariant Technique

2. Bilinear Transform Technique

FIR IIR

Impulse response is finite Impulse Response is infinite

They have perfect linear phase They do not have perfect linear

phase

Non recursive Recursive

Greater flexibility to control the Less flexibility

shape of magnitude response

2. Find the value of major and minor axis. λ

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.

6. If n is even put s=0 and divide it by (1+e2)1/2

2. Find the transfer function from the value of N

3. Find øc

4. Find the transfer function ha(s) for the above value of øc by su s by that value.

EE T65 DIGITAL SIGNAL PROCESSING

The various properties of a systems are Stability, Memory, Invertibility, Time invariance &

Linearity

PART-B

1. With suitable examples, describe the realization of linear phase FIR filters

2. Convert the following analog transfer function H(s) = (s+0.2) / [(s+0.2)2 + 4] into equivalent

digital transfer function H (z) by using impulse invariance method assuming T= 1

sec.

3. Convert the following analog transfer function H(s) = 1 / (s+2) (s+4) into equivalent digital

transfer function H (z) by using bilinear transformation with T = 0.5 sec.

4. Design a high pass filter of length 7 samples with cut off frequency of 2 rad / sec using

Hamming window. Plot its magnitude and phase response.

5. For the constraints

0.8 ≤ │H(ω)│≤ 1.0 , 0 ≤ ω ≤ 0.2π

│ H(ω)│ ≤ 0.2, 0.6 π ≤ ω ≤ π

With T= 1 sec determine the system function H(z) for a Butterworth filter using bilinear

transformation.

6. Discuss about the window functions used in design of FIR filters

7. Design a digital Chebyshev filter satisfying the following constraints with T= 1 sec, using

Bilinear transformation.

0.707 ≤ │H (ω) │≤ 1.0, 0 ≤ ω ≤ π/2

│ H (ω) │ ≤ 0.2, 3π/4 ≤ ω ≤ π

8. Using the bilinear transformation and a low pass analog Butterworth prototype, design a low

pass digital filter operating at a rate of 20 KHz and having pass band extending to a 4 KHz with a

maximum pass band attenuation of 0.5 dB and stop band starting at 5KHzwith a minimum stop

band attenuation of 10 dB.

9. Using the bilinear transformation and a low pass analog Chebyshev type I prototype, design a

low pass digital filter operating at a rate of 20 KHz and having pass band extending to a 4 KHz

with a maximum pass band attenuation of 0.5 dB and stop band starting at 5KHzwith a minimum

stop band attenuation of 10 dB.

10. Design a low pass FIR filter of order 7 with cut off frequency Π/3 rad/sec using Hanning

window

11. Convert the following analog filter into digital using IIM method.

H(S) = S2 /(S2 + 0.3S+0.02)

12. Design and realize a low pass filter using a rectangular window by taking 9 samples of w(n)

and with a cutoff frequency of 1.2 rad/sec.

13. Find the order N and the transfer function of analog Chebychev low pass filter for the

following specification: Pass band ripple 3 dB and pass band cut off frequency 1 KHz, stop band

attenuation of 16 dB at stop band frequency of 2 KHz.

EE T65 DIGITAL SIGNAL PROCESSING

Unit – V

1. What are all the blocks are used to represent the CT signals by its samples?

* Sampler

* Quantizer

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

* Up sampling

* Down sampling

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

amplitude.

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.

Truncation & Rounding

7. Define truncation.

Truncating the sequence by multiplying with window function to get the finite value

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.

EE T65 DIGITAL SIGNAL PROCESSING

EE T65 DIGITAL SIGNAL PROCESSING

23. How would you relate the steady-state noise power due to quantization and the b bits

representing the binary sequence?

Where b is the number of bits excluding sign bit.

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

2. Overflow limit cycle oscillations

EE T65 DIGITAL SIGNAL PROCESSING

The autocorrelation of a sequence is the correlation of a sequence with its shifted version, and

this indicates how fast the signal changes.

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 filter co-efficient

Errors due to rounding the product in multiplications

Limit cycles due to product quantization and overflow in addition.

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.

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.

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.

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.

EE T65 DIGITAL SIGNAL PROCESSING

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.

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.

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.

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.

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.

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.

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.

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.

EE T65 DIGITAL SIGNAL PROCESSING

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

The overflow limit cycles can be eliminated either by using saturation arithmetic or by

scaling the input signal to the adder.

The saturation arithmetic introduces nonlinearity in the adder which creates signal

distortion.

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.

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

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.

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.

EE T65 DIGITAL SIGNAL PROCESSING

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.

The overflow limit cycles can be eliminated either by using saturation arithmetic

or by scaling the input signal to the adder.

The saturation arithmetic introduces nonlinearity in the adder which creates signal

distortion.

PART-B

1.Explain the quantization effects in design of digital filters.

2. Illustrate the impact of quantization of filter coefficients on the poles and zeros with an

example

3.Obtain the cascade and parallel realization of system described by difference equation

y(n) = -0.1 y(n-1) + 0.2 y(n-2) + 3x(n) +3.6 x(n-1) + 0.6 x(n-2)

4.Draw the structure for IIR filter in direct form – I and II for the following transfer function H

(z) = (2 + 3 z-1) (4+ 2 z-1 +3 z-2) / (1+0.6 z-1) (1+ z-1+0.5 z-2)

5.Obtain the direct form – I, direct form – II, cascade and parallel form of realization for the

system y(n) = -0.1 y(n-1) + 0.2 y(n-2) + 3 x(n) + 3.6 x (n-1) + 0.6 x(n-2)

6.Write short note on (a) Truncation and rounding. (b) Coefficient Quantization.

7.Explain about zero input limit cycle oscillations

8. Explain the structure of IIR system

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