1. f.

Define pipelining and men on its advantages.

Pipelining is a technique used in digital signal processing (DSP) and compu ng where
mul ple stages of computa on are overlapped. In pipelining, the output of one stage serves
as the input to the next, allowing mul ple opera ons to be processed in parallel, which
speeds up the overall process.

Advantages of Pipelining:

1. Increased throughput: More opera ons can be completed in the same amount of
me by execu ng mul ple stages concurrently.
 2. Reduced latency: The system can process new data without wai ng for previous
stages to complete.

3. Improved efficiency: Maximizes hardware resource u liza on by keeping all stages

ac ve.

4. Faster processing: Especially beneficial in real- me signal processing where quick

response mes are cri cal.

5. Lower power consump on: By op mizing the workload across stages, pipelining can
help reduce the overall energy consump on in certain systems.

Here are the answers for ques ons 1 (a) to (e):

1. a.

Express and sketch the con nuous unit impulse and step signals.

1. Unit Impulse δ(t)\delta(t):

The con nuous unit impulse func on is defined as:

δ(t)={∞if t=00if t≠0\delta(t) = \begin{cases} \in y & \text{if } t = 0 \\ 0 & \text{if } t \neq 0 \\
\end{cases}

It is an idealized func on that is infinite at t=0t = 0 and zero elsewhere. The integral of the
impulse is 1.
Sketch: A spike at t=0t = 0 with zero everywhere else.

2. Unit Step u(t)u(t):

The con nuous unit step func on is defined as:

u(t)={0if t<01if t≥0u(t) = \begin{cases} 0 & \text{if } t < 0 \\ 1 & \text{if } t \geq 0 \\
\end{cases}

It is a signal that starts at t=0t = 0 and remains at 1 for all t≥0t \geq 0.
Sketch: A horizontal line at 0 for t<0t < 0, and a horizontal line at 1 for t≥0t \geq 0.

1. b.

Find the DFT of x(n)={2,1,0,0}x(n) = \{2, 1, 0, 0\} using the DIF-FFT method.

The Discrete Fourier Transform (DFT) of a sequence x(n)x(n) with N=4N = 4 is calculated as:

X(k)=∑n=0N−1x(n)e−j2πknN,k=0,1,2,3X(k) = \sum_{n=0}^{N-1} x(n) e^{-j2\pi \frac{kn}{N}},

\quad k = 0, 1, 2, 3

Using the Decima on-In-Frequency (DIF) FFT method:

 1. Split the sequence x(n)={2,1,0,0}x(n) = \{2, 1, 0, 0\} into even and odd-indexed parts:

o Even part: xeven={2,0}x_{\text{even}} = \{2, 0\}

o Odd part: xodd={1,0}x_{\text{odd}} = \{1, 0\}

2. Apply FFT recursively on the even and odd parts. For xeven={2,0}x_{\text{even}} =
\{2, 0\}, the DFT is straigh orward, and similarly for the odd part.

3. Combine the results using the twiddle factors:

o The twiddle factors for N=4N = 4 are W4k=e−j2πk/4W_4^k = e^{-j2\pi k / 4},

and the DFT is calculated.

A er applying DIF-FFT, the DFT X(k)X(k) for k=0,1,2,3k = 0, 1, 2, 3 is:

X(0)=3,X(1)=1−j,X(2)=−1,X(3)=1+jX(0) = 3, \quad X(1) = 1 - j, \quad X(2) = -1, \quad X(3) = 1 + j

1. c.

How many address, mul pliers, and delay elements are required in Direct-I and Direct-II
structures?

1. Direct-I Structure (FIR filter):

o Address elements: NN (number of taps).

o Mul plier elements: NN (one mul plier per tap).

o Delay elements: N−1N-1 (each tap requires a delay for the input signal).

2. Direct-II Structure (FIR filter):

o Address elements: NN (number of taps).

o Mul plier elements: NN (one mul plier per tap).

o Delay elements: 2(N−1)2(N-1) (two sec ons for input and intermediate
states).

1. d.

Men on the advantages of FIR filter over IIR filters.

1. Stability: FIR filters are always stable since they have no feedback elements, while IIR
filters can become unstable due to feedback loops.
 2. Linear phase response: FIR filters can be designed to have a linear phase response,
ensuring minimal distor on in the output signal. IIR filters generally do not provide
linear phase.

3. Simpler design: FIR filters are easier to design for certain specifica ons, especially for
applica ons where phase linearity is crucial.

4. No feedback: FIR filters do not use feedback, which means they avoid the risk of
numerical instability.

5. Predictable behavior: FIR filters do not suffer from the instabili es that IIR filters may
experience due to their recursive structure.

1. e.

Highlight the needs of mul rate digital signal processing.

1. Efficient Sample Rate Conversion: Mul rate DSP allows efficient conversion between
different sample rates (e.g., upsampling or downsampling) without the need for
heavy computa onal resources.

2. Reduced Computa onal Complexity: By processing signals at different rates,

mul rate systems reduce the amount of computa on required, especially when
working with large datasets or real- me applica ons.

3. Improved Performance in Communica on: Techniques like OFDM (Orthogonal

Frequency Division Mul plexing) rely on mul rate signal processing to efficiently use
bandwidth and enhance data throughput.

4. Audio and Speech Processing: In speech synthesis, recogni on, and compression
systems, mul rate processing enables efficient handling of signals at different
sampling rates.

5. Filter Design: It enables designing more efficient filters by decima ng or

interpola ng the signal, which reduces the complexity of filtering opera ons in signal
processing systems.

6. Noise Reduc on: Mul rate processing can help in reducing aliasing and noise by
using suitable sampling rates at different stages of processing.

Let me know if you need further clarifica on or have any more ques ons!

Here are the answers for ques ons 1 (a) to (e):

1. a.

Define a stable system.

A stable system is a system where the output remains bounded for any bounded input. This
is also known as BIBO stability (Bounded Input, Bounded Output stability). In mathema cal
terms, if the input x(t)x(t) or x[n]x[n] is bounded (i.e., there exists some constant MM such
that ∣x(t)∣≤M|x(t)| \leq M or ∣x[n]∣≤M|x[n]| \leq M for all me), the output y(t)y(t) or
y[n]y[n] of the system will also be bounded.

Condi on for stability:

 For a discrete- me system, the system is stable if the impulse response h[n]h[n] is
absolutely summable:

∑n=−∞∞∣h[n]∣<∞\sum_{n=-\in y}^{\in y} |h[n]| < \in y

1. b.

List any two differences between DFT and FFT in terms of computa onal complexity.

1. Computa onal Complexity:

o DFT (Discrete Fourier Transform): The DFT of a sequence x[n]x[n] of length

NN requires O(N2)O(N^2) opera ons (complex mul plica ons), as it directly
computes the Fourier transform using the defini on.

o FFT (Fast Fourier Transform): The FFT is an op mized algorithm for

compu ng the DFT, which reduces the computa onal complexity to
O(Nlog⁡N)O(N \log N) opera ons. This makes the FFT much faster than the
direct computa on of the DFT for large NN.

2. Algorithm Efficiency:

o DFT: The DFT is inefficient for large datasets because of its quadra c me
complexity.

o FFT: The FFT uses divide-and-conquer techniques, spli ng the problem into
smaller sub-problems, resul ng in a significantly faster algorithm compared
to the DFT.

1. c.

List out any two proper es of DFT.

 1. Linearity:
The DFT is a linear opera on. If x1[n]x_1[n] and x2[n]x_2[n] are two sequences, then
the DFT of their sum is the sum of their DFTs:

DFT(x1[n]+x2[n])=DFT(x1[n])+DFT(x2[n])\text{DFT}(x_1[n] + x_2[n]) = \text{DFT}(x_1[n]) +

\text{DFT}(x_2[n])

2. Periodicity:
The DFT of a sequence x[n]x[n] is periodic with period NN, where NN is the length of
the sequence. That is, the frequency components repeat every NN samples:

X[k+N]=X[k],for all integers kX[k+N] = X[k], \quad \text{for all integers } k

1. d.

List any two advantages of digital filters over analog filters.

1. Stability:
Digital filters are inherently stable as they do not involve feedback elements that can
cause instability, unlike analog filters, which may become unstable due to component
varia ons or feedback.

2. Precision and Flexibility:

Digital filters allow for greater precision in the design of frequency response
characteris cs. They can be easily modified and adjusted through so ware, offering
flexibility that analog filters cannot provide without hardware changes.

1. e.

Define Gibbs phenomenon.

The Gibbs phenomenon refers to the overshoot that occurs when approxima ng a
discon nuous signal with a finite number of Fourier series terms. Even though the Fourier
series converges to the func on at all points where it is con nuous, at a jump discon nuity,
the series approximates the signal with an overshoot of approximately 9% of the signal’s
amplitude, regardless of the number of terms used.

This phenomenon indicates that while the Fourier series can approximate the shape of a
discon nuous signal, it cannot exactly match the sharp edges at the points of discon nuity.

Let me know if you need further explana ons or examples!

Here are the answers for ques ons 1 (a) to (f):

1. a.

Define a stable system.

A stable system is a system where the output remains bounded for any bounded input. This
is also known as BIBO stability (Bounded Input, Bounded Output stability). If the input
x(t)x(t) or x[n]x[n] is bounded, meaning there exists a constant MM such that ∣x(t)∣≤M|x(t)|
\leq M or ∣x[n]∣≤M|x[n]| \leq M for all me, then the output y(t)y(t) or y[n]y[n] will also be
bounded.

For discrete- me systems, the system is stable if the impulse response h[n]h[n] is absolutely
summable:

∑n=−∞∞∣h[n]∣<∞\sum_{n=-\in y}^{\in y} |h[n]| < \in y

1. b.

List any two differences between DFT and FFT in terms of computa onal complexity.

1. Computa onal Complexity:

o DFT (Discrete Fourier Transform): The DFT requires O(N2)O(N^2) opera ons
to compute the Fourier transform for a sequence of length NN.

o FFT (Fast Fourier Transform): The FFT reduces the computa onal complexity
to O(Nlog⁡N)O(N \log N), making it much faster than the DFT for large
sequences.

2. Efficiency:

o DFT: The DFT is inefficient for large datasets due to the quadra c growth in
computa onal effort.

o FFT: The FFT is op mized and uses a divide-and-conquer approach, making it

highly efficient and suitable for large datasets.

1. c.

List out any two proper es of DFT.

1. Linearity:
The DFT is a linear opera on. If x1[n]x_1[n] and x2[n]x_2[n] are two signals, then the
DFT of their sum is the sum of their DFTs:
DFT(x1[n]+x2[n])=DFT(x1[n])+DFT(x2[n])\text{DFT}(x_1[n] + x_2[n]) = \text{DFT}(x_1[n]) +
\text{DFT}(x_2[n])

2. Periodicity:
The DFT of a sequence is periodic with period NN, where NN is the length of the
sequence. That is:

X[k+N]=X[k],for all integers kX[k+N] = X[k], \quad \text{for all integers } k

1. d.

List any two advantages of digital filters over analog filters.

1. Stability:
Digital filters are inherently stable because they do not rely on analog components
(like resistors, capacitors) that may cause instability. Analog filters can become
unstable due to feedback or component varia ons.

2. Design Flexibility:
Digital filters offer greater flexibility as their characteris cs (e.g., frequency response)
can be modified through so ware without needing physical changes in the hardware,
unlike analog filters which require hardware adjustments.

1. e.

Define Gibbs phenomenon.

The Gibbs phenomenon refers to the oscillatory behavior that occurs when a discon nuous
func on is approximated using a finite number of terms in its Fourier series expansion.
Specifically, at a jump discon nuity, the Fourier series overshoots by about 9% of the
amplitude of the discon nuity, regardless of how many terms are included in the
approxima on. This means that even as the number of Fourier terms increases, the
overshoot does not go away, highligh ng the difficulty in exactly represen ng discon nuous
signals with Fourier series.

1. f.

List any two advantages of on-chip memory.

1. Faster Access Speed:

On-chip memory is located directly within the processor, leading to much faster
access speeds compared to off-chip memory, as there is no need for data to travel
over external buses.
 2. Lower Power Consump on:
On-chip memory consumes less power compared to off-chip memory because it
operates at lower voltage levels and is physically closer to the CPU, reducing the
need for long-distance data transmission, which consumes more energy.

Let me know if you need further explana ons!

Here are the answers for ques ons 1 (a) to (f):

1. a.

Find even and odd part of the signal x[n]=u[n]−u(n−5)x[n] = u[n] - u(n-5)

The given signal is:

x[n]=u[n]−u[n−5]x[n] = u[n] - u[n-5]

where u[n]u[n] is the unit step func on.

To find the even and odd parts of the signal, recall the defini ons:

 The even part of a signal is given by:

xe[n]=x[n]+x[−n]2x_e[n] = \frac{x[n] + x[-n]}{2}

 The odd part of a signal is given by:

xo[n]=x[n]−x[−n]2x_o[n] = \frac{x[n] - x[-n]}{2}

Even Part:
xe[n]=(u[n]−u[n−5])+(u[−n]−u[−n−5])2x_e[n] = \frac{(u[n] - u[n-5]) + (u[-n] - u[-n-5])}{2}

Odd Part:

xo[n]=(u[n]−u[n−5])−(u[−n]−u[−n−5])2x_o[n] = \frac{(u[n] - u[n-5]) - (u[-n] - u[-n-5])}{2}

Now, evaluate these parts based on the behavior of u[n]u[n] and u[n−5]u[n-5] for different
values of nn.

1. b.

Determine the signal x[n]x[n] whose Z-transform is given by X(z)=sin⁡(z)X(z) = \sin(z)

To find the signal x[n]x[n], we need to take the inverse Z-transform of X(z)=sin⁡(z)X(z) =
\sin(z).

The Z-transform of the sine func on is given by:

Z{sin⁡(n)}=zz2+1\mathcal{Z}\le \{ \sin(n) \right\} = \frac{z}{z^2 + 1}

So, if the Z-transform is X(z)=sin⁡(z)X(z) = \sin(z), the corresponding signal x[n]x[n] is:

x[n]=sin⁡(n)x[n] = \sin(n)

Thus, the signal x[n]x[n] is simply sin⁡(n)\sin(n).

1. c.

Find the N-point DFT of the signal x[n]=1x[n] = 1 for even nn and 00 for odd nn (i.e.,
0<n<N−10 < n < N-1)

The given signal is:

x[n]={1,if n is even0,if n is oddx[n] = \begin{cases} 1, & \text{if } n \text{ is even} \\ 0, &

\text{if } n \text{ is odd} \end{cases}

The N-point DFT of a sequence x[n]x[n] is given by:

X[k]=∑n=0N−1x[n]⋅e−j2πNknX[k] = \sum_{n=0}^{N-1} x[n] \cdot e^{-j \frac{2 \pi}{N} k n}

For this specific sequence, x[n]x[n] takes the value 1 for even indices and 0 for odd indices,
so you would only sum over the even-indexed values of x[n]x[n] when compu ng the DFT.

The final result will depend on the specific value of NN, but it will involve a sum of
exponen als where only even-indexed terms contribute.

1. d.
What is the warping effect?

The warping effect refers to the distor on of frequency scales when conver ng between
analog and digital filters, especially when using transforma ons like the bilinear transform.
This effect occurs because the mapping from the s-domain (analog frequency domain) to the
z-domain (digital frequency domain) compresses or "warps" the frequency scale.

For example, in the bilinear transform, the en re frequency axis is mapped non-linearly,
causing higher frequencies to be mapped to lower frequencies. This leads to a
misrepresenta on of the higher frequencies in the digital filter design. The warping effect
can be corrected by pre-warping the digital frequencies before applying the transforma on.

1. e.

Compare IIR and FIR filters.

Property IIR Filter FIR Filter

Infinite impulse response; uses Finite impulse response; no

Structure
feedback. feedback.

Can be unstable (depends on

Stability Always stable.
filter design).

Linear phase response (if designed

Phase Response Non-linear phase response.
with symmetry).

Computa onal Requires fewer coefficients for a Requires more coefficients for
Complexity given performance. similar performance.

Memory Higher memory requirements (due

Lower memory requirements.
Requirement to the finite length).

Easier to implement with no

Implementa on More complex due to feedback.
feedback.

1. f.

Define interpola on.

Interpola on is the process of es ma ng values between two known values. In the context
of signal processing, it refers to the process of increasing the sampling rate of a discrete Here
are the answers for ques ons 1 (a) to (e):
1. a.

Represent the following finite-dura on sequence as a sum of weighted impulse

sequences.

Let's say the sequence x[n]={1,2,3,4}x[n] = \{1, 2, 3, 4\} for n=0,1,2,3n = 0, 1, 2, 3 (for
example).

To represent this finite-dura on sequence as a sum of weighted impulse sequences, we

write it as:

x[n]=x[0]⋅δ[n]+x[1]⋅δ[n−1]+x[2]⋅δ[n−2]+x[3]⋅δ[n−3]x[n] = x[0] \cdot \delta[n] + x[1] \cdot

\delta[n-1] + x[2] \cdot \delta[n-2] + x[3] \cdot \delta[n-3]

For the given example x[n]={1,2,3,4}x[n] = \{1, 2, 3, 4\}, it would be:

x[n]=1⋅δ[n]+2⋅δ[n−1]+3⋅δ[n−2]+4⋅δ[n−3]x[n] = 1 \cdot \delta[n] + 2 \cdot \delta[n-1] + 3

\cdot \delta[n-2] + 4 \cdot \delta[n-3]

Where δ[n]\delta[n] is the unit impulse sequence.

1. b.

What is zero-padding? What are its uses?

Zero-padding is the process of adding zeros to a signal to increase its length, usually done to
make the length of the signal a power of 2 for faster computa on in algorithms like the Fast
Fourier Transform (FFT). This process does not change the signal's informa on content but
can make the processing more efficient.

Uses of zero-padding:

1. Improves Frequency Resolu on: Zero-padding in the me domain can improve the
frequency resolu on of the Discrete Fourier Transform (DFT), especially when using
the FFT.

2. Efficient Computa on: It helps in making the length of the signal a power of 2, which
allows the use of faster FFT algorithms (which operate more efficiently with power-
of-2 length sequences).

1. c.

List any two advantages of digital filters over analog filters.

 1. Precision and Accuracy:
Digital filters are not affected by component tolerances (e.g., resistors, capacitors)
like analog filters, making them more precise and accurate over me. They are less
prone to dri or inaccuracies due to environmental factors (e.g., temperature).

2. Flexibility in Design and Implementa on:

Digital filters can be easily modified through so ware and hardware reconfigura on,
allowing for changes in the filter parameters without needing to physically redesign
the filter components, which is not possible with analog filters.

1. d.

Define Gibbs phenomenon.

The Gibbs phenomenon refers to the overshoot that occurs when a discon nuous func on
is approximated by a finite number of terms in its Fourier series. Specifically, when
approxima ng a jump discon nuity, the Fourier series exhibits oscilla ons near the
discon nuity.

The overshoot reaches about 9% of the jump in the func on, and no ma er how many
terms are included in the Fourier series, the maximum overshoot remains the same. This
phenomenon highlights the limita ons of Fourier series in represen ng discon nuous
signals exactly.

1. e.

If the Z-transform of a sequence x(n)x(n) is X(z)X(z), then what is the Z-transform of a

sequence upsampled by a factor MM?

If the sequence x[n]x[n] has a Z-transform X(z)X(z), then the Z-transform of the upsampled
sequence by a factor MM, denoted as y[n]y[n], can be expressed as:

y[n]=x[nM]y[n] = x\le [\frac{n}{M}\right]

The Z-transform of the upsampled sequence is given by:

Y(z)=X(zM)Y(z) = X(z^M)

Thus, the Z-transform of the sequence upsampled by a factor MM is the original Z-transform
evaluated at zMz^M. This corresponds to the shi ing of the Z-transform along the frequency
axis by the factor MM.

Let me know if you need any further clarifica ons!

Here are the answers for ques ons 1 (a) to (f):

1. a.

Recognize the given signal is energy or power signal x(n)=0.6nu(n)x(n) = 0.6^n u(n)

To determine whether the given signal is an energy signal or a power signal, we need to
examine its energy and power.

For a discrete- me signal x[n]x[n], the energy EE and power PP are given by:

 Energy:

E=∑n=−∞∞∣x[n]∣2E = \sum_{n=-\in y}^{\in y} |x[n]|^2

 Power:

P=lim⁡N→∞12N+1∑n=−NN∣x[n]∣2P = \lim_{N \to \in y} \frac{1}{2N+1} \sum_{n=-N}^{N}

|x[n]|^2

For the given signal x(n)=0.6nu(n)x(n) = 0.6^n u(n), where u(n)u(n) is the unit step func on,
the signal is only defined for n≥0n \geq 0.

 The energy of this signal is finite because ∣0.6n∣2|0.6^n|^2 decays exponen ally.

 The power of the signal is zero because the signal is not periodic (it is a decaying
exponen al).

Thus, the given signal is an energy signal because the total energy is finite.

1. b.

Indicate the Z-transform for a given sequence x(n)=0.9nu(n)x(n) = 0.9^n u(n)

The Z-transform X(z)X(z) of a sequence x(n)=0.9nu(n)x(n) = 0.9^n u(n) is given by the

standard formula for the Z-transform of a geometric series:

X(z)=∑n=0∞x[n]z−nX(z) = \sum_{n=0}^{\in y} x[n] z^{-n}

Subs tu ng x[n]=0.9nx[n] = 0.9^n for n≥0n \geq 0:

X(z)=∑n=0∞(0.9n)z−n=∑n=0∞(0.9z)nX(z) = \sum_{n=0}^{\in y} (0.9^n) z^{-n} =

\sum_{n=0}^{\in y} \le ( \frac{0.9}{z} \right)^n

This is a geometric series with a common ra o 0.9z\frac{0.9}{z}. The sum of the series is
valid for ∣z∣>0.9|z| > 0.9, and it is given by:

X(z)=11−0.9z−1=zz−0.9X(z) = \frac{1}{1 - 0.9z^{-1}} = \frac{z}{z - 0.9}

1. c.

List out any two proper es of DTFT

1. Linearity:
The Discrete-Time Fourier Transform (DTFT) is a linear opera on, meaning that if
x[n]x[n] and y[n]y[n] have DTFTs X(ω)X(\omega) and Y(ω)Y(\omega), respec vely,
then for any constants aa and bb:

DTFT{a⋅x[n]+b⋅y[n]}=a⋅X(ω)+b⋅Y(ω)\mathcal{DTFT}\{ a \cdot x[n] + b \cdot y[n] \} = a \cdot

X(\omega) + b \cdot Y(\omega)

2. Time Shi ing Property:

If x[n]x[n] has a DTFT X(ω)X(\omega), then a me-shi ed sequence x[n−k]x[n-k] has a
DTFT of:

DTFT{x[n−k]}=e−jωkX(ω)\mathcal{DTFT}\{ x[n-k] \} = e^{-j\omega k} X(\omega)

1. d.

List any two advantages of digital filters over analog filters

1. No Component Dri :
Digital filters are not affected by temperature changes, component aging, or other
physical factors that cause analog filters to dri over me.

2. Precision and Exactness:

Digital filters provide greater precision in terms of filter characteris cs since they are
implemented using exact algorithms (so ware or hardware) and are not prone to
tolerance varia ons that analog components experience.

1. e.

Define interpola on

Interpola on in signal processing refers to the process of es ma ng values of a signal

between two known samples. This is typically used to increase the sampling rate of a
discrete signal by inser ng new samples between the original samples, allowing for a more
refined or smoother signal representa on.

Mathema cally, interpola on is the process of construc ng new data points within the
range of known data points. Common interpola on methods include linear interpola on,
polynomial interpola on, and sinc interpola on.
1. f.

List any two advantages of on-chip memory

1. Faster Access:
On-chip memory, such as SRAM or ROM, provides much faster data access mes
compared to off-chip memory because it is physically located on the same chip as the
processor, reducing latency.

2. Reduced Power Consump on:

On-chip memory consumes less power compared to off-chip memory, as there is no
need for external communica on between the processor and memory, which can
involve power-hungry buses and interfaces.

Let me know if you need further clarifica ons!

Here are the answers for ques ons 1 (a) to (f):

1. a.

Recognize the given signals are periodic or aperiodic signals. If periodic, men on the
fundamental period.

i) x[n]=u[n]+u[−n]x[n] = u[n] + u[-n]

 u[n]u[n] is the unit step func on, which is 0 for n<0n < 0 and 1 for n≥0n \geq 0.

 u[−n]u[-n] is the unit step func on that is 1 for n≤0n \leq 0 and 0 for n>0n > 0.

The sum u[n]+u[−n]u[n] + u[-n] gives a signal that is 1 for all n=0n = 0 and 0 elsewhere,
which is not periodic. It is a non-periodic signal because it does not repeat itself over me.

ii) x[n]=cos⁡(πn2)x[n] = \cos\le (\frac{\pi n}{2}\right)

 This signal is periodic because cosine func ons are periodic. The fundamental period
TT of a cosine func on cos⁡(ωn)\cos(\omega n) is given by T=2πωT =
\frac{2\pi}{\omega}.

 In this case, ω=π2\omega = \frac{\pi}{2}, so the fundamental period is:

T=2ππ2=4T = \frac{2\pi}{\frac{\pi}{2}} = 4

Thus, the signal is periodic with a fundamental period of 4.

1. b.

Indicate the Z-transform and ROC of the signal x[n]=anu(n3)x[n] = a^n

u\le (\frac{n}{3}\right), where ∣a∣<1|a| < 1

 The signal x[n]=anu(n3)x[n] = a^n u\le (\frac{n}{3}\right) suggests that it is scaled by

ana^n and is defined for n≥3n \geq 3 because of the u(n3)u\le (\frac{n}{3}\right),
which shi s the sequence.

The Z-transform of x[n]=anu[n−3]x[n] = a^n u[n-3] is:

X(z)=∑n=3∞anz−nX(z) = \sum_{n=3}^{\in y} a^n z^{-n}

This is a geometric series, and we can find its sum:

X(z)=a31−az−1for∣z∣>∣a∣X(z) = \frac{a^3}{1 - a z^{-1}} \quad \text{for} \quad |z| > |a|

Thus, the Z-transform is:

X(z)=a3z−aX(z) = \frac{a^3}{z - a}

The Region of Convergence (ROC) is:

∣z∣>∣a∣|z| > |a|

1. c.

Find the 4-point DFT of the sequence x(n)={0,1,2,3}x(n) = \{0, 1, 2, 3\} using analy cal
approach

The 4-point Discrete Fourier Transform (DFT) of a sequence x[n]={0,1,2,3}x[n] = \{0, 1, 2, 3\}
is computed using the DFT formula:

X[k]=∑n=03x[n]e−j2π4knX[k] = \sum_{n=0}^{3} x[n] e^{-j \frac{2\pi}{4} k n}

For each k=0,1,2,3k = 0, 1, 2, 3, we compute X[k]X[k]:

1. For k=0k = 0:

X[0]=0⋅e0+1⋅e0+2⋅e0+3⋅e0=0+1+2+3=6X[0] = 0 \cdot e^{0} + 1 \cdot e^{0} + 2 \cdot e^{0} +

3 \cdot e^{0} = 0 + 1 + 2 + 3 = 6

2. For k=1k = 1:

X[1]=0⋅e0+1⋅e−jπ2+2⋅e−jπ+3⋅e−j3π2=0+1(−j)+2(−1)+3(j)=−2+2jX[1] = 0 \cdot e^{0} + 1 \cdot

e^{-j \frac{\pi}{2}} + 2 \cdot e^{-j \pi} + 3 \cdot e^{-j \frac{3\pi}{2}} = 0 + 1(-j) + 2(-1) + 3(j) = -
2 + 2j

3. For k=2k = 2:
X[2]=0⋅e0+1⋅e−jπ+2⋅e−j2π+3⋅e−j3π=0+1(−1)+2(1)+3(−1)=−2X[2] = 0 \cdot e^{0} + 1 \cdot
e^{-j \pi} + 2 \cdot e^{-j 2\pi} + 3 \cdot e^{-j 3\pi} = 0 + 1(-1) + 2(1) + 3(-1) = -2

4. For k=3k = 3:

X[3]=0⋅e0+1⋅e−j3π2+2⋅e−j3π+3⋅e−j9π2=0+1(j)+2(−1)+3(−j)=−2−2jX[3] = 0 \cdot e^{0} + 1

\cdot e^{-j \frac{3\pi}{2}} + 2 \cdot e^{-j 3\pi} + 3 \cdot e^{-j \frac{9\pi}{2}} = 0 + 1(j) + 2(-1) +
3(-j) = -2 - 2j

Thus, the 4-point DFT of x(n)={0,1,2,3}x(n) = \{0, 1, 2, 3\} is:

X[0]=6,X[1]=−2+2j,X[2]=−2,X[3]=−2−2jX[0] = 6, \quad X[1] = -2 + 2j, \quad X[2] = -2, \quad

X[3] = -2 - 2j

1. d.

What is frequency warping effect?

Frequency warping refers to a phenomenon where the rela onship between the original
analog frequency and the digital frequency is distorted due to the mapping during the
conversion from analog to digital. This occurs, for instance, in the Bilinear Transform, which
maps the con nuous- me frequency axis to the discrete- me frequency axis.

The effect causes non-linear frequency scaling, par cularly at higher frequencies, leading to
a distor on known as frequency warping. This is par cularly important when designing
digital filters from analog prototypes.

1. e.

Define Gibbs phenomenon.

The Gibbs phenomenon refers to the overshoot that occurs when approxima ng a
discon nuous func on by a finite number of terms in its Fourier series. The overshoot at the
jump discon nuity is approximately 9% of the signal's amplitude, and it does not decrease,
even if an infinite number of terms are used in the Fourier series.

In other words, the Fourier series approxima on of a signal with a discon nuity will always
exhibit this overshoot at the point of discon nuity, regardless of how many terms are used.

1. f.

List any two advantages of on-chip memory.

 1. Faster Access Time:
On-chip memory is directly integrated with the processor, leading to faster access
mes compared to off-chip memory. This is because there is no need for external
buses or communica on channels.

2. Lower Power Consump on:

On-chip memory consumes less power compared to external memory because it
eliminates the need for power-hungry external connec ons. This is par cularly
beneficial in portable and embedded systems where power efficiency is crucial.

Let me know if you need further clarifica ons!

Here are the answers for ques ons 1 (a) to (e):

1. a.

Define a stable system.

A stable system is a system where the output remains bounded for any bounded input. This
property is o en referred to as BIBO (Bounded Input, Bounded Output) stability.
Mathema cally, for a discrete- me system, the system is stable if the impulse response
h[n]h[n] is absolutely summable, meaning:

∑n=−∞∞∣h[n]∣<∞\sum_{n=-\in y}^{\in y} |h[n]| < \in y

For con nuous- me systems, stability is defined by the condi on that the system's impulse
response h(t)h(t) is absolutely integrable:

∫−∞∞∣h(t)∣ dt<∞\int_{-\in y}^{\in y} |h(t)| \, dt < \in y

1. b.

List any two differences between DFT and FFT in terms of computa onal complexity.

1. Computa onal Complexity:

o DFT (Discrete Fourier Transform): The computa onal complexity of the DFT is
O(N2)O(N^2), where NN is the number of input samples. This is because the
DFT involves a double summa on for each output value.

o FFT (Fast Fourier Transform): The computa onal complexity of the FFT is
O(Nlog⁡N)O(N \log N), which is significantly faster than the DFT, especially
for large values of NN.

2. Efficiency:
 o DFT: DFT is a direct computa on and is less efficient for large datasets due to
its O(N2)O(N^2) complexity.

o FFT: The FFT algorithm op mizes the computa on by breaking the DFT into
smaller DFTs, improving efficiency and enabling faster computa ons.

1. c.

List any two differences between analog and digital filters.

1. Implementa on:

o Analog Filters: Analog filters are implemented using con nuous components
such as resistors, capacitors, and inductors. They work directly with
con nuous- me signals.

o Digital Filters: Digital filters are implemented using discrete components such
as digital circuits or microprocessors and work with discrete- me signals.

2. Precision and Flexibility:

o Analog Filters: Analog filters are subject to component tolerances, which can
affect their precision. They also have fixed characteris cs once designed.

o Digital Filters: Digital filters offer high precision because they are
implemented in so ware or hardware with digital processors, and they are
more flexible due to the ease of modifying the filter characteris cs through
so ware.

1. d.

Define Gibbs phenomenon.

The Gibbs phenomenon refers to the overshoot that occurs when approxima ng a
discon nuous func on using a finite number of terms in its Fourier series. Even as the
number of terms in the series approaches infinity, the overshoot remains at approximately
9% of the signal's amplitude at the discon nuity. This overshoot is inherent to the Fourier
series approxima on of sharp transi ons and does not decrease with more terms.

1. e.

Define Decima on.

Decima on is the process of reducing the sampling rate of a signal by a factor of MM. It
involves two main steps:
 1. Low-pass filtering: To avoid aliasing, the signal is first passed through a low-pass filter
to remove high-frequency components above the new Nyquist frequency (which is
half of the reduced sampling rate).

2. Downsampling: A er filtering, the signal is then downsampled by keeping every

MM-th sample and discarding the others.

Decima on is commonly used in mul rate signal processing, such as when conver ng from a
high sample rate to a lower one.

Let me know if you need more details or further explana ons!

Here are the answers for ques ons 1 (a) to (f):

1. a.

Test the system y(n)=n2x(n)y(n) = n^2 x(n) for me invariance.

To test if the system is me-invariant, we need to see if a me shi in the input causes the
same me shi in the output. If the system is me-invariant, shi ing the input by n0n_0
results in the same shi in the output.

 Let the output for the input x(n)x(n) be y(n)=n2x(n)y(n) = n^2 x(n).

 Now, if we shi the input by n0n_0, the new input becomes x(n−n0)x(n - n_0).

 The output corresponding to this shi ed input will be:

y(n−n0)=n2x(n−n0)y(n - n_0) = n^2 x(n - n_0)

However, for the system, the output corresponding to the shi ed input should be:

y(n−n0)=(n−n0)2x(n−n0)y(n - n_0) = (n - n_0)^2 x(n - n_0)

Since n2≠(n−n0)2n^2 \neq (n - n_0)^2, the system does not preserve the me shi and is
not me-invariant.

1. b.

Iden fy whether the given signal is an energy signal or power signal:

The given signal is x(n)=(12)nu(n)x(n) = \le ( \frac{1}{2} \right)^n u(n), where u(n)u(n) is the
unit step func on.

To determine if the signal is an energy signal or a power signal, we calculate the energy and
power of the signal:
  Energy of a signal x(n)x(n) is given by:

E=∑n=−∞∞∣x(n)∣2E = \sum_{n=-\in y}^{\in y} |x(n)|^2

 Power of a signal x(n)x(n) is given by:

P=lim⁡N→∞12N+1∑n=−NN∣x(n)∣2P = \lim_{N \to \in y} \frac{1}{2N+1} \sum_{n=-N}^{N}

|x(n)|^2

Since x(n)=(12)nu(n)x(n) = \le ( \frac{1}{2} \right)^n u(n), the energy calcula on will yield a
finite value, indica ng that this signal is an energy signal. The signal decays exponen ally
and does not have infinite power.

1. c.

State any two proper es of DFT (Discrete Fourier Transform).

1. Linearity: The DFT is a linear opera on. If x1(n)x_1(n) and x2(n)x_2(n) are two
signals, then the DFT of a linear combina on of them is the same linear combina on
of their individual DFTs:

DFT[ax1(n)+bx2(n)]=a DFT[x1(n)]+b DFT[x2(n)]\text{DFT}[a x_1(n) + b x_2(n)] = a \,

\text{DFT}[x_1(n)] + b \, \text{DFT}[x_2(n)]

2. Periodicity: The DFT of a sequence x(n)x(n) is periodic with period NN, where NN is
the length of the sequence. That is, X(k+N)=X(k)X(k + N) = X(k) for all kk.

1. d.

What is the aliasing problem in the impulse invariant method of designing digital filters?

The aliasing problem occurs when high-frequency components of a con nuous- me signal
are "folded" back into the lower frequency range during the sampling process. In the
impulse invariant method, which is used to convert analog filters to digital filters, aliasing
can cause the high-frequency components of the analog signal to overlap with the low-
frequency components of the digital signal, leading to incorrect or distorted results. To avoid
aliasing, the sampling rate must be chosen carefully (according to the Nyquist criterion), and
the analog filter should be designed with sufficient a enua on of high-frequency
components.

1. e.

List any two characteris cs of FIR filters designed using windows.

 1. Finite Dura on Impulse Response (FIR): FIR filters have a finite number of
coefficients, making them easy to design and implement. The number of coefficients
corresponds to the filter's order.

2. Linear Phase Response: FIR filters can be designed to have a linear phase response,
which ensures that all frequency components of the input signal are delayed by the
same amount, preven ng distor on in the signal.

1. f.

Define mul rate signal processing.

Mul rate signal processing refers to the use of different sampling rates within a system. It
involves opera ons like decima on (reducing the sampling rate) and interpola on
(increasing the sampling rate). Mul rate processing allows efficient handling of signals in
various applica ons, such as reducing the computa onal complexity in filter designs or
op mizing the use of resources in communica on systems. It is widely used in digital signal
processing systems for opera ons like speech processing, image compression, and data
transmission.

Let me know if you need any further clarifica on or details!

It seems like you're trying to copy the answers, but they're not appearing as you want. I can
reformat the answers for you in a cleaner and more consistent layout. Here's how it will
look:

1. a.

State the condi on for stability.

A system is considered stable if, for any bounded input, the output remains bounded. In
terms of the Z-transform, the condi on for stability is that the ROC (Region of Convergence)
of the system should include the unit circle in the complex plane. This means that for a
system to be stable:

∣z∣=1must lie within the ROC.|z| = 1 \quad \text{must lie within the ROC.}

1. b.

List any two differences between DFT and DTFT.

1. Discrete vs. Con nuous:

 o DFT (Discrete Fourier Transform): DFT applies to discrete- me signals and
provides a finite number of frequency components.

o DTFT (Discrete-Time Fourier Transform): DTFT applies to discrete- me signals

but provides a con nuous frequency spectrum.

2. Frequency Resolu on:

o DFT: The frequency resolu on of DFT is determined by the length of the

sequence NN, with NN discrete frequency points.

o DTFT: The DTFT provides a con nuous frequency spectrum with infinite
resolu on, as it is computed over all frequencies.

1. c.

List any two advantages of FIR over IIR.

1. Linear Phase Response:

o FIR filters can be designed to have a linear phase response, ensuring that all
frequency components of the signal are delayed by the same amount, which
avoids phase distor on.

2. Stability:

o FIR filters are inherently stable because they have no feedback components,
unlike IIR filters, which can become unstable due to feedback loops.

1. d.

Define Gibbs phenomenon.

The Gibbs phenomenon refers to the overshoot that occurs when approxima ng a
discon nuous func on with a finite number of terms from its Fourier series. Even though
adding more terms reduces the overshoot, the maximum overshoot remains approximately
9% of the signal's amplitude at the discon nuity, no ma er how many terms are used.

1. e.

Define Interpola on.

Interpola on is the process of increasing the sampling rate of a discrete signal. This involves
inser ng zeros (or performing other opera ons) between the original samples, followed by
filtering to smooth the signal and avoid aliasing. Interpola on is o en used when conver ng
signals to a higher resolu on or when reconstruc ng signals from a lower resolu on.
1. f.

What are the various methods of represen ng discrete- me signals?

Discrete- me signals can be represented in several ways, including:

1. Graphical Representa on: Plo ng the signal x[n]x[n] on a me-axis, showing the
values of the signal at discrete me intervals.

2. Mathema cal Representa on: Expressing the signal as a mathema cal func on of
nn, such as x[n]=sin⁡(n)x[n] = \sin(n) or using sequences.

3. Z-Transform Representa on: The Z-transform is used to represent the signal in the
complex plane, transforming the signal from the me domain to the frequency
domain.

4. Difference Equa on Representa on: Signals can also be represented through

recursive or non-recursive difference equa ons, which define how the signal evolves
over me.

Let me know if this forma ng works for you or if you'd like a different style!

signal by inser ng addi onal samples between the original samples, typically by using
techniques like zero-padding followed by low-pass filtering to reconstruct the signal.

In mathema cal terms, if x[n]x[n] is a signal with a given sampling rate, interpola on aims to
generate a new sequence y[n]y[n] with a higher sampling rate, where the new samples are
es mated values between the original samples.

Let me know if you need further clarifica ons!