Q 1.

Answer all questions

The even and odd parts of the signal x(t) can be sketched as follows:

Formulas

 Even part:
x_e(t) = 1/2[x(t) + x(-t)]

 Odd part:

x_o(t) = 1/2[x(t) - x(-t)]

Even part sketch

1. 1. Reflect x(t):
Reflect the signal x(t) across the y-axis to obtain x(-t).

2. 2. Add x(t) and x(-t):

Add the waveforms of x(t) and x(-t) point by point.

3. 3. Scale by 1/2:
Multiply the result by 1/2.

The even part, x_e(t), will be a rectangular pulse of amplitude 1/2, extending from -a to
a.

Odd part sketch

1. 1. Reflect x(t):

Reflect the signal x(t) across the y-axis to obtain x(-t).

2. 2. Subtract x(-t) from x(t):

Subtract the waveform of x(-t) from x(t) point by point.

3. 3. Scale by 1/2:
Multiply the result by 1/2.

The odd part, x_o(t), will have:

 Amplitude of 1/2 from 0 to a.

 Amplitude of -1/2 from -a to 0.

  Zero amplitude elsewhere.
C) Determine the Fourier series representation of the following discrete time signals x(n)=2cos√3 n

Absolutely! Let's break down how to find the Fourier series representation of the discrete-time signal
x(n) = 2cos(√3πn).

Understanding Discrete-Time Fourier Series (DTFS)

 Periodicity: A signal x(n) has a DTFS representation if it's periodic. This means x(n + N) = x(n)
for some integer N (the period).

 Fundamental Frequency: The fundamental frequency (ω₀) is related to the period by ω₀ =

2π/N.

 DTFS Formula: The DTFS coefficients (ak) are calculated as:

 ak = (1/N) * Σ[n=0 to N-1] x(n) * e^(-j2πkn/N)

And the signal is reconstructed as:

x(n) = Σ[k=0 to N-1] ak * e^(j2πkn/N)

Analyzing x(n) = 2cos(√3πn)

1. Periodicity Check:

o For the signal to be periodic, √3πn must be a rational multiple of 2π. In other words,
√3π/2π = √3/2 must be a rational number.

o However, √3 is irrational. Therefore, √3/2 is irrational. This means x(n) is not

periodic.

Consequences of Non-Periodicity

 No DTFS: Since x(n) is not periodic, it does not have a Discrete-Time Fourier Series (DTFS)
representation. What if the signal was periodic?
If the signal were periodic (e.g., x(n) = 2cos(πn/3)), you would follow these steps:

1. Find the Period (N): Determine the smallest integer N for which x(n + N) = x(n).
2. Calculate the Fundamental Frequency (ω₀): ω₀ = 2π/N.

3. Calculate the DTFS Coefficients (ak): Use the formula:

4. ak = (1/N) * Σ[n=0 to N-1] x(n) * e^(-j2πkn/N)

5. Write the DTFS Representation: Express x(n) as:

6. x(n) = Σ[k=0 to N-1] ak * e^(j2πkn/N)

Important Note: The key to DTFS is periodicity. If your signal is not periodic, you cannot represent it
using a DTFS.

In summary, the signal x(n) = 2cos(√3πn) is not periodic and therefore does not have a DTFS
representation
D) Write a note on bilinear transformation used in filter design
Absolutely! Let's discuss the bilinear transformation, a crucial technique in digital filter design.

Bilinear Transformation: Bridging Analog and Digital

The bilinear transformation is a mathematical mapping used to convert an analog filter's transfer
function, H(s), into a digital filter's transfer function, H(z). It allows us to leverage well-established
analog filter design techniques (like Butterworth, Chebyshev, and Elliptic filters) to create digital
filters.

The Transformation
The core of the bilinear transformation lies in the substitution:

s = (2/T) * (1 - z⁻¹) / (1 + z⁻¹)

Where:

 s: The Laplace transform variable (analog frequency domain).

 z: The Z-transform variable (digital frequency domain).

 T: The sampling period (1/fs, where fs is the sampling frequency).

Key Properties and Advantages

1. Mapping the Entire s-plane to the z-plane: The bilinear transformation maps the entire left-
half of the s-plane (stable analog filters) to the inside of the unit circle in the z-plane (stable
digital filters). This ensures stability is preserved during the transformation.

2. No Aliasing: Unlike some other methods (like impulse invariance), the bilinear
transformation avoids aliasing. This is because it maps the entire imaginary axis (analog
frequency) onto the unit circle (digital frequency) without overlap.

3. One-to-One Mapping: Each point in the s-plane maps to a unique point in the z-plane, and
vice versa.

Disadvantages and Considerations

1. Frequency Warping: The bilinear transformation introduces non-linear warping of the

frequency axis. This means that the relationship between analog frequency (Ω) and digital
frequency (ω) is not linear:

2. ω = 2 * arctan(ΩT/2)

This warping needs to be accounted for during the design process, especially when dealing with
filters with specific cutoff frequencies.

2. Pre-Warping: To compensate for frequency warping, we often pre-warp the critical

frequencies of the analog filter before applying the bilinear transformation. This involves
calculating the analog frequencies that will map to the desired digital frequencies after the
warping.

3. Non-Linear Phase: The bilinear transformation generally does not preserve linear phase
characteristics. If linear phase is critical, other techniques might be more suitable.
Design Procedure
1. Specify Digital Filter Requirements: Define the desired filter characteristics (cutoff
frequencies, passband ripple, stopband attenuation) in the digital domain.

2. Pre-Warp Critical Frequencies: Calculate the corresponding analog frequencies using the
inverse of the warping equation:

3. Ω = (2/T) * tan(ω/2)

4. Design the Analog Filter: Design an analog filter (Butterworth, Chebyshev, Elliptic) that
meets the pre-warped requirements.
5. Apply Bilinear Transformation: Substitute the bilinear transformation expression for 's' into
the analog filter's transfer function H(s) to obtain the digital filter's transfer function H(z).

6. Implement the Digital Filter: Implement the H(z) transfer function using a suitable digital
filter structure (e.g., direct form, cascade form).

Applications
The bilinear transformation is widely used in various digital signal processing applications, including:

 Audio Processing: Designing equalizers, filters, and effects.

 Communications: Implementing digital filters for channel equalization and signal

conditioning.

 Control Systems: Creating digital controllers based on analog designs.

 Image Processing: Designing filters for image enhancement and noise reduction.

In summary, the bilinear transformation is a powerful and widely used technique for converting
analog filters into digital filters. It offers advantages like stability preservation and no aliasing, but
requires careful consid

Q 2 a) State whether the following system is linear, causal, time-invariant and stable
y(n)=nx(n)+x(n+2)+y(n-2)

Let's analyze the system described by the difference equation:

y(n) = nx(n) + x(n+2) + y(n-2)

to determine if it's linear, causal, time-invariant, and stable.

1. Linearity
A system is linear if it satisfies the superposition principle:

 Homogeneity: If y1(n) is the output for x1(n), then ay1(n) is the output for ax1(n) (where 'a'
is a constant).

 Additivity: If y1(n) is the output for x1(n) and y2(n) is the output for x2(n), then y1(n) + y2(n)
is the output for x1(n) + x2(n).

Let's test:

 Input 1: x1(n) -> y1(n) = nx1(n) + x1(n+2) + y1(n-2)

  Input 2: x2(n) -> y2(n) = nx2(n) + x2(n+2) + y2(n-2)
Now, consider the input x3(n) = ax1(n) + bx2(n) (where 'a' and 'b' are constants).

The output y3(n) is:

y3(n) = n[ax1(n) + bx2(n)] + [ax1(n+2) + bx2(n+2)] + y3(n-2) y3(n) = a[nx1(n) + x1(n+2)] + b[nx2(n) +
x2(n+2)] + y3(n-2)
For linearity, we need y3(n) = ay1(n) + by2(n). Let's check:

ay1(n) + by2(n) = a[nx1(n) + x1(n+2) + y1(n-2)] + b[nx2(n) + x2(n+2) + y2(n-2)] ay1(n) + by2(n) =
a[nx1(n) + x1(n+2)] + b[nx2(n) + x2(n+2)] + ay1(n-2) + by2(n-2)
For linearity, we need y3(n-2) = ay1(n-2) + by2(n-2). This is true because the system is recursive.

Therefore, the system is linear.

2. Causality

A system is causal if the output y(n) depends only on present and past inputs, not future inputs.
In the given equation:

y(n) = nx(n) + x(n+2) + y(n-2)

 nx(n): Depends on the present input.

 x(n+2): Depends on the future input x(n+2).

 y(n-2): Depends on past output.

Since the system depends on x(n+2), which is a future input, the system is non-causal.
3. Time-Invariance

A system is time-invariant if a time shift in the input results in the same time shift in the output.

Let's test:

 Input: x(n) -> Output: y(n) = nx(n) + x(n+2) + y(n-2)

 Shifted Input: x(n-k) -> Output: y'(n) = nx(n-k) + x(n-k+2) + y'(n-2)

Now, let's look at the time-shifted output y(n-k):

y(n-k) = (n-k)x(n-k) + x(n-k+2) + y(n-k-2)

Comparing y'(n) and y(n-k), we see that they are not the same because of the 'n' term in 'nx(n)'.

Therefore, the system is time-variant.

4. Stability

A system is stable (BIBO - Bounded Input Bounded Output) if a bounded input produces a bounded
output.

Let's consider a bounded input |x(n)| < B for all n.

y(n) = nx(n) + x(n+2) + y(n-2)

The term 'nx(n)' can become unbounded as 'n' increases, even if x(n) is bounded.
Therefore, the system is unstable.

Summary

The system y(n) = nx(n) + x(n+2) + y(n-2) is:

 Linear: Yes

 Causal: No (non-causal)

 Time-Invariant: No (time-variant)
 Stable: No (unstable)

Sources and related content

Q 2 b) Determine the transfer function H(z) of the following system y(n)-0.5y(n-1)=x(n)

Let's find the transfer function H(z) for the system described by the difference equation:

y(n) - 0.5y(n-1) = x(n)

Steps to Find the Transfer Function H(z)

1. Take the Z-transform of both sides of the equation.

Recall that the Z-transform of y(n) is Y(z), the Z-transform of x(n) is X(z), and the Z-transform of y(n-k)
is z^(-k)Y(z).

Applying the Z-transform to the given equation:

Z[y(n)] - 0.5 * Z[y(n-1)] = Z[x(n)]

Y(z) - 0.5 * z^(-1)Y(z) = X(z)

2. Rearrange the equation to express Y(z) in terms of X(z).

Factor out Y(z) from the left side:

Y(z) * (1 - 0.5z^(-1)) = X(z)

Divide both sides by (1 - 0.5z^(-1)) to isolate Y(z)/X(z):

Y(z) / X(z) = 1 / (1 - 0.5z^(-1))

3. Recognize that H(z) = Y(z) / X(z).

The transfer function H(z) is defined as the ratio of the Z-transform of the output Y(z) to the Z-
transform of the input X(z):

H(z) = Y(z) / X(z)

Therefore:

H(z) = 1 / (1 - 0.5z^(-1))

4. Optionally, express H(z) in terms of positive powers of z.

Multiply the numerator and denominator by z:
H(z) = z / (z - 0.5)

Final Answer

The transfer function H(z) of the system y(n) - 0.5y(n-1) = x(n) is:

H(z) = 1 / (1 - 0.5z^(-1)) or H(z) = z / (z - 0.5)

(ii) Determine whether the given signals are periodic or not, if periodic find the fundamental period.
A) ( ) B) x ( ) = 3 cos(4 ) + 2 sin(
Absolutely! Let's analyze the periodicity of these signals.

A) sin(1.2πt)
1. General Form: The general form of a sinusoidal signal is sin(ωt), where ω is the angular
frequency.

2. Frequency: In this case, ω = 1.2π.

3. Period: The period (T) of a sinusoidal signal is given by T = 2π/ω.

o T = 2π / (1.2π) = 2 / 1.2 = 20 / 12 = 5 / 3

Therefore, T = 5/3.

Conclusion:

 Periodic: Yes, the signal is periodic.

 Fundamental Period: T = 5/3

B) x(t) = 3cos(4t) + 2sin(πt)

1. Component Periods:

o 3cos(4t): The period T1 = 2π/4 = π/2.

o 2sin(πt): The period T2 = 2π/π = 2.

2. Ratio of Periods: To determine if the sum of two periodic signals is periodic, we need to
check if the ratio of their periods is a rational number.

o T1 / T2 = (π/2) / 2 = π/4

3. Rationality:

o π is an irrational number.

o Therefore, π/4 is also an irrational number.

Conclusion:

 Periodic: No, the signal is not periodic because the ratio of the periods of its components is
irrational.

Summary
  A) sin(1.2πt): Periodic, with a fundamental period of 5/3.
 B) 3cos(4t) + 2sin(πt): Not periodic.

Let's break down how to show that the given equation is satisfied for a linear-phase FIR filter.

Understanding Linear-Phase FIR Filters

 Symmetry: Linear-phase FIR filters have impulse responses that are either symmetric (even
symmetry) or anti-symmetric (odd symmetry) around their midpoint.

 Group Delay: Linear-phase filters have a constant group delay (τ), meaning all frequency
components of the signal are delayed by the same amount.

Types of Linear-Phase FIR Filters

There are four types of linear-phase FIR filters, based on symmetry and length:
1. Type 1: Symmetric, odd length (M is odd)

2. Type 2: Symmetric, even length (M is even)

3. Type 3: Anti-symmetric, odd length (M is odd)

4. Type 4: Anti-symmetric, even length (M is even)

Derivation

1. Frequency Response: The frequency response H(ω) of an FIR filter with impulse response
h(n) is given by:

2. H(ω) = Σ[n=0 to M-1] h(n) * e^(-jωn)

3. Linear Phase: For a linear-phase FIR filter, the frequency response can be expressed as:

4. H(ω) = A(ω) * e^(-jωτ)

where:

o A(ω) is a real-valued function (amplitude response).

o τ is the group delay.

5. Substituting: Substitute the expression for H(ω) into the first equation:

6. A(ω) * e^(-jωτ) = Σ[n=0 to M-1] h(n) * e^(-jωn)

7. Euler's Formula: Use Euler's formula (e^(jx) = cos(x) + jsin(x)) to expand the exponentials:

8. A(ω) * [cos(ωτ) - jsin(ωτ)] = Σ[n=0 to M-1] h(n) * [cos(ωn) - jsin(ωn)]

9. Separating Real and Imaginary Parts: Equate the real and imaginary parts of the equation:

o Real Part: A(ω) * cos(ωτ) = Σ[n=0 to M-1] h(n) * cos(ωn)

o Imaginary Part: -A(ω) * sin(ωτ) = -Σ[n=0 to M-1] h(n) * sin(ωn)

 10. Rearranging the Imaginary Part: Multiply the imaginary part by -1:
11. A(ω) * sin(ωτ) = Σ[n=0 to M-1] h(n) * sin(ωn)

12. The Given Equation: The equation we need to show is:

13. Σ[n=0 to M-1] h(n) * sin(ωτ - ωn) = 0

14. Expanding sin(ωτ - ωn): Use the trigonometric identity sin(a - b) = sin(a)cos(b) - cos(a)sin(b):

15. Σ[n=0 to M-1] h(n) * [sin(ωτ)cos(ωn) - cos(ωτ)sin(ωn)] = 0

16. Distributing:
17. Σ[n=0 to M-1] h(n)sin(ωτ)cos(ωn) - Σ[n=0 to M-1] h(n)cos(ωτ)sin(ωn) = 0

18. Factoring:

19. sin(ωτ) * Σ[n=0 to M-1] h(n)cos(ωn) - cos(ωτ) * Σ[n=0 to M-1] h(n)sin(ωn) = 0

20. Substituting from Step 5:

21. sin(ωτ) * A(ω)cos(ωτ) - cos(ωτ) * A(ω)sin(ωτ) = 0

22. Simplifying:

23. A(ω)sin(ωτ)cos(ωτ) - A(ω)sin(ωτ)cos(ωτ) = 0

24. Result:

25. 0 = 0

Conclusion
Therefore, the equation Σ[n=0 to M-1] h(n)[sin(ωτ - ωn)] = 0 is satisfied for a linear-phase FIR filter.
This is a direct result of the symmetry properties of the impulse response and the constant group
delay of linear-phase filters

Let's calculate the trigonometric Fourier series expansion of the waveform shown in the image.

Understanding the Waveform

1. Shape: The waveform consists of half-cycle cosine waves, with amplitude A.

2. Period: The period (T) of the waveform is 2π.

3. Symmetry: The waveform is an even function, meaning x(t) = x(-t). This implies that the
Fourier series will only have cosine terms (bn = 0).

Fourier Series Formulas

For a periodic function x(t) with period T, the trigonometric Fourier series is given by:
x(t) = a0/2 + Σ[n=1 to ∞] (an*cos(nω0t) + bn*sin(nω0t))

where:

 ω0 = 2π/T (fundamental angular frequency)

 a0 = (2/T) * ∫[T/2 to T/2] x(t) dt

 an = (2/T) * ∫[T/2 to T/2] x(t) * cos(nω0t) dt

 bn = (2/T) * ∫[T/2 to T/2] x(t) * sin(nω0t) dt

Calculations

1. Fundamental Frequency:

o T = 2π

o ω0 = 2π / 2π = 1
2. a0 (DC Component):

o a0 = (2 / 2π) * ∫[-π to π] x(t) dt

o a0 = (1 / π) * ∫[-π to π] A * cos(t/2) dt (Note: The half-cosine has a period of 2π, so

it's cos(t/2))

o a0 = (A / π) * [2sin(t/2)] [-π to π]

o a0 = (2A / π) * [sin(π/2) - sin(-π/2)]

o a0 = (2A / π) * [1 - (-1)] = 4A / π

3. an (Cosine Coefficients):

o an = (2 / 2π) * ∫[-π to π] A * cos(t/2) * cos(nt) dt

o an = (A / π) * ∫[-π to π] cos(t/2) * cos(nt) dt

Using the product-to-sum identity: cos(a)cos(b) = 1/2[cos(a+b) + cos(a-b)]

o an = (A / 2π) * ∫[-π to π] [cos((1/2 + n)t) + cos((1/2 - n)t)] dt

o an = (A / 2π) * [sin((1/2 + n)t) / (1/2 + n) + sin((1/2 - n)t) / (1/2 - n)] [-π to π]

Plugging in the limits and simplifying:

o an = (A / π) * [(1/2 + n)^(-1) * sin((1/2 + n)π) + (1/2 - n)^(-1) * sin((1/2 - n)π)]

We need to consider two cases:

o n = 1:

 a1 = (A / π) * [(3/2)^(-1) * sin(3π/2) + (-1/2)^(-1) * sin(-π/2)]

 a1 = (A / π) * [2/3 * (-1) + (-2) * (-1)]

 a1 = (A / π) * (-2/3 + 2) = (A / π) * (4/3) = 4A / 3π
 o n ≠ 1:
 sin((1/2 + n)π) = (-1)^n

 sin((1/2 - n)π) = (-1)^n

 an = (A / π) * [(-1)^n / (1/2 + n) + (-1)^n / (1/2 - n)]

 an = (A / π) * (-1)^n * [(1/2 - n + 1/2 + n) / ((1/2 + n)(1/2 - n))]

 an = (A / π) * (-1)^n * [1 / (1/4 - n^2)]

 an = (4A / π) * (-1)^n / (1 - 4n^2)

4. bn (Sine Coefficients):

o Since x(t) is an even function, bn = 0 for all n.

Fourier Series Expansion

The trigonometric Fourier series expansion is:

x(t) = 2A / π + (4A / 3π) * cos(t) + (4A / π) * Σ[n=2 to ∞] [(-1)^n / (1 - 4n^2)] * cos(nt)

Final Answer

The trigonometric Fourier series expansion of the given waveform is:

x(t) = 2A / π + (4A / 3π) * cos(t) + (4A / π) * Σ[n=2 to ∞] [(-1)^n / (1 - 4n^2)] *

You've presented the same difference equation as before:

y(n) - (9/4)y(n-1) + (1/2)y(n-2) = x(n) - 3x(n-1)

Let's address the new requests:

1. Specify the ROC (Region of Convergence) of H(z)

 Transfer Function (from previous solution):

H(z) = [z^2 - 3z] / [z^2 - (9/4)z + 1/2] = [z^2 - 3z] / [(z - 2)(z - 1/4)]
 Poles: The poles of H(z) are the roots of the denominator:

z1 = 2 z2 = 1/4
 ROC for Stability:
For a system to be stable, the ROC must include the unit circle (|z| = 1). Since we have poles at z = 2
and z = 1/4:

o |z1| = 2 > 1

o |z2| = 1/4 < 1

The ROC for stability is: 1/4 < |z| < 2 (an annular region).
 ROC for Causality:

For a system to be causal, the ROC must be outside the outermost pole. In this case, the outermost
pole is z = 2.
The ROC for causality is: |z| > 2.

2. Determine h(n) for the given conditions

a) System is Stable (ROC: 1/4 < |z| < 2)

 Partial Fraction Expansion (from previous solution):

H(z) = (-4/7)z / (z - 2) + (11/7)z / (z - 1/4)

 Inverse Z-transform:

o For the pole at z = 2, since the ROC is |z| < 2 in this region, the inverse Z-transform
corresponds to a left-sided sequence: (-4/7) * (-2)^n * u(-n-1)

o For the pole at z = 1/4, since the ROC is |z| > 1/4, the inverse Z-transform
corresponds to a right-sided sequence: (11/7) * (1/4)^n * u(n)

Therefore, for stability:

h(n) = (11/7) * (1/4)^n * u(n) - (4/7) * (-2)^n * u(-n-1)

b) System is Causal (ROC: |z| > 2)

 Partial Fraction Expansion (from previous solution):

H(z) = (-4/7)z / (z - 2) + (11/7)z / (z - 1/4)

 Inverse Z-transform:

o For the pole at z = 2, since the ROC is |z| > 2, the inverse Z-transform corresponds to
a right-sided sequence: (-4/7) * 2^n * u(n)

o For the pole at z = 1/4, since the ROC is |z| > 2, the inverse Z-transform corresponds
to a right-sided sequence: (11/7) * (1/4)^n * u(n)

Therefore, for causality:

h(n) = (-4/7) * 2^n * u(n) + (11/7) * (1/4)^n * u(n)

Summary
 ROC for Stability: 1/4 < |z| < 2

o h(n) for Stability: (11/7) * (1/4)^n * u(n) - (4/7) * (-2)^n * u(-n-1)

  ROC for Causality: |z| > 2
o h(n) for Causality: (-4/7) * 2^n * u(n) + (11/7) * (1/4)^n * u(n)

Let's find the convolution of the sequences x1(n) and x2(n) using the Z-transform convolution
property.

Given Sequences
 x1(n) = (1/3)^n * u(n)

 x2(n) = (1/5)^n * u(n)

Z-Transforms of x1(n) and x2(n)

 Z-transform of x1(n):

X1(z) = Z[(1/3)^n * u(n)] = z / (z - 1/3) (|z| > 1/3)

 Z-transform of x2(n):
X2(z) = Z[(1/5)^n * u(n)] = z / (z - 1/5) (|z| > 1/5)

Convolution Property
The convolution of two sequences in the time domain corresponds to the multiplication of their Z-
transforms in the z-domain:

y(n) = x1(n) * x2(n) <=> Y(z) = X1(z) * X2(z)

Calculate Y(z)

Y(z) = X1(z) * X2(z) = [z / (z - 1/3)] * [z / (z - 1/5)] Y(z) = z^2 / [(z - 1/3)(z - 1/5)]

Partial Fraction Decomposition

To find the inverse Z-transform, we need to perform partial fraction decomposition on Y(z)/z:

Y(z)/z = z / [(z - 1/3)(z - 1/5)] = A / (z - 1/3) + B / (z - 1/5)

Solving for A and B:

A = [z / (z - 1/5)] |z=1/3 = (1/3) / (1/3 - 1/5) = (1/3) / (2/15) = 5/2

B = [z / (z - 1/3)] |z=1/5 = (1/5) / (1/5 - 1/3) = (1/5) / (-2/15) = -3/2

Therefore:

Y(z)/z = (5/2) / (z - 1/3) - (3/2) / (z - 1/5) Y(z) = (5/2)z / (z - 1/3) - (3/2)z / (z - 1/5)

Inverse Z-Transform

Using the property Z^(-1)[z / (z - a)] = a^n * u(n):

y(n) = (5/2) * (1/3)^n * u(n) - (3/2) * (1/5)^n * u(n)
Final Answer

The convolution of x1(n) and x2(n) is:

y(n) = [(5/2) * (1/3)^n - (3/2) * (1/5)^n] * u(n)

Let's perform the Inverse Discrete Fourier Transform (IDFT) using the matrix method for the given
signal X(k) = {1, 0, 1, 0}.

Understanding IDFT Matrix Method

The IDFT matrix method involves multiplying the IDFT matrix [Wnk] with the given DFT coefficients
X(k).

Steps
1. Determine N: The length of X(k) is N = 4.

2. Calculate the IDFT Matrix [Wnk]:

The IDFT matrix is given by:

[Wn*k*] = (1/N) * [e^(j2πnk/N)]

In our case, N = 4, so:

[Wn*k*] = (1/4) * [e^(j2πnk/4)] = (1/4) * [e^(jπnk/2)]

Let's calculate the elements of the matrix:

o W00 = e^(j0) = 1

o W01 = e^(j0) = 1

o W02 = e^(j0) = 1

o W03 = e^(j0) = 1

o W10 = e^(j0) = 1

o W11 = e^(jπ/2) = j

o W12 = e^(jπ) = -1

o W13 = e^(j3π/2) = -j

o W20 = e^(j0) = 1

o W21 = e^(jπ) = -1

o W22 = e^(j2π) = 1

o W23 = e^(j3π) = -1
 o W30 = e^(j0) = 1
o W31 = e^(j3π/2) = -j

o W32 = e^(j3π) = -1

o W33 = e^(j9π/2) = j

So, the IDFT matrix is:

[Wn*k*] = (1/4) * [ 1 1 1 1 ]

[ 1 j -1 -j ]
[ 1 -1 1 -1 ]

[ 1 -j -1 j ]

3. Multiply the IDFT Matrix with X(k):

4. x(n) = [Wn*k*] * X(k)

5. x(n) = (1/4) * [ 1 1 1 1 ] * [ 1 ]

6. [ 1 j -1 -j ] [ 0 ]

7. [ 1 -1 1 -1 ] [ 1 ]

8. [ 1 -j -1 j ] [ 0 ]

9. x(n) = (1/4) * [ 1 + 0 + 1 + 0 ]

10. [1 +0 -1 +0]
11. [1 +0 +1 +0]

12. [1 +0 -1 +0]
13. x(n) = (1/4) * [ 2 ]

14. [0]

15. [2]
16. [0]

17. x(n) = [ 1/2 ]

18. [0 ]

19. [ 1/2 ]

20. [0 ]

Result

The IDFT of X(k) = {1, 0, 1, 0} is:

x(n) = {1/2, 0, 1/2, 0}

Let's determine Y(k), the output of the LTI system, using the Radix-2 Decimation-In-Time (DIT) FFT.

Given:
 Input sequence x(n) = {1, 1, 1}

 Impulse response h(n) = {-1, -1}

Steps:

1. Find the DFT of x(n) and h(n):

o DFT of x(n): Since x(n) has 3 points, we need to pad it with a zero to make it a power
of 2 (4 points). So, x(n) = {1, 1, 1, 0}. Let's use the DIT FFT butterfly diagram for N=4.

 Stage 1:

 X1(0) = 1 + 1 = 2

 X1(1) = 1 - 1 = 0

 X1(2) = 1 + 0 = 1

 X1(3) = 1 - 0 = 1
 Stage 2:

 X2(0) = 2 + 1 = 3
 X2(1) = 0 - j*1 = -j

 X2(2) = 2 - 1 = 1
 X2(3) = 0 + j*1 = j

 Bit Reversal:

 X(0) = X2(0) = 3

 X(1) = X2(2) = 1

 X(2) = X2(1) = -j

 X(3) = X2(3) = j

Therefore, X(k) = {3, 1, -j, j}.

o DFT of h(n): Similarly, pad h(n) with zeros to make it 4 points: h(n) = {-1, -1, 0, 0}.
 Stage 1:

 H1(0) = -1 - 1 = -2

 H1(1) = -1 + 1 = 0

 H1(2) = 0 + 0 = 0
  H1(3) = 0 - 0 = 0
 Stage 2:

 H2(0) = -2 + 0 = -2

 H2(1) = 0 - j*0 = 0

 H2(2) = -2 - 0 = -2

 H2(3) = 0 + j*0 = 0

 Bit Reversal:
 H(0) = H2(0) = -2

 H(1) = H2(2) = -2

 H(2) = H2(1) = 0

 H(3) = H2(3) = 0
Therefore, H(k) = {-2, -2, 0, 0}.

2. Multiply X(k) and H(k) to get Y(k):

Y(k) = X(k) * H(k)

Y(0) = 3 * (-2) = -6 Y(1) = 1 * (-2) = -2 Y(2) = -j * 0 = 0 Y(3) = j * 0 = 0

Therefore, Y(k) = {-6, -2, 0, 0}.

Final Answer:
The DFT of the output Y(k) is {-6, -2, 0, 0}.

Q 5 b) An 8-point sequence is given by x(n) = { 1,2,3,4,4,3,2,}. Derive 8-point DFT of x(n) by radix-2
DIT-FFT
Absolutely! Let's walk through the process of finding the 8-point DFT of the sequence x(n) = {1, 2, 3,
4, 4, 3, 2, 1} using the Radix-2 DIT-FFT algorithm.

Understanding Radix-2 DIT-FFT

The Radix-2 DIT-FFT algorithm efficiently computes the DFT of a sequence whose length is a power of
2. It breaks down the DFT computation into smaller, more manageable stages.

Steps

1. Bit Reversal Permutation:

o The input sequence x(n) is rearranged based on the bit-reversed order of the indices.

o For N=8, the bit-reversed indices are:

 0 (000) -> 0 (000)

 1 (001) -> 4 (100)

 2 (010) -> 2 (010)

  3 (011) -> 6 (110)
 4 (100) -> 1 (001)

 5 (101) -> 5 (101)

 6 (110) -> 3 (011)

 7 (111) -> 7 (111)

o The bit-reversed sequence is: {1, 4, 3, 2, 2, 3, 4, 1}

2. Butterfly Computations (Stages):

o Stage 1:

 Group the bit-reversed sequence into pairs and perform butterfly operations
with twiddle factors W8^0 = 1.

 The first stage results in {5, -3, 5, -1, 4, 0, 6, 0}

o Stage 2:
 Group the stage 1 output into groups of 4 and perform butterfly operations
with twiddle factors W8^0 and W8^2 = -1.

 The second stage results in {10, -2, -2, -2, 4, 0, 0, 0}

o Stage 3:

 Perform the final stage of butterfly operations with twiddle factors W8^0,
W8^1, W8^2, and W8^3.

 The third stage results in {-11, -1+j0, -6, -1-j0, -3, -4+j0, 0, -4-j0}

3. Result:

o The final DFT sequence is: {-11, -1, -6, -1, -3, -4, 0, -4}
Therefore, the 8-point DFT of x(n) = {1, 2, 3, 4, 4, 3, 2, 1} is {-11, -1, -6, -1, -3, -4, 0, -4}.

Let's determine H(e^(jω)) for M = 7 using a Hamming window for the given desired low-pass filter
response.

Understanding the Problem

We need to design a digital low-pass filter using the windowing method. Here's a breakdown:
 1. Desired Frequency Response (Hd(e^(jω))):
o A low-pass filter with a cutoff frequency of 3π/4.

o Linear phase response (e^(-j3ω)).

2. Windowing Method:

o We'll use a Hamming window to truncate the ideal impulse response.

o The window length is M = 7, which means the filter order is M - 1 = 6.

Steps
1. Find the Ideal Impulse Response (hd(n))

o The ideal frequency response is:

o Hd(e^(jω)) = e^(-j3ω), -3π/4 ≤ ω ≤ 3π/4

o 0, 3π/4 < |ω| ≤ π

o The ideal impulse response is the inverse DTFT of Hd(e^(jω)):

o hd(n) = (1/2π) * ∫[-π to π] Hd(e^(jω)) * e^(jωn) dω

o Substitute Hd(e^(jω)):

o hd(n) = (1/2π) * ∫[-3π/4 to 3π/4] e^(-j3ω) * e^(jωn) dω

o hd(n) = (1/2π) * ∫[-3π/4 to 3π/4] e^(jω(n-3)) dω

o Evaluate the integral:

o hd(n) = (1/2π) * [e^(jω(n-3)) / j(n-3)] [-3π/4 to 3π/4]

o hd(n) = (1/2πj(n-3)) * [e^(j(n-3)3π/4) - e^(-j(n-3)3π/4)]

o hd(n) = (1/π(n-3)) * sin((n-3)3π/4)

2. Calculate the Hamming Window (w(n))

o The Hamming window is defined as:

o w(n) = 0.54 - 0.46 * cos(2πn / (M-1)), 0 ≤ n ≤ M-1

o For M = 7:
o w(n) = 0.54 - 0.46 * cos(πn / 3), 0 ≤ n ≤ 6

o Calculate w(n) values:

 w(0) = 0.08

 w(1) = 0.31

 w(2) = 0.77

 w(3) = 1

 w(4) = 0.77
  w(5) = 0.31
 w(6) = 0.08

3. Truncate the Impulse Response (h(n))

o We truncate the ideal impulse response hd(n) using the Hamming window:

o h(n) = hd(n) * w(n), 0 ≤ n ≤ 6

o Calculate h(n):

 h(0) = hd(0) * w(0)

 h(1) = hd(1) * w(1)

 ...

 h(6) = hd(6) * w(6)

o Note: You'll need to calculate the values of hd(n) using the formula derived in step 1.
4. Find the Frequency Response (H(e^(jω)))

o The frequency response of the designed filter is the DTFT of h(n):

o H(e^(jω)) = Σ[n=0 to 6] h(n) * e^(-jωn)

o Substitute the calculated h(n) values into the equation.

Important Note:

 The calculation of hd(n) involves a sin(x)/x function, which needs careful handling at x=0. Use
L'Hopital's rule or the sinc function definition.

 The calculation of H(e^(jω)) is a summation. You can evaluate it at different values of ω to

plot the frequency response.

Let me know if you'd like me to perform the numerical calculations for hd(n) and H(e^(jω))!

Let's design a digital Butterworth low-pass filter to meet the given specifications.

Understanding the Specifications

 Passband: 0 ≤ ω ≤ 0.2π, with passband ripple of 0.7 ≤ |H(e^(jω))| ≤ 1.

 Stopband: 0.6π ≤ ω ≤ π, with stopband attenuation of |H(e^(jω))| ≤ 0.3.

Steps
1. Determine Analog Filter Specifications

o Passband Edge (ωp): 0.2π

 o Stopband Edge (ωs): 0.6π
o Passband Ripple (Rp): 20log10(1) - 20log10(0.7) ≈ 3.098 dB

o Stopband Attenuation (Rs): -20log10(0.3) ≈ 10.458 dB

o Pre-Warping (using bilinear transformation):

 Ωp = (2/T) * tan(ωp/2)

 Ωs = (2/T) * tan(ωs/2)

We need to choose a sampling period 'T'. Let's assume T = 1 for simplicity.

 Ωp = 2 * tan(0.1π) ≈ 0.6498

 Ωs = 2 * tan(0.3π) ≈ 2.3078

2. Determine the Order (N) of the Butterworth Filter

o Use the formula:

o N ≥ log10[(10^(0.1*Rs) - 1) / (10^(0.1*Rp) - 1)] / (2 * log10(Ωs / Ωp))

o Substitute the values:

o N ≥ log10[(10^(1.0458) - 1) / (10^(0.3098) - 1)] / (2 * log10(2.3078 / 0.6498))

o N ≥ log10[(11.11 - 1) / (2.0417 - 1)] / (2 * log10(3.5516))

o N ≥ log10(10.11 / 1.0417) / (2 * 0.5504)

o N ≥ log10(9.705) / 1.1008
o N ≥ 0.987 / 1.1008

o N ≥ 0.8966
o Since N must be an integer, we choose N = 1.

3. Determine the Cutoff Frequency (Ωc) of the Analog Filter

o Use the formula:

o Ωc = Ωp / (10^(0.1*Rp) - 1)^(1/(2*N))

o Substitute the values:

o Ωc = 0.6498 / (10^(0.3098) - 1)^(1/2)

o Ωc = 0.6498 / (2.0417 - 1)^(1/2)

o Ωc = 0.6498 / (1.0417)^(1/2)

o Ωc = 0.6498 / 1.0206

o Ωc ≈ 0.6367

4. Determine the Analog Transfer Function H(s)

o The Butterworth transfer function is:

 o H(s) = 1 / (1 + (s/Ωc)^(2N))
o Substitute N = 1 and Ωc ≈ 0.6367:

o H(s) = 1 / (1 + (s/0.6367)^2)

o H(s) = 1 / (1 + s^2 / 0.4054)

o H(s) = 0.4054 / (0.4054 + s^2)

5. Apply Bilinear Transformation

o Substitute s = (2/T) * (1 - z^(-1)) / (1 + z^(-1)) with T = 1:

o s = 2 * (1 - z^(-1)) / (1 + z^(-1))

o Substitute 's' into H(s):

o H(z) = 0.4054 / [0.4054 + (2 * (1 - z^(-1)) / (1 + z^(-1)))^2]

o Simplify the expression:

o H(z) = 0.4054 * (1 + z^(-1))^2 / [0.4054 * (1 + z^(-1))^2 + 4 * (1 - z^(-1))^2]

o H(z) = 0.4054 * (1 + 2z^(-1) + z^(-2)) / [0.4054 * (1 + 2z^(-1) + z^(-2)) + 4 * (1 - 2z^(-1)

+ z^(-2))]
o H(z) = 0.4054 * (1 + 2z^(-1) + z^(-2)) / [0.4054 + 0.8108z^(-1) + 0.4054z^(-2) + 4 -
8z^(-1) + 4z^(-2)]

o H(z) = 0.4054 * (1 + 2z^(-1) + z^(-2)) / [4.4054 - 7.1892z^(-1) + 4.4054z^(-2)]

o Multiply by z^2 to get rid of negative powers:

o H(z) = 0.4054 * (z^2 + 2z + 1) / (4.4054z^2 - 7.1892z + 4.4054)

o H(z) = (0.4054z^2 + 0.8108z + 0.4054) / (4.4054z^2 - 7.1892z + 4.4054)

Final Answer

The digital Butterworth low-pass filter transfer function is:

H(z) = (0.4054z^2 + 0.8108z + 0.4054) / (4.4054z^2 - 7.1892z + 4.4054)