Texas Instruments
10+ Uttarakhand Power Corporation Interview Questions and Answers
Q1. There is a gun in which 2 consecutive slots of the 6 slots are filled. One of the 6 slots is chosen at random and fired at you. It misses. Choose an option between taking the next slot after the one that missed...
read moreIt is better to take another random shot.
The probability of hitting the target is higher when taking another random shot.
Taking the next slot after the one that missed you does not increase the chances of hitting the target.
Each slot has an equal probability of being chosen, so the odds are the same for both options.
Q2. If two sine waves of different frequency are added will resultant wave be periodic. If so what is period
Yes, the resultant wave will be periodic with a period equal to the least common multiple of the two frequencies.
The period of the resultant wave is determined by the least common multiple of the two frequencies.
If the frequencies are incommensurable, the resultant wave will not be periodic.
If the frequencies are harmonically related, the resultant wave will have a period equal to the fundamental period of the lower frequency.
The amplitude and phase of the resultant wave will...read more
Q3. If a sine wave is sampled at 1.5 times its original frequency, can the original wave be retained? (Ans: Yes)
Sampling a sine wave at 1.5 times its original frequency retains the original wave.
The original wave can be reconstructed using interpolation techniques.
The Nyquist-Shannon sampling theorem states that a signal can be perfectly reconstructed if it is sampled at twice its highest frequency component.
Sampling at 1.5 times the original frequency satisfies the Nyquist-Shannon sampling theorem.
This technique is used in digital audio processing.
Q4. A transfer function H(Z) in the form of a/b was given. I was asked to implement the block diagram for the given transfer function.
The question asks to implement a block diagram for a given transfer function in the form of a/b.
Identify the numerator and denominator polynomials of the transfer function
Draw blocks for each polynomial term, representing multiplication and addition operations
Connect the blocks according to the transfer function equation
Include any necessary delays or feedback loops
Label the inputs and outputs of the block diagram
Q5. State the sampling Theorem? Is it better to have a the sampling frequency slightly more than twice the bandwidth? Why?
The sampling theorem states that a signal must be sampled at a rate at least twice its bandwidth to avoid aliasing.
The sampling theorem is also known as the Nyquist-Shannon sampling theorem.
Sampling at a rate slightly more than twice the bandwidth ensures that all the information in the signal is captured without distortion.
If the sampling frequency is too low, aliasing can occur, where high-frequency components are incorrectly represented as lower frequencies.
For example, if...read more
Q6. What does odd and even harmonics of fourier series signify
Odd harmonics represent asymmetry in a signal while even harmonics represent symmetry.
Odd harmonics are multiples of the fundamental frequency and have a phase shift of 90 degrees.
Even harmonics are also multiples of the fundamental frequency but have a phase shift of 0 degrees.
Odd harmonics represent the asymmetry in a signal, while even harmonics represent the symmetry.
For example, a square wave has odd harmonics only, while a triangle wave has both odd and even harmonics.
Q7. What is the difference between DTFT and DFT?
DTFT is a continuous function that represents the frequency content of a discrete-time signal, while DFT is a discrete function that represents the frequency content of a finite-length sequence.
DTFT is defined for both finite and infinite-length signals, while DFT is only defined for finite-length sequences.
DTFT is a continuous function in frequency domain, while DFT is a discrete function in frequency domain.
DTFT provides a complete representation of the frequency content of...read more
Q8. Explain Fourier Series based on your understanding
Fourier series is a mathematical representation of a periodic function as a sum of sine and cosine functions.
Fourier series is used to analyze and synthesize periodic signals.
It decomposes a periodic function into a sum of sine and cosine functions with different frequencies and amplitudes.
The Fourier coefficients represent the amplitude and phase of each frequency component.
The Fourier series can be used to approximate non-periodic functions by extending the function periodi...read more
Q9. H(Z)=1/(1-Z^(-1)). Is the system with the given transfer function stable?
No
The system is not stable because the transfer function has a pole at z = 1
A stable system should have all poles inside the unit circle in the z-plane
In this case, the pole at z = 1 lies on the unit circle, making the system marginally stable
Q10. What are IIR and FIR filters?
IIR and FIR filters are two types of digital filters used in signal processing.
IIR (Infinite Impulse Response) filters use feedback to create a response to an input signal.
FIR (Finite Impulse Response) filters only use feedforward and have a finite duration of response.
IIR filters can be implemented with fewer coefficients but may be less stable than FIR filters.
FIR filters have linear phase response and are generally more stable than IIR filters.
Example of IIR filter: Butter...read more
Q11. A sequence x[n] was given. Find y[n] if H(Z)=1-Z^(-4)
The output sequence y[n] can be obtained by convolving the input sequence x[n] with the impulse response h[n] = [1, 0, 0, 0, -1].
The given transfer function H(Z) represents a discrete-time system with a finite impulse response (FIR) filter.
To find y[n], we need to convolve x[n] with the impulse response h[n] = [1, 0, 0, 0, -1].
Convolution can be performed by sliding the impulse response over the input sequence and summing the products of corresponding samples.
The resulting se...read more
Q12. Explain aliasing, nyquist sampling theorem
Aliasing occurs when a signal is sampled at a rate lower than the Nyquist rate, resulting in distorted or incorrect signal representation.
Nyquist sampling theorem states that a signal must be sampled at a rate at least twice its highest frequency component to avoid aliasing.
Aliasing can be avoided by using a low-pass filter to remove high-frequency components before sampling.
Examples of aliasing include the wagon-wheel effect in movies and the distorted sound of a guitar stri...read more
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