Signal Processing
10+ Signal Processing Interview Questions and Answers

Asked in Texas Instruments

Q. 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.

Asked in Texas Instruments

Q. If two sine waves of different frequencies are added, will the resultant wave be periodic? If so, what is its 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
Signal Processing Interview Questions and Answers for Freshers

Asked in Texas Instruments

Q. If a sine wave is sampled at 1.5 times its original frequency, can the original wave be retained?
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.

Asked in Texas Instruments

Q. Implement the block diagram for the given transfer function H(Z) = a/b.
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

Asked in Texas Instruments

Q. 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

Asked in Texas Instruments

Q. What do 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.
Signal Processing Jobs

Asked in Texas Instruments

Q. What does a Z^(-1) system represent?
Z^(-1) represents a unit delay in discrete-time systems, shifting the signal back by one sample.
Z^(-1) is the z-transform representation of a delay operator.
It shifts the input signal x[n] to x[n-1], introducing a one-sample delay.
In digital filters, Z^(-1) is used to implement feedback and feedforward paths.
Example: If x[n] = {1, 2, 3}, then Z^(-1)x[n] = {0, 1, 2}.

Asked in Texas Instruments

Q. 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
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Asked in Texas Instruments

Q. 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

Asked in Texas Instruments

Q. 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

Asked in Texas Instruments

Q. 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

Asked in Texas Instruments

Q. Given a sequence x[n], 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

Asked in Texas Instruments

Q. 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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