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Covers the principles and techniques of analyzing and processing signals, including spectral analysis, filtering, and modulation, for various applications.

Find the DTFT of x[n] = 5n u[-n]

**[No Answer]**

An algorithm that computes the Discrete Fourier Transform is called a/an _________

- DTFT
- Transfer Function
- FTF
**Fast Fourier Transform**

Sampling is the transformation of a collection of data typically into a smaller file size.

- True
**False**

The filter described by following specifications has a stopband attenuation of

**[No Answer]**

Given the two sequences,

**[No Answer]**

Which among the signals is equivalent to u(n-1)?

**uNo – δNo**- uNo + δNo
- uNo – δNo -δ (n-1)
- uNo – δNo -δ(n-1)

A digital filter is considered stable if the poles lie ___________ unit circle.

**inside**- not specified
- both inside and outside
- outside

The notation uNo refers to a __________

**Unit step signal**- Sinusoidal signal
- Exponential signal
- Unit sample signal

In multirate digital signal processing, the factors must be an integer.

**True**- False

In DFT multiplication in the time domain is multiplication in the frequency domain

- True
**False**

A continuous time and discrete time signal varies in how they are expressed as a function. The latter uses ________ as its function.

- frequency
- time
- poles
**sample**

The Z-transform of convolution is multiplication.

**True**- False

A causal signal will neglect the values at the negative side.

**True**- False

The value of yNo in DFT can be determined using N point DFT

- True
**False**

In audio signal processing, a microphone acts as the filter of the system.

- True
**False**

How many gradations can an 8 bit ADC represent?

- 260
- 270
**250**- 300

Signal which cannot be expressed in simple mathematical form and are often expressed using probability.

- deterministic signal
- periodic signals
**random signals**- power signals

Which of the following is described by the notation x (t) = -x(-t) or xNo = -x(-n)?

- even signal
- periodic signal
**odd signal**- aperiodic signal

What is the discrete signal obtained after sampling x(t) = 2.5 sin 200πt with fs = 250 Hz?

**[No Answer]**

The DTFT of x[n] = 0.2 n u[-n]

- 0
- converge
**diverge**- does not exist

Which of the following is not a way to represent a discrete-time signal?

- Functional representation
- Graphical
- Sequence
**n with rounded values**

Determine the value of M for the downsampling represented below:

- 1
**2**- 0.5
- 1.5

Half of the sampling rate referred to as the Nyquist limit determines the value of

**folding frequency**- recursion
- quantization
- gradation

If a signal is desired to be filtered in a triangular response, which of the following windows could give the best response?

- Hanning
- Cosine Tapered
**Bartlett**- Blackman

If the signal can be expressed in mathematical form and exhibits a complete pattern in one cycle, the signal is deterministic and periodic.

**True**- False

____allows a complex plane representation of the signal or the system.

- Y-transform
- Zero transform
**Z-transform**- Complex plane

Which of the following windows is best when the desired response requires the least sidelobes?

- Hanning
- Hamming
**Blackman-Harris**- Rectangular

_________ algorithm is used to compute the Discrete Fourier Transform coefficients efficiently

- Z transform
- STFT
**FFT**- Sampling

It is distinguished from other areas in computer science by the unique type of data it uses.

- Electromagnetic wave
- Noise
**Signals**- Transducer

The typical unwanted result of upsampling in images is in the form of debris and artifacts

- True
**False**

The frequency ranges of DFT is

- -n to n
**-π to π**- from 0 to L-1
- -∞ to ∞

The value of yNo in DFT can be determined using Inverse Fourier Transform

**True**- False

____corresponds to how many levels or gradations can be made to a waveform.

- Error signals
**Sampling resolution**- Periodic signal
- Z-transform

Changing the sampling rate of an audio signal in time domain also affects the characteristics of the audio in frequency domain.

**True**- False

__convert ac voltage at one frequency to another ac voltage at another frequency.

- inverter
- prime mover
- generators
**frequency converter**

This procedure uses a special device to detect the sound that is reflected from a beating of the heart.

- Mammography
**Echocardiography**- Fluoroscopy
- Arthrogram

Which of the following signals is continuous time, odd and periodic?

**[No Answer]**

A train of unit sample sequence which is theoretically infinite is referred to as a sinusoidal sequence.

- True
**False**

Changing the sampling rate of an audio signal in time domain does not affect the characteristics of the audio in frequency domain since they have inverse relations.

- True
**False**

The analysis and synthesis equation for Fourier Series of continuous periodic signals is equivalent to N point DFT and the inverse DFT, respectively

**True**- False

Going extremely higher than twice will also reconstruct the signal but is not practical.

**True**- False

Compression is the transformation of a collection of data typically into a smaller file size.

**True**- False

Signal which exhibits rotational symmetry with respect to the origin is referred to as odd signal.

- True
- False

DFT provides a discrete frequency representation of infinite-duration sequence in the frequency domain

- True
- False

Where are the poles of H(z) = 1+ z-2 located?

- ±1
- 1-i , 1+i
- ±i
- origin

According to the French mathematician and physicist, ___any continuous periodic signal is could be represented by sum of sinusoids.

- Hans Christian Orsted
- Michael Faraday
- Georges Lemaitre
**Jean-Baptiste Joseph Fourier**

____ is a discipline that spans electrical engineering, computing, mathematics and the physical sciences.

**Digital Signal Processing**- ADC
- Inverter
- DAC

The root/s of the denominator that will make the transfer function equal to 0 making the transfer function undefined is referred to as the

**Pole/s**- impulse response
- transfer function
- Zero/s

How many poles does the transfer functionH(z) = z-2 have?

- 0
- undefined
- 1
**2**

If xNo = [ 1 1 0 0 0.5] for 0 ≤ n ≤ 4, the z- transform of the signal is

- X(z) = z-1 + 0.5 z-4
- X(z) = 1 + z-2 + 0.5 z-5
- X(z) = z + z-1 + 0.5 z-4
**X(z) = 1 + z-1 + 0.5 z-4**

The order of the digital filter given by H(z)= 1 + z-2

**2**- 1
- not specified
- -2

With zero insertion, no information is added to the signal during upsampling.

**True**- False

The Nyquist theorem states that in order for a signal to be properly reconstructed, the signal must be sampled twice the sum of the frequency components of the signal.

- True
**False**

Upsampling in the time-domain is _________ in frequency domain

**compression**- filtering
- stretching
- downsampling

The filter described by following specifications has a stopband frequency at

**[No Answer]**

If an audio signal is downsampled, the sound would be

- slow and chipmunk-like
- fast with lower pitch
**slow with lower pitch**- fast and chipmunk-like

_____are often referred to as analog signals and they take on values in continuous interval (a, b), where a can be -∞ and b, ∞.

**Continuous time signals**- Fourier Analysis
- Error Signal
- Discrete time signals

Which of the following Z-transforms is equivalent to xNo = u(n-1)

- Question 5
- correct
- Mark 1.00 out of 1.00
- FlaggedFlaggedRemove flag

What characteristic of a signal is described by the completion of a certain pattern in one cycle?

- odd
**periodicity**- aperiodicity
- even

How many bits are needed to represent 1,000,000?

**20**- 22
- 19
- 21

The Discrete Time Fourier Transform (DTFT) is just DFT with ____.

**zero padding**- decreasing value
- increasing value
- equal padding

To produce a 250Hz signal from 400 Hz, thefactor is

- 1.6
**5 then 8**- 0.635
- 8 then 5

Poles are defined as the value/s of z where the ______ will become zero.

**denominator**- difference equation
- transfer function
- numerators

What is the z-transform of the signal

**[No Answer]**

In multirate digital signal processing, the factors must be any positive number.

- True
**False**

The z transform of the signal is given by X(z) = , its inverse z is:

- xNo = 0.3 uNo
- xNo = 0.3n u(-n)
**xNo = 0.3n uNo**

In a low pass filter design, the IIR filter that would give the least transition region but with passband ripple is

- Cheby I
- Cheby II
**Elliptic**- Butterworth

The magnitude response of a rectangular pulse is a sine function.

- True
**False**

A Z-transform has limits from 0 to positive infinity is called a rational Z-transform.

- True
- False

The inverse z-transform of X(z) = z – z-1 + z -2 is:

- xNo = 1 - δ(n-1 ) + δ(n-2 )
- xNo = δ(n-1 ) +δ(n+1 ) + δ(n+2 )
**xNo = δ(n+1 ) - δ(n-1 ) + δ(n-2 )**- xNo = δ(n-1 ) - δ(n+1 ) + δ(n+2 )

The values of the new samples when employing linear interpolation is computed by

**averaging of neighboring samples**- zero padding
- zero insertion
- adding the previous and next value

A positive exponent of z denotes that the shift is

- folded
- flipped
- delayed
**advanced**

Classification of signals are often referred to as the analog signal

- discrete-time signal
**continuous-time signal**- deterministic signal
- periodic signal

Find the DFT of the discrete time signal xNo = [j, 1, -j, 1]

- X(k) = [2 , -2j,2, -2j]
- X(k) = [2j, 2,-2j, 2]
- X(k) = [-2 , 2j,2, 2j]
**X(k) = [2 , 2j,-2, 2j]**

Which of the following represents the process of downsampling

- 192kHz to 36kHz
- 48kHz to 192kHz
- 36kHz to 192kHz
**48kHz to 22kHz**

In order for us to convert a continuous time signal to discrete time time, _____is performed.

- ionization
**sampling**- signaling
- doping

The insertion of zero also called the zero stuffing in between samples is an example of resampling.

- True
**False**

The ROC of X(z) = z – z-1 + z -2 is:

- -∞ < z < ∞
**all except z= 0 and ∞**- 0 ≤ z ≤ ∞
- -∞ ≤ z ≤ ∞

____are signals which are the preprocessed signals which are to be used in digital signal processing. It is represented mathematically by a sequence of numbers x.

- Deterministic signal
- Non-deterministic signal
**Discrete time-signals**- Spectra

An even signal may be expressed by xNo =

- -x(-n)
**x(-n)**- -xNo
- x(-t)

How many poles and zeros does the transfer function H(z) = z-5 have

- 5 zeros and no poles
- no zero and 5 poles
- 1 zero and 5 poles
- 5 zeros and 5 poles

Signals which have both time and amplitude are discrete and referred to as:

- sinusoidal
- continuous-time
**digital signal**- discrete-time

To make the signal or system real, the pole/zero components must also be real or complex conjugate pairs.

**True**- False

____is a discipline that spans electrical engineering, computing, mathematics and the physical sciences.

- ADC
- Digital to analog
- Envelope Detector
**Digital Signal Processing**

The typical unwanted result of downsampling in images is in the form of debris and artifacts

**True**- False

The exponent of z of an advanced impulse is negative.

- True
**False**

The value of z is the equivalent to the complex value in Fourier Transform if r = 1.

**True**- False

If the signal xNo = [ 0 1 2 3 4] will use linear interpolation to upsample the output by 2 would be

**xNo = [ 0 0.5 1 1.5 2 2.5 3 3.5 4 ]**- xNo = [ 1 3 5 7]
- xNo = [ 1 1.5 2 2.5 3 3.5 4 ]
- xNo = [ 0 1 1 2 2 3 3 4 4 ]

__________ allows a complex plane representation of a digital signal or the system using poles and zeros.

**Z-transform**- Fourier Transform
- FFT
- Laplace Transform

Linearity in DSP systems states that the principle of _____________ exists.

- causality
**superposition**- non-causality
- interdependence

The relationship of the dualities of Fourier series and Transform

- Is totally the same
- Differ only in the coefficient of the the corresponding complex exponential
- Differ only in the exponent f the exponent of the corresponding complex exponential
**Differ only inthe sign of the exponent of the corresponding complex exponential**

To reduce a 192kHz to 44.1 kHz, resampling must be employed.

**True**- False

In signal processing, all input signal begin with a:

- filter
- antenna
**transducer**- microphone

A train of unit sample sequence which is theoretically infinite is referred to as unit step sequence.

**True**- False

Property of DFT which shows additivity and scaling.

- Time reversal
**Linearity**- Shifting
- Periodicity

In video signals, if the frame rate of the original signal is 30 frames per second, to convert it to 25 frames per second,

**Up-sample the original signal by 6 then down-sample by 5**- Up-sample the original signal by 5 then down-sample by 6
- Down-sample the original signal by 6 then up-sample by 5
- Up-sample the original signal by 6 then by 5

Which of the following sampling frequency give the lowest quality audio signal?

- 96 Khz
- 48 kHz
**32 kHz**- 44.1 kHz

Anlinear time invariant (LTI) causal discrete time system

- has n>0
**has n≥0**- has n<0
- has n≤0

Find the z transform of

**[No Answer]**

Signal input is a real world signal which is in____.

- y-transform
- complex plane
- digital form
**continuous time form**

Identify which of the following signals are periodic?

- sin 3.5(ω/2*pi )t
**1.5 cos ωt**- cos 3.5 t
- 2 sin 3.5t

Going extremely higher than twice the maximum frequency componentis the best practice since it is practical.

- True
**False**

When ω = π this corresponds to the____ possible rate of oscillation.

- 0%
- Lowest
**Highest**- 100%

For higher-order IIR filters, ______ form can be used for more practical realization.

**cascade**- series
- series-parallel
- parallel

A property of Z-Transform which involves scaling is referred to as

- Multipication
- Shift invariance
**Homogeneity**- Notation

______are signals which are the preprocessed signals which are to be used in digital signal processing. It is represented mathematically by a sequence of numbers x.

- Alising
- Noise
**Discrete time-signals**- Continuous time signals

An exponential sequence can be expressed as an arithmetic series.

- True
- False

An audio CD player uses a ____________ and oversampling.

- 44.1 kHz sampling frequency selection
**multirate signal processing**- single channel
- single rate signal processing

Describe the magnitude response of the 6thorder Butterworth filter as .

- Butterworth Filter has a steep roll off
- Butterworth Filter has minimally flat pass band magnitude response with maximum ripples.
- Butterworth Filter has maximally flat pass band magnitude response with minimum ripples.
- Butterworth Filter has maximally flat pass band magnitude response with no ripples.

The transfer function of the difference equation given yNo = xNo –x(n-1) – 2y(n-1) – y(n-2) is:

- H(z) = 1-z-1/ 2z-1 + z-2
**H(z) = 1-z-1/ 1+ 2z-1 + z-2**- H(z) = 1-z-1/ - 2z-1- z-2
- H(z) = 1-z-1/ 1- 2z-1- z-2

It is a process which converts a continuous time signal to discrete-time form.

- Processing
**Sampling**- Aliasing
- Quantization

Discrete time signal is represented mathematically by a sequence of numbers x.

**True**- False

Reconstruction requires that the sampling rate should have a __________ value which is twice the maximum frequency component of the signal.

**Minimum**- Nearest
- Rounded
- Maximum

To make the signal or system real, the pole/zero components must also be real or complex valued even without a pair.

- True
**False**

The ROC of xNo = u(n-1) is:

**ROC: |z| > 1**- ROC: |z| = 1
- ROC: |z| <1
- ROC: z > 1

The Nyquist theorem states that in order for a signal to be properly reconstructed, the signal must be sampled twice the maximum frequency component of the signal.

**True**- False

Time reversal property is similar to the

- Notation
**Shifting property**- Zero padding
- Periodicity

The analysis and decision as to how the signal will be processed happens in the:

- quantizer
**digital signal processor**- low pass filter
- ADC

The downsampling factor involved of the audio signal from 192kHz to 48kHz is

- 0.25
**4**- 1/4
- 6

Find the inverse z

**[No Answer]**

Signal which exhibits symmetry in the vertical axis are referred to as:

**even**- periodic
- odd
- discrete-time

Find the DFT of the discrete time signal xNo = [1, 1, 1, 1]

- X(k) = [4j, 0, 0, 0 ]
**X(k) = [4, 0, 0, 0 ]**- X(k) = [0, 0, 0, -4j]
- X(k) = [0, 0, 0, 4 ]

During upsampling, information is added with the values inserted in between.

- True
**False**

The type of discrete time signal described by a single impulse is referred to as __________

**Unit sample**- Singularity
- Unit step
- Exponential

What is the sampling period if the sampling frequency is 10 Hz?

- 0.01seconds
**100 ms**- 10 seconds
- 1 second

The cut-off frequency of the ideal filter if the normalized passband and stopband frequency is ωp=0.3π and ωs=0.6π, respectively is

- 0.3π
- 0.6 π
- 0.45π
- 0.9 π

Convert the causal system’s transfer functions into difference equation.

**[No Answer]**

___are said to have a range of 20 Hz up to 2kHz.

- Noise
**Audible signals**- Error signal
- Very high frequency

A____ is used to convert digital signals consisting of 0s and 1s to varying analog signals.

**DAC**- Transducer
- Inverter
- ADC

Downsampling in the time-domain is _________ in frequency domain

- upsampling
- filtering
- compression
**stretching**

What is the transfer function of the LTI causal system consist of two poles and zeros located at origin?

- H(z) = z2
- does not exist
**H(z) = z-2**- H(z) = z2/ z2

The insertion of a replica of the value in between samples is an example of upsampling.

**True**- False

Signal which exhibits periodicity or can complete a certain pattern in one cycle.

- Noise
- Aperiodic signal
- Random Signal
**Periodic signal**

Signals are primarily classified into two: periodic and aperiodic.

- True
**False**

The frequency range of discrete-time signals is

- Repetitive
**Finite**- Complete
- Infinite

____________ can be used to applications in communications such as band limiting the signal for transmission

- Interpolation
**Downsampling**- Upsampling
- Oversampling

DFT provides a discrete frequency representation of a finite-duration sequence in the frequency domain

**True**- False

____is a property which refers to the scaling or multiplying by a constant.

- Scalar
**Homogeneity**- Complex number
- Determinants

Which of the following is not expressed in Hertz?

**Sampling interval**- Sampling frequency
- Nyquist rate
- Maximum frequency component

What is the resulting sampling rate if the original sampling rate is 6000 Hz, up-sampled by 10 and down-sampled by 3?

- 5000 Hz
- 20000 Hz
- 20,000 Hz
- 16000 Hz

Signals which are the preprocessed signals which are to be used in digital signal processing. It is represented mathematically by a sequence of numbers x.

- Cut-off frequency
- Fourier analysis
- Deterministic signal
**Discrete time-signals**

In DFT multiplication in the time domain is circular convolution to the frequency domain

- True
- False

____is represented by computers. It is where the analysis and decision takes place. Using computer algorithms, the signals are processed.

- Analog to digital converter
- Inverter
**Digital signal processor**- Antenna

The value of the second sample after upsampling using linear interpolation would be xNo = [ 0 1 2 3 4] by 3 would be:

- 2
- 3
**0.5**- 1.5

The zero padding extends the non-zero value without changing the value of theentire signal

**True**- False

An exponential sequence can be expressed as a geometric series.

**True**- False

Signal which cannot be expressed in simple mathematical form example is random noise.

- Transducer
**Non-deterministic signal**- Electromagnetic wave
- Deterministic signal

Aliasing is a phenomenon which occurs when the sampling frequency is below twice the maximum frequency component of the signal.

**True**- False

Which of the following in not a discrete time signal?

- xNo = δNo
**x No = sin 3.5πt**- xNo = an uNo
- xNo = cos 2πn

The upsampling and downsampling factors to convert 25 Hz to 60 Hz is

- 5 and 12
- 2.4
- 2 and 4/5
**12 and 5**

Signal which can be expressed in mathematical form example is y = A sin ωtwhere it can be described as a sinusoidal signal with amplitude, A and is a function of time, t.

- Noise
**Deterministic signal**- Continuous time signals
- Discrete time-signals

Signal which can be expressed in mathematical form is referred to as deterministic signal.

**True**- False

Which of the following is does not employ downsampling of xNo = [0 1 2 3 4 5 6 7 8 9]?

- xNo = [0 2 4 6 8 0]
**xNo = [ 0 0.5 1 1.5 2 2.5 3 3.5 4… ]**- xNo = [04 8 …]
- xNo = [0 9]

Sampling period is the fixed interval between two samples in the time domain, and the reciprocal of the sampling period is called

- folding frequency
- transform
- maximum frequency
**sampling rate**

In audio, after downsampling, the signal is compressed in time domain but behave as expansion in the frequency domain.

**True**- False

Find the DFT of xNo = [ -1 1 1 -1]

- X(k) = [2 +2j, 0, 2 - 2j, 0 ]
**X(k) = [0,-2 - 2j, 0,-2 + 2j]**- X(k) = [0, 2 +2j, 0,2 - 2j]
- X(k) = [0,-2 +2j, 0, -2 - 2j]

In practice, audio signals are sampled at 8 bits and below.

- True
**False**

The Discrete Time Fourier Transform is just DFT with ____.

- decreasing value
**zero padding**- equal padding
- increasing value

The transfer function of the LTI causal system given by yNo = xNo + 2x(n-1) + x(n-2) is?

- H(z) = z2 + 2z+ 1
**H(z) = 1 + 2z-1 + z-2**- H(z) = 1 + 2δ(n-1)+ δ(n-2)
- H(z) = δ(n+2)+ 2δ(n+1)+ δNo

What is the z-transform of the signal xNo = 0.5 δ(n-3)?

**[No Answer]**

Which of the following is true about Butterworth, Type I and Type II Chebyshev filters?

**Chebyshev I has more passband ripple than Butterworth filters**- Chebyshev II have steeper roll-off and less stopband ripple than Butterworth filters
- Butterworth filtershave a steeper roll-off and more passband ripple than Chebyshev I
- Butterworth have a steeper roll-off and more stopband ripple than Chebyshev II

If there are two poles that represent a transfer function, the number of zeros can be 0 or 1.

- True
**False**

H(z) or the design of the filter is as easy as finding the ratio of the input over the output.

- True
**False**

What is natural pacemaker of the body?

**Sinus node**- Retina
- Iris
- Aorta

An anti-causal signal will neglect the values at the negative side.

- True
**False**

A/an ____________ function, if applied to a signal before DFT reduces the spectral leakage due to abrupt truncation of the data sequence.

- even
- odd
- sinc
**window**

The insertion of zero also called the zero stuffing in between samples is an example of upsampling.

**True**- False

Method of creating images of the inside of opaque organs in living organisms as well as detecting the amount of bound water in geological structures.

- Fluoroscopy
**MRI (Magnetic Resonance Imaging)**- Echocardiography
- Arthrogram

One reasons why downsampling is employed in transmission

- to remove noise
- to increase bandwidth
**to reduce data during transmission**- to remove debris and artifacts

An advanced impulse is placed after the reference 0.

- True
**False**

The equation that shows the relationship of the past output and present and past input samples and the present output sample is called

**Difference equation**- Transformationequation
- Analysis equation
- Synthesis equation

Suppose for a signal represented by the sequence, xNo = [0 1 2 3 4 5 6 7 8 9], if it is downsampled by 3, the output yNo would be:

- yNo = [0 4 8 0]
**yNo = [0 3 6 9]**- yNo = [ 0 2.5 5.5 7.5]
- yNo = [ 3 6 9 ]

Digital signal samples are represented by their amplitude versus

- time in milliseconds
**sample index**- frequency in Hertz
- frequency and time

A property of Z-Transform which involves ashift in the input will have a corresponding shift in the output.

- Time reversal
**Shift invariance**- Multiplication
- Homogeneity

If the signal has a sampling rate of 48kHz, to produce a 44.1kHz signal, using integer factors, the signal has to be

- Down-sample the original signal by 160 then up-sample by 147
- Up-sampled by a factor of 147, down-sampled by a factor of 160
**Up-sampled by a factor of 160, down-sampled by a factor of 147**- Up-sample the original signal by 160 then by 147

Convert the transfer functions of a causal system into difference equation.

**[No Answer]**

The output of Discrete Time Fourier transform is continuous periodic.

- True
- False

Find the unit impulse response of yNo= -2x(n-5)?

**[No Answer]**

The magnitude response of a rectangular pulse is a sinc function.

**True**- False

It is non-invasive test that record the electrical activity of the heart.

**Electrocardiogram**- EEG
- Biometrics
- Radiography ( X-Ray)

An advanced impulse is placed before the reference 0.

**True**- False

______________ by an integer factor of L means inserting L-1 zeros for every sample in the data sequence xNo.

- Decimation
- Downsampling
**Upsampling**- Zero padding

Multiplication of two sequences in time-domain is _______ in frequency domain

- Multiplication
- Division
- Cross multiplication
**Convolution**

The process involved in converting a continuous time to discrete time signal is referred to as:

- aliasing
**sampling**- extraction
- transform

Classify the signals with the notation given below:

**deterministic, periodic, odd**- continuous-time, aperiodic, odd
- deterministic, aperiodic, even
- continuous-time, periodic, even

This device detects electrical signals from the brain using the 8-16 pair of electrodes attached to the scalp.

- Biometrics
- Radiography
**EEG**- ECG

The rectangular window has a value of one over its appropriate length.

**True**- False

What is the order of the filter

- 6
- 2
- 5
- 4

The main concept behind the ____is that from the electrical signal coming from the transducer, it is converted into a stream of 0s and 1s which can be read by the digital signal processor.

**analog to digital converter**- digital to analog converter
- envelope detector
- filter

The Z-transform of convolution is a circular convolution.

- True
**False**

Which of the following is true about the side lobe roll-off rate (dB/decade) of windows?

**The approximate side lobe roll-off rateof a Blackman and a Hanning window is both 60dB/decade.**- The approximate side lobe roll-off rate of a Blackman and a Hanning window is both 20 dB/decade.
- The approximate side lobe roll-off rate of a Blackman and a Rectangular window is both 20 dB/decade.
- The approximate side lobe roll-off rate of a Blackman and a Rectangular window is both 60 dB/decade.

The ROC of

**[No Answer]**

A filter with two poles and 2 zeros inside the unit circle is _________ order filter.

- 4th
- 1st
- 3rd
**2nd**

Which among the following steps are not included in FFT algorithm?

**Storing the signal xNo into a row vector**- Multiply the resulting array by the phase factors
- Storing the signal xNo into a column vector
- Compute the L-point DFT of each column

From the given analog signal: xa(t) = 4 sin 4000πt + 8 cos 3000πt. Determine the minimum sampling frequency to avoid aliasing.

**4000**- 2000
- 2500
- 3000

Given the difference equation

**[No Answer]**

If the system's outputdepends only on its current input sample and past input samples, then it is referred to as

- time invariant
- non-causal
**causal**- time variant

Nyquist theorem states that, for a signal to be properly reconstructed, a signal must be sampled twice the ________of the signal.

**maximum frequency component**- frequency component
- sum of the frequency components
- minimum frequency component

Continuous time signal is represented mathematically by a sequence of numbers x

- True
**False**

The _________ of digital signal are applied to compute the spectrum's amplitude,power, or phase.

- Fourier integral
**DFT coefficients**- Fourier series
- series equivalent

A property which shows that xNo in time domain is X(k) in frequency domain

- Time reversal
- Periodicity
- Linearity
**Notation**

An odd signal may be expressed in continuous time as x (t) is equal to

- -x(t)
- -x(-n)
**-x(-t)**- x(-t)

The difference between the Bartlett and triangular windows is that the Fourier transform of a triangular windowis always negative.

- True
**False**

When dealing with non-causal system, convolution is practically the same with causal; the impulse will always start at n=0.

- True
- False

Zero stuffing is a method which is classified as

**upsampling**- resampling
- filtering
- downsampling

The output of Discrete Time Fourier transform is discreteand finite.

- True
**False**

It is evident that production of audio CDsfollows the Nyquist theorem since the sampling frequency used in this is 44.1 kHz

**True**- False

The Fourier transform of a Rectangular window is?

**a sinc function**- a triangular function
- a tapered cosine function
- a rectangular function

A marginally stable digital filter have pole/s which are located at

- r = 0
- r > 1
**r = 1**- r < 1

The __________________ of a periodic signal can be used to develop the DFT

- frequency
**Fourier series coefficients**- integral
- summation

Where would the other pole be located if the transfer functions is composed of two zeros in the origin and a pole in -0.7i?

- 1+ 0.7i
- 0.7i
- 0
- 0.7

If the signal needs to be represented in 100 gradations, how many bits are needed?

- 8
**7**- 9
- 6

It is the process which involves rounding off discrete values from the sampled signal.

- Sampling
- Processing
**Quantization**- Aliasing

In practice, audio signals are sampled at 8 bits, 16 bits and 24 bits.

**True**- False

The passband region of the filter is described by following specifications has a gain of

**[No Answer]**

What is the z-transform of the signal xNo = 2n uNo ?

**[No Answer]**

Generally, for a finite duration two-sided signal, the region of convergence is

**Entire z plane except 0 and ∞**- Entire z plane except 0
- Entire z plane except ∞
- Entire z plane except - ∞

While referring to the difference equation, if there is a past and present output, the filter is an FIR filter.

- True
**False**

____ is used to convert digital signals consisting of 0s and 1s to varying analog signals (such as a voltage signal).

- ADC
- FET
- Envelope Detector
**DAC**

If the signal has a sampling rate of 192kHz, to produce a 48kHz signal, the signal has to be

- Up-sample by a factor of 4
- Up-sampled by a factor of 10, down-sampled by a factor of 4
- Up-sampled by a factor of 5, down-sampled by a factor of 2
**Down sampled by a factor of 4**

From the given analog signal: xa(t) = 4 sin 4000πt + 8 cos 3000πt Determine the frequency components of the signal.

- 200 and 150
**2000 and 1500**- 2000 and 1000
- 2500 and 1500

Signal which can be expressed in mathematical form is referred to as discrete time signal.

- True
**False**

The synthesis and analysis equation for Fourier Series of continuous periodic signals is equivalent to N point DFT and the inverse DFT, respectively

**True**- False

What is the functional representation of the discrete signal

**[No Answer]**

A____ is used to convert digital signals consisting of 0s and 1s to varying analog signals such as a voltage signal.

- Notch filter
**DAC**- ADC
- Detector

The ROC of xNo = δ(n+1) - δ(n-1) is:

**All except z= 0 and ∞**- -∞ < z <∞
- -∞ ≤z ≤ ∞
- 0 ≤ z≤∞

A Z-transform has limits from 0 to positive infinity is called unilateral Z-transform.

**True**- False

Find the magnitude response of the transfer function given by

**[No Answer]**

The difference between the Bartlett and triangular windows is that the Fourier transform of a Bartlett window is negative for n even.

**True**- False

Random signal are expressed using____.

- multiplication
- subtraction
**probability**- summation

An IIR filter requires more delay impulses as compare with an FIR filter which can utilize the same delay element multiple times.

- True
**False**

Zeros are defined as the value/s of z where the ______ will become zero.

- root
- denominator
- difference equation
**transfer function**

The zeros of the transfer function given by H(z) = z-3 is located at

- -i and i
**origin**- 1 and -1
- undefined

If there are two poles that represent a transfer function, it is expected to have two zeros also.

**True**- False

A converter used to change dc voltage into ac voltage.

- Motor
- Prime mover
- Generators
**Inverter**

The input and output of Discrete Fourier transform is discrete and finite.

**True**- False

If up sampling a signal involves by inserting non-zero values in between, this multirate DSP is referred to as

- rounding
**interpolation**- insertion
- decimation

___________ by an integer factor of M means taking one sample from the data sequence x No for every M samples and discarding the last M -1 samples.

**Downsampling**- Zero padding
- Upsampling
- Interpolation

A _______ is used to convert digital signals consisting of 0s and 1s to varying analog signals (such as a voltage signal).

- quantizer
- analog-to-digital converter
**digital-to-analog converter**- dequantizer

The advantage of a Butterworth and elliptic filters is that their roll off is faster but suffers from passband ripple.

- True
**False**

How many zeros are there for the given transfer function

**3**- 5
- 6
- 4

An 8-bit ADC channel accepts analog input ranging from 5 to 5 volts, determine the number of quantization levels

- 64
- 255
- 128
**256**

Unprocessed physical quantities such as the audio signal that we hear are in the form of ___.

- frequency domain
**continuous time**- time domain

Generally, for a finite duration causal signal, the region of convergence is

- Entire z plane except 0 and ∞
**Entire z plane except 0**- Entire z plane except -∞
- Entire z plane except ∞

Which of the following signalsis discrete-time, deterministic and odd?

**x[n] = 1.5 sin ωn**- x(t)= 1.5 sin ωt
- x(t)= 1.5 cos ωt
- x[n] = 1.5 cos ωn

The analog __________is used before ADC to remove frequency components higher than the Fs/2 to avoid aliasing.

- dequantizer
- deemphasis
**anti-aliasing lowpass filter**- buffer

A filter which has a feedback is considered an IIR filter.

**True**- False

The transformation of a signal from continuous-time to discrete-time form through sampling doesn’t just involve the conversion of the nature of the signal. This may also allow us to analyze the stability of the system through the use of the____.

- complex plane
- y-transform
- zero transform
**z-transform**

The unit for sampling resolution is

**No unit**- τ
- Volts
- Seconds

What type of filter is described by following specifications

- High pass
- Bandpass
- Bandstop
- Low pass

Signal which can be expressed in mathematical form example is y = A sin ωt where it can be described as a sinusoidal signal with amplitude, A and is a function of time, t.

- Error signals
- Continuous time signals
- Noise
**Deterministic signal**

The input and output of Discrete Fourier transform is discrete and infinite.

- True
**False**

From the given analog signal:

- 2000
- 2500
- 3000
**4000**

To reduce a 192kHz to 44.1 kHz, downsampling may be readily used.

- True
**False**

Another term that refers to the transfer function, H(z) is

- gain
**frequency response**- impulse response
- difference equation

Digital filters are classified according to

- highest exponent of z
- delay elements
**frequency selectivity**- complexity

What should be the sampling frequency for the signal x(t) = 1.5 sin 100πt – 2 sin 50πt?

- 50 Hz
**100 Hz**- 75 Hz
- 150 Hz

Region of convergenceis the set of values of z where the value of X(z) will be_____

- 0
- undetermined
- infinite
**finite**

It states that, for a signal to be properly reconstructed, a signal must be sampled twice the maximum frequency component of the signal.

- Faraday’s Law
**Nyquist theorem**- (ADC) analog to digital converter
- DAC) digital to analog converter

Generally, for a finite duration anti-causal signal, the region of convergence is

- Entire z plane except - ∞
- Entire z plane except 0 and ∞
- Entire z plane except 0
**Entire z plane except ∞**

Interpolation is a method similar to

- filtering
- resampling
- downsampling
**upsampling**

In digital signal processing, the _______converts an analog signal to typically an electrical signal

**transducer**- compressor
- anti-alias filters
- DAC

The value of z is the equivalent to the complex value in Fourier Transform if r = 0.

- True
- False

The trade-off of using a higher order Butterworth filter is the

- less complexity
- flat response
- narrower transition region
**presence of ripples**

If the signal can be expressed in mathematical form and exhibits a complete pattern in one cycle, the signal is odd and periodic.

- True
**False**

Given a specification for filter requirements, IIR can be implemented with less order than the FIR filters.

**True**- False

Express xNo = uNo – δNo – 0.5 δ(n-1) in sequence form.

- xNo = [ 0 0.5 000 …]
- xNo = [ 0 0.5 0 0 0 …]
- xNo = [ 0 0.5 1 1 1 …]

During downsampling, information is added with the values inserted in between.

- True
**False**

If the shifted input generate the corresponding shifted output in the same amount of time then the system is

- causal
- time variant
- non-causal
**time invariant**

Which of the following is true about a random signal?

- It can be predicted with certainty
- Random signal cannot be expressed using probability.
**It is in the form of a non-deterministic signal**- It is periodic in nature.

The angular frequency is equal to the frequency multiplied by a factor of 2.

**True**- False

The location of the poles and zeros is simply a representation of the digital filter and does not provide the characterization of the filter's response.

- True
**False**

A sampling technique which is the result of the combination of upsampling and downsampling is referred to as

- zero padding
- interpolation
- up-down
**resampling**

Signal which exhibits symmetry in the vertical axis is classified as

**even signal**- non-deterministic signal
- odd signal
- deterministic signal

The difference equation of a digital system is described by

**[No Answer]**

In audio, after downsampling, the signal is compressed both in time and frequency domain.

- True
**False**

A filter which has a feedback is considered an FIRfilter.

- True
**False**

Signal which exhibits rotational symmetry with respect to the origin is referred to as even signal.

- True
**False**

Find the DTFT of the discrete time signal xNo = [1, 0, -1, 0]

- X(k) = [0j 0-j]
- X(k) = [j 0 0 -j]
- X(k) = [20 -2 0]
**X(k) = [0 2 0 2]**

The primary advantage of FIR filters over IIR filters is that they typically meet a given set of specifications with a much lower filter order than IIR.

- True
**False**

It is evident that production of audio CDs follows the Nyquist theorem since the sampling frequency used in this is 32 kHz

- True
**False**

A signal is described as an analog signal whose graph is symmetrical to the vertical axis and has complete pattern in one cycle. The signal's classification is completely given as:

**continuous time, even, deterministic and periodic**- continuous time, odd, aperiodic
- continuous time, even,non-deterministic
- continuous time, odd, deterministic and periodic

Quantization is a phenomenon which occurs when the sampling frequency is below twice the maximum frequency component of the signal

- True
**False**

The approximate mainlobe width of a Bartlett window is:

**[No Answer]**

Signal which cannot complete a certain pattern in one cycle is classified as

- deterministic signal
**aperiodic signal**- odd signal
- periodic signal

Which of the following signals is continuous time, deterministic, aperiodic?

- x(t) = cos 0.5 ωt
- x(t) = sin ωt
**x(t) = 3e-2t**- x(t) = sin ωt + cos 0.5 ωt

Which of the following factors represents resampling?

- 3
- 1.5
- 2/3
**4 then 5**

The impulse response of yNo = xNo + 0.5y(n-1) is a/an _________________ function.

- unit sample
- non-existent
- unit step
**exponential**

To reduce the sampling rate from 96 kHz to 32kHz, the downsampling factor is

- 1/3
- 96kHz then 32kHz
**3**- 96/32

All dual relations differ only in the sign of the exponent of the corresponding complex exponential which can be thought of either as _______________ of the spectrum

- multiplication
- padding
**folding**- subtraction

The desired filter length of a Hamming window if N is 30 is

**31**- 30
- 61
- 2*30

The ____converts an analog signal to typically an electrical signal.

- Receiver
- Transmitter
**Transducer**- Filter

Compute for the minimum sampling resolution that could be represented by a 4 bit ADC?

- 1/12
- 0.0625
- 0.05
**1/15**

What is the sequence representation of the discrete signal described by the functional representation given below?

**[No Answer]**

While referring to the difference equation, if there is a past and present output, the filter is an IIR filter.

**True**- False

In audio signal processing, a microphone acts as the transducer in the system.

- True
- False

Find the DTFT of the signal x[n] = 0.9n u[n]

**[No Answer]**

Given the following difference equation with the input-output relationship of a certain initially relaxed system, (all initial conditions are zero). Find the impulse response yNo due to the impulse sequence

**[No Answer]**

An FIR filter requires more delay impulses as compare with an IIR filter which can utilize the same delay element multiple times.

**True**- False

Which of the following signals is continuous time and aperiodic?

**[No Answer]**

The zero padding extends the non-zero value by changing the value of the entire signal through rounding off

- True
**False**

A _________ is the one in which the output yNo at time n depends only on the present input xNo at time n, and its past input sample values

- cascade system
**causal system**- non-causal system
- IIR system

Which among the windows have the highest peak side lobe?

- Hann
**Hamming**- Blackman
- Bartlett

____are often referred to as analog signals and they take on values in continuous interval (a, b), where a can be -∞ and b, ∞.

- Discrete time signals
- Audio signal
**Continuous time signals**- Noise

Which of the following represents the process of upsampling?

- 36kHz to 44.1kHz
**25 frames per second to 50 frames per second**- 44.1kHz to 48kHz
- 24 frames per second to 60 frames per second

It is a phenomenon which occurs when the signal is sampled below the Nyquist rate

- Sampling
**Aliasing**- Quantization
- Processing

Signals are primarily classified into two: continuous time signal and discrete time.

**True**- False

Unprocessed physical quantities such as the ____ that we hear are in the form of continuous time.

- cut-off frequency
**audio signal**- error signal
- ramdom noise

If the value inserted in between samples is just a replica value of a neighboring sample, it is not considered upsampling.

- True
**False**

The magnitude frequency response represents the ________ of the digital filter.

- amplification
- attenuation
**gain**

Signal which cannot complete a certain pattern in one cycle.

- Noise
- Periodic signal
**Aperiodic signal**- Error signal

Common among characteristics of both Butterworth and Chebyshev Type II filters are having wide transition bands and flat pass bands.

**True**- False

Find the unit impulse response of yNo= xNo+0.7 x(n-1)

**[No Answer]**

Convolution in the time domain is equivalent to __________ in frequency domain.

- correlation
- integration
- summation
**multiplication**

A signal defined as xNo = nan uNo has an ROC at

- |z| >1
- |z| <|a|
- |z| = |a|
- |z| > |a|

A functional representation given by

**rectangular pulse**- Fourier transform
- discrete-time sequence
- sinusoidal signal

If the sampling frequency is fs = 50 Hz, which among the signals will experience aliasing?

**x(t) = sin 2π (50)t + cos 2π (25)t**- x(t) = sin 2π (25)t + cos 2π (10)t
- x(t) = sin 2π (20)t
- x(t) = sin 50πt + cos 25πt

Determine the digital sequence for the analog signals given by

**[No Answer]**

What is the impulse response of the function H(z) = 1+ z-2

**hNo = δNo + δ(n-2)**- yNo =δNo + δ(n-2)
- xNo = δNo + δ(n-2)
- hNo = δ(n+2) + δNo

A mathematical operation that closely resembles convolution by measuring the degree to which the two signals are similar.

- summation
- transfer function
- cross multiplication
**correlation**

DSP is a discipline that spans electrical engineering, computing, mathematics and the physical sciences. It is distinguished from other areas in computer science by the unique type of data it uses as ____.

- current
- noise
- alising
**signals**

DFT property which shows that a signal in time domain and frequency domain is a result of a shifted by N samples.

**Periodicity**- Linearity
- Notation
- Time reversal

The location of the poles and zeros provide the characterization of the filter's response.

**True**- False

In audio signal processing, a _____acts as the transducer in the system. In communications, an _____converts electromagnetic waves.

- capacitor, inductor
- resistor, antenna
**microphone, antenna**- antenna,microphone

A type of digital filter which is used to eliminate a certain amount of frequency (e.g. 60 Hz in power line)

- high pass
- bandpass
- low pass
**notch**

The advantage of a Chebyshev Type I and elliptic filters isthat their roll off is faster but suffers from passband ripple.

**True**- False

The exponent of z of an advanced impulse is positive.

**True**- False

Where are the zeros of H(z) = 1+ z-2 located?

- 1-i , 1+i
- ±i
**origin**- ±1

The filter described by following specifications has a passband region at

**[No Answer]**

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