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

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

**True**- False

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

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

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

The Z-transform of convolution is a circular convolution.

- True
**False**

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

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

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

An exponential sequence can be expressed as an arithmetic series.

- True
- False

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

**True**- False

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

- True
**False**

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

- 0
- undetermined
- infinite
**finite**

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

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

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

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

- Sampling
**Aliasing**- Quantization
- Processing

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

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

- 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.

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

**True**- False

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

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

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

**[No Answer]**

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

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

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

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**

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

____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

How many gradations can an 8 bit ADC represent?

- 260
- 270
**250**- 300

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

- True
**False**

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

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

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

- True
**False**

An advanced impulse is placed after the reference 0.

- True
**False**

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

- True
**False**

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

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

**20**- 22
- 19
- 21

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

- 1-i , 1+i
- ±i
**origin**- ±1

A converter used to change dc voltage into ac voltage.

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

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

- True
**False**

Interpolation is a method similar to

- filtering
- resampling
- downsampling
**upsampling**

Property of DFT which shows additivity and scaling.

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

A functional representation given by

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

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 ]

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

**True**- False

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

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

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

**True**- False

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

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

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

**True**- False

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

**DAC**- Transducer
- Inverter
- ADC

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

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

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

**True**- False

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

- 8
**7**- 9
- 6

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

Which among the windows have the highest peak side lobe?

- Hann
**Hamming**- Blackman
- Bartlett

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

**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

The Z-transform of convolution is multiplication.

**True**- False

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

- True
**False**

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

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

Anlinear time invariant (LTI) causal discrete time system

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

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

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

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

- 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 )

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

- Mammography
**Echocardiography**- Fluoroscopy
- Arthrogram

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**

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

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

**True**- False

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

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

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 - ∞

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

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

The unit for sampling resolution is

**No unit**- τ
- Volts
- Seconds

_____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

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

**True**- False

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

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

What is natural pacemaker of the body?

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

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

**True**- False

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

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

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

What is the functional representation of the discrete signal

**[No Answer]**

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**

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**

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 …]

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

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.

Signal which exhibits symmetry in the vertical axis is classified as

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

Signals are primarily classified into two: periodic and aperiodic.

- True
**False**

Downsampling in the time-domain is _________ in frequency domain

- upsampling
- filtering
- compression
**stretching**

Determine the value of M for the downsampling represented below:

- 1
**2**- 0.5
- 1.5

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

- True
**False**

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

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

The ____converts an analog signal to typically an electrical signal.

- Receiver
- Transmitter
**Transducer**- Filter

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

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

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

An audio CD player uses a ____________ and oversampling.

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

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

- True
**False**

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

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

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**

Find the z transform of

**[No Answer]**

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

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

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

**[No Answer]**

Which of the following represents the process of downsampling

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

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

- True
**False**

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

**True**- False

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

**[No Answer]**

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

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

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

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

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

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

- 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]

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**

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**

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

The exponent of z of an advanced impulse is positive.

**True**- False

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

- ±1
- 1-i , 1+i
- ±i
- origin

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

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

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

- True
**False**

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

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

The filter described by following specifications has a stopband frequency at

**[No Answer]**

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

- True
**False**

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

**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

An even signal may be expressed by xNo =

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

An exponential sequence can be expressed as a geometric series.

**True**- False

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

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

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

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

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**

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

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

The Discrete Time Fourier Transform is just DFT with ____.

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

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

- Processing
**Sampling**- Aliasing
- Quantization

Signal input is a real world signal which is in____.

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

The filter described by following specifications has a passband region at

**[No Answer]**

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

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

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

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

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

A positive exponent of z denotes that the shift is

- folded
- flipped
- delayed
**advanced**

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

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

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

- Interpolation
**Downsampling**- Upsampling
- Oversampling

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

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

**True**- False

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

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

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

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

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

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

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

____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

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 ]

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

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

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

Zero stuffing is a method which is classified as

**upsampling**- resampling
- filtering
- downsampling

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

- True
**False**

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

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

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

**[No Answer]**

___________ 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

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**

Which of the following signals is continuous time and aperiodic?

**[No Answer]**

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

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

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

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

Which of the following is not expressed in Hertz?

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

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

**True**- False

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

- True
**False**

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

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

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

- rounding
**interpolation**- insertion
- decimation

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**

Find the magnitude response of the transfer function given by

**[No Answer]**

Which of the following in not a discrete time signal?

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

Given the difference equation

**[No Answer]**

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

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

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

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

Given the two sequences,

**[No Answer]**

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

- True
**False**

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

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

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

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

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

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

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

- True
**False**

What is the order of the filter

- 6
- 2
- 5
- 4

____ 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**

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]**

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

**True**- False

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

- True
- False

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

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

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

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]**

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

**True**- False

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

**True**- False

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

- True
- False

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

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

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

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

**True**- False

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

**True**- False

Classify the signals with the notation given below:

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

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

**True**- False

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

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

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

**[No Answer]**

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

- True
**False**

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

**True**- False

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

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

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

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 ∞

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

- True
**False**

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

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

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

- True
- False

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

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

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

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

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]

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

- 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**

The ROC of xNo = δ(n+1) - δ(n-1) is:

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

The approximate mainlobe width of a Bartlett window is:

**[No Answer]**

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

- True
**False**

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

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

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

**True**- False

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

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

- True
**False**

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

- True
**False**

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

**True**- False

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

- True
- False

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

**True**- False

The notation uNo refers to a __________

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

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**

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

____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

Find the inverse z

**[No Answer]**

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**

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.

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

- correlation
- integration
- summation
**multiplication**

From the given analog signal:

- 2000
- 2500
- 3000
**4000**

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 ∞**

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

**[No Answer]**

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

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

- amplification
- attenuation
**gain**

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

- 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]**

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

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

Upsampling in the time-domain is _________ in frequency domain

**compression**- filtering
- stretching
- downsampling

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

**[No Answer]**

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

**[No Answer]**

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

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

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

- True
- False

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

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

**True**- False

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

- Biometrics
- Radiography
**EEG**- ECG

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**

Digital filters are classified according to

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

The output of Discrete Time Fourier transform is continuous periodic.

- True
- False

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

- True
**False**

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

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

A filter which has a feedback is considered an FIRfilter.

- True
**False**

Classification of signals are often referred to as the analog signal

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

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

**True**- False

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

One reasons why downsampling is employed in transmission

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

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

- True
**False**

An advanced impulse is placed before the reference 0.

**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 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

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**

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

- 64
- 255
- 128
**256**

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

**True**- False

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

- True
**False**

Signal which cannot complete a certain pattern in one cycle.

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

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

**True**- False

What type of filter is described by following specifications

- High pass
- Bandpass
- Bandstop
- Low pass

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

The frequency range of discrete-time signals is

- Repetitive
**Finite**- Complete
- Infinite

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

- True
**False**

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

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

Digital signal samples are represented by their amplitude versus

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

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

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

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

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

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

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

**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**

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

- True
**False**

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

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

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

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**

How many zeros are there for the given transfer function

**3**- 5
- 6
- 4

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

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

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

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

- 0
- undefined
- 1
**2**

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

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

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

**True**- False

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

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

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

- ionization
**sampling**- signaling
- doping

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

- Sampling
- Processing
**Quantization**- Aliasing

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

- True
**False**

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

**[No Answer]**

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

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

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 ]

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

**True**- False

Which of the following factors represents resampling?

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

The ROC of

**[No Answer]**

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

- aliasing
**sampling**- extraction
- transform

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

**True**- False

Determine the digital sequence for the analog signals given by

**[No Answer]**

In signal processing, all input signal begin with a:

- filter
- antenna
**transducer**- microphone

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

The difference equation of a digital system is described by

**[No Answer]**

The ROC of X(z) = z – z-1 + z -2 is:

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

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

**True**- False

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

**[No Answer]**

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

**[No Answer]**

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

**[No Answer]**

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

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**

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

- odd
**periodicity**- aperiodicity
- even

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

**True**- False

The exponent of z of an advanced impulse is negative.

- True
**False**

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

**True**- False

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

**True**- False

The filter described by following specifications has a stopband attenuation of

**[No Answer]**

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

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

What is the z-transform of the signal

**[No Answer]**

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

**Minimum**- Nearest
- Rounded
- Maximum

The output of Discrete Time Fourier transform is discreteand finite.

- True
**False**

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**

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

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

______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

Time reversal property is similar to the

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

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

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

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

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

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

The frequency ranges of DFT is

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

The Fourier transform of a Rectangular window is?

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

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