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Covers the principles and techniques of analyzing and processing signals, including spectral analysis, filtering, and modulation, for various applications.
If the sampling frequency is fs = 50 Hz, which among the signals will experience aliasing?
In digital signal processing, the _______converts an analog signal to typically an electrical signal
The Z-transform of convolution is multiplication.
If there are two poles that represent a transfer function, it is expected to have two zeros also.
The approximate mainlobe width of a Bartlett window is:
A positive exponent of z denotes that the shift is
How many gradations can an 8 bit ADC represent?
The Discrete Time Fourier Transform (DTFT) is just DFT with ____.
The difference between the Bartlett and triangular windows is that the Fourier transform of a Bartlett window is negative for n even.
The transfer function of the difference equation given yNo = xNo –x(n-1) – 2y(n-1) – y(n-2) is:
Determine the digital sequence for the analog signals given by
Signal which exhibits symmetry in the vertical axis are referred to as:
When ω = π this corresponds to the____ possible rate of oscillation.
In audio signal processing, a _____acts as the transducer in the system. In communications, an _____converts electromagnetic waves.
_________ algorithm is used to compute the Discrete Fourier Transform coefficients efficiently
In audio signal processing, a microphone acts as the transducer in the system.
Zero stuffing is a method which is classified as
Which of the following signalsis discrete-time, deterministic and odd?
Which of the following is true about Butterworth, Type I and Type II Chebyshev filters?
How many bits are needed to represent 1,000,000?
Multiplication of two sequences in time-domain is _______ in frequency domain
The Z-transform of convolution is a circular convolution.
Find the DTFT of the discrete time signal xNo = [1, 0, -1, 0]
Nyquist theorem states that, for a signal to be properly reconstructed, a signal must be sampled twice the ________of the signal.
Signal which exhibits symmetry in the vertical axis is classified as
According to the French mathematician and physicist, ___any continuous periodic signal is could be represented by sum of sinusoids.
It is evident that production of audio CDsfollows the Nyquist theorem since the sampling frequency used in this is 44.1 kHz
____is a discipline that spans electrical engineering, computing, mathematics and the physical sciences.
_____are often referred to as analog signals and they take on values in continuous interval (a, b), where a can be -∞ and b, ∞.
A causal signal will neglect the values at the negative side.
The exponent of z of an advanced impulse is positive.
Generally, for a finite duration two-sided signal, the region of convergence is
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.
It is a phenomenon which occurs when the signal is sampled below the Nyquist rate
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
Aliasing is a phenomenon which occurs when the sampling frequency is below twice the maximum frequency component of the signal.
In order for us to convert a continuous time signal to discrete time time, _____is performed.
The location of the poles and zeros provide the characterization of the filter's response.
The analysis and synthesis equation for Fourier Series of continuous periodic signals is equivalent to N point DFT and the inverse DFT, respectively
It is a process which converts a continuous time signal to discrete-time form.
Determine the value of M for the downsampling represented below:
The downsampling factor involved of the audio signal from 192kHz to 48kHz is
It states that, for a signal to be properly reconstructed, a signal must be sampled twice the maximum frequency component of the signal.
If the signal has a sampling rate of 48kHz, to produce a 44.1kHz signal, using integer factors, the signal has to be
If the system's outputdepends only on its current input sample and past input samples, then it is referred to as
Interpolation is a method similar to
During upsampling, information is added with the values inserted in between.
Convert the causal system’s transfer functions into difference equation.
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
A train of unit sample sequence which is theoretically infinite is referred to as a sinusoidal sequence.
Describe the magnitude response of the 6thorder Butterworth filter as .
An advanced impulse is placed before the reference 0.
The unit for sampling resolution is
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.
If up sampling a signal involves by inserting non-zero values in between, this multirate DSP is referred to as
The difference between the Bartlett and triangular windows is that the Fourier transform of a triangular windowis always negative.
A property which shows that xNo in time domain is X(k) in frequency domain
Signal which exhibits rotational symmetry with respect to the origin is referred to as even signal.
Which of the following is not expressed in Hertz?
What is the z-transform of the signal xNo = 2n uNo ?
How many zeros are there for the given transfer function
The zero padding extends the non-zero value without changing the value of theentire signal
It is evident that production of audio CDs follows the Nyquist theorem since the sampling frequency used in this is 32 kHz
The ROC of xNo = δ(n+1) - δ(n-1) is:
Digital filters are classified according to
One reasons why downsampling is employed in transmission
The ROC of X(z) = z – z-1 + z -2 is:
The input and output of Discrete Fourier transform is discrete and finite.
What is the transfer function of the LTI causal system consist of two poles and zeros located at origin?
In multirate digital signal processing, the factors must be any positive number.
Sampling period is the fixed interval between two samples in the time domain, and the reciprocal of the sampling period is called
An exponential sequence can be expressed as a geometric series.
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.
DFT property which shows that a signal in time domain and frequency domain is a result of a shifted by N samples.
Changing the sampling rate of an audio signal in time domain also affects the characteristics of the audio in frequency domain.
Find the unit impulse response of yNo= xNo+0.7 x(n-1)
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.
Find the unit impulse response of yNo= -2x(n-5)?
The magnitude response of a rectangular pulse is a sinc function.
An exponential sequence can be expressed as an arithmetic series.
The ROC of xNo = u(n-1) is:
What type of filter is described by following specifications
The values of the new samples when employing linear interpolation is computed by
A____ is used to convert digital signals consisting of 0s and 1s to varying analog signals such as a voltage signal.
What is the sequence representation of the discrete signal described by the functional representation given below?
Find the inverse z
Signals are primarily classified into two: periodic and aperiodic.
Find the DTFT of the signal x[n] = 0.9n u[n]
Anlinear time invariant (LTI) causal discrete time system
____is represented by computers. It is where the analysis and decision takes place. Using computer algorithms, the signals are processed.
A _______ is used to convert digital signals consisting of 0s and 1s to varying analog signals (such as a voltage signal).
__convert ac voltage at one frequency to another ac voltage at another frequency.
Which among the signals is equivalent to u(n-1)?
Find the DFT of xNo = [ -1 1 1 -1]
__________ allows a complex plane representation of a digital signal or the system using poles and zeros.
A property of Z-Transform which involves scaling is referred to as
A filter which has a feedback is considered an FIRfilter.
The type of discrete time signal described by a single impulse is referred to as __________
Compute for the minimum sampling resolution that could be represented by a 4 bit ADC?
It is non-invasive test that record the electrical activity of the heart.
The zeros of the transfer function given by H(z) = z-3 is located at
Generally, for a finite duration anti-causal signal, the region of convergence is
Time reversal property is similar to the
The insertion of zero also called the zero stuffing in between samples is an example of upsampling.
Common among characteristics of both Butterworth and Chebyshev Type II filters are having wide transition bands and flat pass bands.
Upsampling in the time-domain is _________ in frequency domain
What is the resulting sampling rate if the original sampling rate is 6000 Hz, up-sampled by 10 and down-sampled by 3?
How many poles does the transfer functionH(z) = z-2 have?
The passband region of the filter is described by following specifications has a gain of
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____.
For higher-order IIR filters, ______ form can be used for more practical realization.
Half of the sampling rate referred to as the Nyquist limit determines the value of
What is the z-transform of the signal
____________ can be used to applications in communications such as band limiting the signal for transmission
Signal which cannot complete a certain pattern in one cycle is classified as
The frequency range of discrete-time signals is
In practice, audio signals are sampled at 8 bits, 16 bits and 24 bits.
Property of DFT which shows additivity and scaling.
Which of the following is true about the side lobe roll-off rate (dB/decade) of windows?
Which of the following Z-transforms is equivalent to xNo = u(n-1)
A functional representation given by
The value of z is the equivalent to the complex value in Fourier Transform if r = 1.
How many poles and zeros does the transfer function H(z) = z-5 have
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:
What is the discrete signal obtained after sampling x(t) = 2.5 sin 200πt with fs = 250 Hz?
If xNo = [ 1 1 0 0 0.5] for 0 ≤ n ≤ 4, the z- transform of the signal is
The notation uNo refers to a __________
The advantage of a Chebyshev Type I and elliptic filters isthat their roll off is faster but suffers from passband ripple.
The value of yNo in DFT can be determined using Inverse Fourier Transform
In audio, after downsampling, the signal is compressed in time domain but behave as expansion in the frequency domain.
The value of yNo in DFT can be determined using N point DFT
What is the sampling period if the sampling frequency is 10 Hz?
The zero padding extends the non-zero value by changing the value of the entire signal through rounding off
The ROC of
DFT provides a discrete frequency representation of a finite-duration sequence in the frequency domain
Which among the windows have the highest peak side lobe?
___________ 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.
A converter used to change dc voltage into ac voltage.
What should be the sampling frequency for the signal x(t) = 1.5 sin 100πt – 2 sin 50πt?
The z transform of the signal is given by X(z) = , its inverse z is:
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.
The input and output of Discrete Fourier transform is discrete and infinite.
Downsampling in the time-domain is _________ in frequency domain
Given the difference equation
Signal which exhibits rotational symmetry with respect to the origin is referred to as odd signal.
Continuous time signal is represented mathematically by a sequence of numbers x
Find the z transform of
A type of digital filter which is used to eliminate a certain amount of frequency (e.g. 60 Hz in power line)
The difference equation of a digital system is described by
Quantization is a phenomenon which occurs when the sampling frequency is below twice the maximum frequency component of the signal
From the given analog signal:
Generally, for a finite duration causal signal, the region of convergence is
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.
A marginally stable digital filter have pole/s which are located at
Which of the following is described by the notation x (t) = -x(-t) or xNo = -x(-n)?
Which of the following signals is continuous time, odd and periodic?
Find the magnitude response of the transfer function given by
A digital filter is considered stable if the poles lie ___________ unit circle.
To produce a 250Hz signal from 400 Hz, thefactor is
The exponent of z of an advanced impulse is negative.
In practice, audio signals are sampled at 8 bits and below.
Where are the poles of H(z) = 1+ z-2 located?
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?
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.
Unprocessed physical quantities such as the ____ that we hear are in the form of continuous time.
____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.
In multirate digital signal processing, the factors must be an integer.
What is the order of the filter
Classify the signals with the notation given below:
Which of the following in not a discrete time signal?
It is distinguished from other areas in computer science by the unique type of data it uses.
Discrete time signal is represented mathematically by a sequence of numbers x.
Find the DFT of the discrete time signal xNo = [1, 1, 1, 1]
A/an ____________ function, if applied to a signal before DFT reduces the spectral leakage due to abrupt truncation of the data sequence.
Which of the following is true about a random signal?
Zeros are defined as the value/s of z where the ______ will become zero.
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
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
Given a specification for filter requirements, IIR can be implemented with less order than the FIR filters.
In audio, after downsampling, the signal is compressed both in time and frequency domain.
Which of the following signals is continuous time and aperiodic?
A property of Z-Transform which involves ashift in the input will have a corresponding shift in the output.
If the signal needs to be represented in 100 gradations, how many bits are needed?
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.
The filter described by following specifications has a stopband attenuation of
The impulse response of yNo = xNo + 0.5y(n-1) is a/an _________________ function.
To make the signal or system real, the pole/zero components must also be real or complex conjugate pairs.
The equation that shows the relationship of the past output and present and past input samples and the present output sample is called
If the shifted input generate the corresponding shifted output in the same amount of time then the system is
Signal which can be expressed in mathematical form is referred to as discrete time signal.
A continuous time and discrete time signal varies in how they are expressed as a function. The latter uses ________ as its function.
If the signal can be expressed in mathematical form and exhibits a complete pattern in one cycle, the signal is deterministic and periodic.
Classification of signals are often referred to as the analog signal
If the signal has a sampling rate of 192kHz, to produce a 48kHz signal, the signal has to be
This device detects electrical signals from the brain using the 8-16 pair of electrodes attached to the scalp.
______________ by an integer factor of L means inserting L-1 zeros for every sample in the data sequence xNo.
Signal which cannot be expressed in simple mathematical form and are often expressed using probability.
A train of unit sample sequence which is theoretically infinite is referred to as unit step sequence.
To reduce a 192kHz to 44.1 kHz, downsampling may be readily used.
The synthesis and analysis equation for Fourier Series of continuous periodic signals is equivalent to N point DFT and the inverse DFT, respectively
Linearity in DSP systems states that the principle of _____________ exists.
To reduce a 192kHz to 44.1 kHz, resampling must be employed.
Going extremely higher than twice will also reconstruct the signal but is not practical.
The filter described by following specifications has a stopband frequency at
If there are two poles that represent a transfer function, the number of zeros can be 0 or 1.
____corresponds to how many levels or gradations can be made to a waveform.
____ is a discipline that spans electrical engineering, computing, mathematics and the physical sciences.
Another term that refers to the transfer function, H(z) is
Which of the following represents the process of upsampling?
The upsampling and downsampling factors to convert 25 Hz to 60 Hz is
The magnitude frequency response represents the ________ of the digital filter.
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 ____.
Which of the following signals is continuous time, deterministic, aperiodic?
A filter with two poles and 2 zeros inside the unit circle is _________ order filter.
In DFT multiplication in the time domain is multiplication in the frequency domain
The magnitude response of a rectangular pulse is a sine function.
A signal defined as xNo = nan uNo has an ROC at
With zero insertion, no information is added to the signal during upsampling.
The _________ of digital signal are applied to compute the spectrum's amplitude,power, or phase.
The inverse z-transform of X(z) = z – z-1 + z -2 is:
____allows a complex plane representation of the signal or the system.
Which among the following steps are not included in FFT algorithm?
Digital signal samples are represented by their amplitude versus
The transfer function of the LTI causal system given by yNo = xNo + 2x(n-1) + x(n-2) is?
Sampling is the transformation of a collection of data typically into a smaller file size.
If the value inserted in between samples is just a replica value of a neighboring sample, it is not considered upsampling.
When dealing with non-causal system, convolution is practically the same with causal; the impulse will always start at n=0.
If a signal is desired to be filtered in a triangular response, which of the following windows could give the best response?
What is the z-transform of the signal xNo = 0.5 δ(n-3)?
____ is used to convert digital signals consisting of 0s and 1s to varying analog signals (such as a voltage signal).
The analog __________is used before ADC to remove frequency components higher than the Fs/2 to avoid aliasing.
In audio signal processing, a microphone acts as the filter of the system.
Which of the following represents the process of downsampling
What is the impulse response of the function H(z) = 1+ z-2
Find the DTFT of x[n] = 5n u[-n]
The filter described by following specifications has a passband region at
Signal which can be expressed in mathematical form is referred to as deterministic signal.
Poles are defined as the value/s of z where the ______ will become zero.
Which of the following is not a way to represent a discrete-time signal?
What is the functional representation of the discrete signal
An algorithm that computes the Discrete Fourier Transform is called a/an _________
In video signals, if the frame rate of the original signal is 30 frames per second, to convert it to 25 frames per second,
______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.
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.
A sampling technique which is the result of the combination of upsampling and downsampling is referred to as
An odd signal may be expressed in continuous time as x (t) is equal to
Reconstruction requires that the sampling rate should have a __________ value which is twice the maximum frequency component of the signal.
The output of Discrete Time Fourier transform is continuous periodic.
Compression is the transformation of a collection of data typically into a smaller file size.
The frequency ranges of DFT is
Region of convergenceis the set of values of z where the value of X(z) will be_____
From the given analog signal: xa(t) = 4 sin 4000πt + 8 cos 3000πt. Determine the minimum sampling frequency to avoid aliasing.
Express xNo = uNo – δNo – 0.5 δ(n-1) in sequence form.
The value of the second sample after upsampling using linear interpolation would be xNo = [ 0 1 2 3 4] by 3 would be:
Which of the following sampling frequency give the lowest quality audio signal?
Given the two sequences,
Signal which exhibits periodicity or can complete a certain pattern in one cycle.
Convolution in the time domain is equivalent to __________ in frequency domain.
H(z) or the design of the filter is as easy as finding the ratio of the input over the output.
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:
The advantage of a Butterworth and elliptic filters is that their roll off is faster but suffers from passband ripple.
Which of the following factors represents resampling?
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.
The order of the digital filter given by H(z)= 1 + z-2
The value of z is the equivalent to the complex value in Fourier Transform if r = 0.
Unprocessed physical quantities such as the audio signal that we hear are in the form of ___.
To reduce the sampling rate from 96 kHz to 32kHz, the downsampling factor is
While referring to the difference equation, if there is a past and present output, the filter is an FIR filter.
____is a property which refers to the scaling or multiplying by a constant.
Convert the transfer functions of a causal system into difference equation.
Signal which cannot complete a certain pattern in one cycle.
If an audio signal is downsampled, the sound would be
An 8-bit ADC channel accepts analog input ranging from 5 to 5 volts, determine the number of quantization levels
Random signal are expressed using____.
The DTFT of x[n] = 0.2 n u[-n]
Signals are primarily classified into two: continuous time signal and discrete time.
What is natural pacemaker of the body?
An advanced impulse is placed after the reference 0.
In signal processing, all input signal begin with a:
The process involved in converting a continuous time to discrete time signal is referred to as:
What characteristic of a signal is described by the completion of a certain pattern in one cycle?
An audio CD player uses a ____________ and oversampling.
A Z-transform has limits from 0 to positive infinity is called unilateral Z-transform.
The Fourier transform of a Rectangular window is?
The typical unwanted result of upsampling in images is in the form of debris and artifacts
The rectangular window has a value of one over its appropriate length.
The output of Discrete Time Fourier transform is discreteand finite.
If the signal can be expressed in mathematical form and exhibits a complete pattern in one cycle, the signal is odd and periodic.
A____ is used to convert digital signals consisting of 0s and 1s to varying analog signals.
While referring to the difference equation, if there is a past and present output, the filter is an IIR filter.
During downsampling, information is added with the values inserted in between.
An even signal may be expressed by xNo =
From the given analog signal: xa(t) = 4 sin 4000πt + 8 cos 3000πt Determine the frequency components of the signal.
Where are the zeros of H(z) = 1+ z-2 located?
An anti-causal signal will neglect the values at the negative side.
DFT provides a discrete frequency representation of infinite-duration sequence in the frequency domain
Signal input is a real world signal which is in____.
An IIR filter requires more delay impulses as compare with an FIR filter which can utilize the same delay element multiple times.
Which of the following is does not employ downsampling of xNo = [0 1 2 3 4 5 6 7 8 9]?
____are often referred to as analog signals and they take on values in continuous interval (a, b), where a can be -∞ and b, ∞.
Signals which have both time and amplitude are discrete and referred to as:
This procedure uses a special device to detect the sound that is reflected from a beating of the heart.
A filter which has a feedback is considered an IIR filter.
The relationship of the dualities of Fourier series and Transform
The typical unwanted result of downsampling in images is in the form of debris and artifacts
Which of the following windows is best when the desired response requires the least sidelobes?
The insertion of zero also called the zero stuffing in between samples is an example of resampling.
To make the signal or system real, the pole/zero components must also be real or complex valued even without a pair.
It is the process which involves rounding off discrete values from the sampled signal.
The insertion of a replica of the value in between samples is an example of upsampling.
The trade-off of using a higher order Butterworth filter is the
The analysis and decision as to how the signal will be processed happens in the:
A mathematical operation that closely resembles convolution by measuring the degree to which the two signals are similar.
In a low pass filter design, the IIR filter that would give the least transition region but with passband ripple is
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
Identify which of the following signals are periodic?
Signal which cannot be expressed in simple mathematical form example is random noise.
Going extremely higher than twice the maximum frequency componentis the best practice since it is practical.
Find the DFT of the discrete time signal xNo = [j, 1, -j, 1]
A Z-transform has limits from 0 to positive infinity is called a rational Z-transform.
An FIR filter requires more delay impulses as compare with an IIR filter which can utilize the same delay element multiple times.
The ____converts an analog signal to typically an electrical signal.
If the signal xNo = [ 0 1 2 3 4] will use linear interpolation to upsample the output by 2 would be
The Discrete Time Fourier Transform is just DFT with ____.
___are said to have a range of 20 Hz up to 2kHz.
In DFT multiplication in the time domain is circular convolution to the frequency domain
The angular frequency is equal to the frequency multiplied by a factor of 2.
The __________________ of a periodic signal can be used to develop the DFT
The desired filter length of a Hamming window if N is 30 is
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