Ctrl + F is the shortcut in your browser or operating system that allows you to find words or questions quickly.
Ctrl + Tab to move to the next tab to the right and Ctrl + Shift + Tab to move to the next tab to the left.
On a phone or tablet, tap the menu icon in the upper-right corner of the window; Select "Find in Page" to search a question.
Share UsSharing is Caring
It's the biggest motivation to help us to make the site better by sharing this to your friends or classmates.
Covers the principles and techniques of analyzing and processing signals, including spectral analysis, filtering, and modulation, for various applications.
An algorithm that computes the Discrete Fourier Transform is called a/an _________
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:
A property of Z-Transform which involves ashift in the input will have a corresponding shift in the output.
Signal which exhibits symmetry in the vertical axis are referred to as:
What is the order of the filter
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.
What is the sampling period if the sampling frequency is 10 Hz?
In order for us to convert a continuous time signal to discrete time time, _____is performed.
This procedure uses a special device to detect the sound that is reflected from a beating of the heart.
Poles are defined as the value/s of z where the ______ will become zero.
Find the DTFT of x[n] = 5n u[-n]
Signal which exhibits rotational symmetry with respect to the origin is referred to as even signal.
Continuous time signal is represented mathematically by a sequence of numbers x
Which of the following signals is continuous time, odd and periodic?
Anlinear time invariant (LTI) causal discrete time system
The analysis and decision as to how the signal will be processed happens in the:
An even signal may be expressed by xNo =
In multirate digital signal processing, the factors must be an integer.
A train of unit sample sequence which is theoretically infinite is referred to as unit step sequence.
Changing the sampling rate of an audio signal in time domain also affects the characteristics of the audio in frequency domain.
The advantage of a Chebyshev Type I and elliptic filters isthat their roll off is faster but suffers from passband ripple.
Signal which can be expressed in mathematical form is referred to as deterministic signal.
Sampling period is the fixed interval between two samples in the time domain, and the reciprocal of the sampling period is called
In digital signal processing, the _______converts an analog signal to typically an electrical signal
In a low pass filter design, the IIR filter that would give the least transition region but with passband ripple is
Which among the following steps are not included in FFT algorithm?
Going extremely higher than twice the maximum frequency componentis the best practice since it is practical.
Zero stuffing is a method which is classified as
A property of Z-Transform which involves scaling is referred to as
Which of the following is not expressed in Hertz?
Given the difference equation
The frequency ranges of DFT is
The exponent of z of an advanced impulse is positive.
Reconstruction requires that the sampling rate should have a __________ value which is twice the maximum frequency component of the signal.
Signals which have both time and amplitude are discrete and referred to as:
The Z-transform of convolution is a circular convolution.
From the given analog signal:
___are said to have a range of 20 Hz up to 2kHz.
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.
What is the z-transform of the signal xNo = 2n uNo ?
How many poles and zeros does the transfer function H(z) = z-5 have
Zeros are defined as the value/s of z where the ______ will become zero.
Unprocessed physical quantities such as the audio signal that we hear are in the form of ___.
Which among the windows have the highest peak side lobe?
____is a discipline that spans electrical engineering, computing, mathematics and the physical sciences.
What is the transfer function of the LTI causal system consist of two poles and zeros located at origin?
Nyquist theorem states that, for a signal to be properly reconstructed, a signal must be sampled twice the ________of the signal.
A continuous time and discrete time signal varies in how they are expressed as a function. The latter uses ________ as its function.
A train of unit sample sequence which is theoretically infinite is referred to as a sinusoidal sequence.
Determine the digital sequence for the analog signals given by
How many zeros are there for the given transfer function
A functional representation given by
To make the signal or system real, the pole/zero components must also be real or complex valued even without a pair.
Classification of signals are often referred to as the analog signal
Find the unit impulse response of yNo= -2x(n-5)?
Determine the value of M for the downsampling represented below:
How many poles does the transfer functionH(z) = z-2 have?
____is represented by computers. It is where the analysis and decision takes place. Using computer algorithms, the signals are processed.
Digital filters are classified according to
The filter described by following specifications has a stopband frequency at
Signal which exhibits rotational symmetry with respect to the origin is referred to as odd signal.
One reasons why downsampling is employed in transmission
The _________ of digital signal are applied to compute the spectrum's amplitude,power, or phase.
The difference between the Bartlett and triangular windows is that the Fourier transform of a triangular windowis always negative.
A sampling technique which is the result of the combination of upsampling and downsampling is referred to as
Which of the following signalsis discrete-time, deterministic and odd?
If xNo = [ 1 1 0 0 0.5] for 0 ≤ n ≤ 4, the z- transform of the signal is
DFT property which shows that a signal in time domain and frequency domain is a result of a shifted by N samples.
What is the impulse response of the function H(z) = 1+ z-2
The Discrete Time Fourier Transform (DTFT) is just DFT with ____.
Generally, for a finite duration causal signal, the region of convergence is
A signal defined as xNo = nan uNo has an ROC at
An audio CD player uses a ____________ and oversampling.
____are often referred to as analog signals and they take on values in continuous interval (a, b), where a can be -∞ and b, ∞.
Which of the following signals is continuous time and aperiodic?
If the signal needs to be represented in 100 gradations, how many bits are needed?
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
If a signal is desired to be filtered in a triangular response, which of the following windows could give the best response?
In audio, after downsampling, the signal is compressed in time domain but behave as expansion in the frequency domain.
Which of the following windows is best when the desired response requires the least sidelobes?
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 often referred to as analog signals and they take on values in continuous interval (a, b), where a can be -∞ and b, ∞.
Which of the following is not a way to represent a discrete-time signal?
Quantization is a phenomenon which occurs when the sampling frequency is below twice the maximum frequency component of the signal
It is evident that production of audio CDsfollows the Nyquist theorem since the sampling frequency used in this is 44.1 kHz
Where are the poles of H(z) = 1+ z-2 located?
_________ algorithm is used to compute the Discrete Fourier Transform coefficients efficiently
The process involved in converting a continuous time to discrete time signal is referred to as:
An exponential sequence can be expressed as an arithmetic series.
The value of yNo in DFT can be determined using N point DFT
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____.
Given the two sequences,
To produce a 250Hz signal from 400 Hz, thefactor is
______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.
If the system's outputdepends only on its current input sample and past input samples, then it is referred to as
The zeros of the transfer function given by H(z) = z-3 is located at
The rectangular window has a value of one over its appropriate length.
The insertion of zero also called the zero stuffing in between samples is an example of upsampling.
The ROC of xNo = u(n-1) is:
Which of the following is described by the notation x (t) = -x(-t) or xNo = -x(-n)?
The Z-transform of convolution is multiplication.
Another term that refers to the transfer function, H(z) is
If the signal can be expressed in mathematical form and exhibits a complete pattern in one cycle, the signal is deterministic and periodic.
The frequency range of discrete-time signals is
Signal which cannot complete a certain pattern in one cycle.
What should be the sampling frequency for the signal x(t) = 1.5 sin 100πt – 2 sin 50πt?
Find the magnitude response of the transfer function given by
Express xNo = uNo – δNo – 0.5 δ(n-1) in sequence form.
What is natural pacemaker of the body?
This device detects electrical signals from the brain using the 8-16 pair of electrodes attached to the scalp.
A____ is used to convert digital signals consisting of 0s and 1s to varying analog signals.
A type of digital filter which is used to eliminate a certain amount of frequency (e.g. 60 Hz in power line)
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.
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
____corresponds to how many levels or gradations can be made to a waveform.
The impulse response of yNo = xNo + 0.5y(n-1) is a/an _________________ function.
Which of the following is does not employ downsampling of xNo = [0 1 2 3 4 5 6 7 8 9]?
The desired filter length of a Hamming window if N is 30 is
The DTFT of x[n] = 0.2 n u[-n]
The Discrete Time Fourier Transform is just DFT with ____.
Generally, for a finite duration two-sided signal, the region of convergence is
Which of the following is true about Butterworth, Type I and Type II Chebyshev filters?
To reduce the sampling rate from 96 kHz to 32kHz, the downsampling factor is
Signal which can be expressed in mathematical form is referred to as discrete time signal.
Region of convergenceis the set of values of z where the value of X(z) will be_____
If the sampling frequency is fs = 50 Hz, which among the signals will experience aliasing?
An FIR filter requires more delay impulses as compare with an IIR filter which can utilize the same delay element multiple times.
The unit for sampling resolution is
During upsampling, information is added with the values inserted in between.
A filter which has a feedback is considered an FIRfilter.
If there are two poles that represent a transfer function, it is expected to have two zeros also.
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 ____.
Identify which of the following signals are periodic?
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
Find the DFT of the discrete time signal xNo = [1, 1, 1, 1]
The output of Discrete Time Fourier transform is continuous periodic.
From the given analog signal: xa(t) = 4 sin 4000πt + 8 cos 3000πt Determine the frequency components of the signal.
Which of the following factors represents resampling?
Going extremely higher than twice will also reconstruct the signal but is not practical.
If the shifted input generate the corresponding shifted output in the same amount of time then the system is
Convert the causal system’s transfer functions into difference equation.
A Z-transform has limits from 0 to positive infinity is called a rational Z-transform.
If the signal can be expressed in mathematical form and exhibits a complete pattern in one cycle, the signal is odd and periodic.
The ROC of X(z) = z – z-1 + z -2 is:
Signals are primarily classified into two: periodic and aperiodic.
The insertion of zero also called the zero stuffing in between samples is an example of resampling.
DFT provides a discrete frequency representation of infinite-duration sequence in the frequency domain
Upsampling in the time-domain is _________ in frequency domain
Generally, for a finite duration anti-causal signal, the region of convergence is
In DFT multiplication in the time domain is circular convolution to the frequency domain
With zero insertion, no information is added to the signal during upsampling.
What is the discrete signal obtained after sampling x(t) = 2.5 sin 200πt with fs = 250 Hz?
If an audio signal is downsampled, the sound would be
To make the signal or system real, the pole/zero components must also be real or complex conjugate pairs.
The analog __________is used before ADC to remove frequency components higher than the Fs/2 to avoid aliasing.
Half of the sampling rate referred to as the Nyquist limit determines the value of
A filter which has a feedback is considered an IIR filter.
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.
If the signal has a sampling rate of 192kHz, to produce a 48kHz signal, the signal has to be
The upsampling and downsampling factors to convert 25 Hz to 60 Hz is
The downsampling factor involved of the audio signal from 192kHz to 48kHz is
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
Which of the following sampling frequency give the lowest quality audio signal?
The input and output of Discrete Fourier transform is discrete and finite.
The z transform of the signal is given by X(z) = , its inverse z is:
An advanced impulse is placed after the reference 0.
The __________________ of a periodic signal can be used to develop the DFT
Find the z transform of
An anti-causal signal will neglect the values at the negative side.
Convolution in the time domain is equivalent to __________ in frequency domain.
The filter described by following specifications has a passband region at
What is the functional representation of the discrete signal
Multiplication of two sequences in time-domain is _______ in frequency domain
The difference equation of a digital system is described by
Describe the magnitude response of the 6thorder Butterworth filter as .
____ is used to convert digital signals consisting of 0s and 1s to varying analog signals (such as a voltage signal).
What characteristic of a signal is described by the completion of a certain pattern in one cycle?
The ROC of xNo = δ(n+1) - δ(n-1) is:
The input and output of Discrete Fourier transform is discrete and infinite.
An IIR filter requires more delay impulses as compare with an FIR filter which can utilize the same delay element multiple times.
The filter described by following specifications has a stopband attenuation of
The equation that shows the relationship of the past output and present and past input samples and the present output sample is called
A filter with two poles and 2 zeros inside the unit circle is _________ order filter.
To reduce a 192kHz to 44.1 kHz, downsampling may be readily used.
Compression is the transformation of a collection of data typically into a smaller file size.
How many gradations can an 8 bit ADC represent?
The ____converts an analog signal to typically an electrical signal.
____is a property which refers to the scaling or multiplying by a constant.
It is evident that production of audio CDs follows the Nyquist theorem since the sampling frequency used in this is 32 kHz
The relationship of the dualities of Fourier series and Transform
Signal which cannot complete a certain pattern in one cycle is classified as
While referring to the difference equation, if there is a past and present output, the filter is an FIR filter.
In practice, audio signals are sampled at 8 bits, 16 bits and 24 bits.
The transfer function of the difference equation given yNo = xNo –x(n-1) – 2y(n-1) – y(n-2) is:
If the signal xNo = [ 0 1 2 3 4] will use linear interpolation to upsample the output by 2 would be
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:
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
An exponential sequence can be expressed as a geometric series.
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.
The inverse z-transform of X(z) = z – z-1 + z -2 is:
The difference between the Bartlett and triangular windows is that the Fourier transform of a Bartlett window is negative for n even.
Aliasing is a phenomenon which occurs when the sampling frequency is below twice the maximum frequency component of the signal.
An odd signal may be expressed in continuous time as x (t) is equal to
It states that, for a signal to be properly reconstructed, a signal must be sampled twice the maximum frequency component of the signal.
__________ allows a complex plane representation of a digital signal or the system using poles and zeros.
Sampling is the transformation of a collection of data typically into a smaller file size.
__convert ac voltage at one frequency to another ac voltage at another frequency.
Which of the following is true about the side lobe roll-off rate (dB/decade) of windows?
Convert the transfer functions of a causal system into difference equation.
If there are two poles that represent a transfer function, the number of zeros can be 0 or 1.
The type of discrete time signal described by a single impulse is referred to as __________
According to the French mathematician and physicist, ___any continuous periodic signal is could be represented by sum of sinusoids.
What is the z-transform of the signal xNo = 0.5 δ(n-3)?
When dealing with non-causal system, convolution is practically the same with causal; the impulse will always start at n=0.
A/an ____________ function, if applied to a signal before DFT reduces the spectral leakage due to abrupt truncation of the data sequence.
The value of z is the equivalent to the complex value in Fourier Transform if r = 1.
The trade-off of using a higher order Butterworth filter is the
Find the DTFT of the signal x[n] = 0.9n u[n]
Unprocessed physical quantities such as the ____ that we hear are in the form of continuous time.
In signal processing, all input signal begin with a:
It is a phenomenon which occurs when the signal is sampled below the Nyquist rate
Signal which cannot be expressed in simple mathematical form example is random noise.
While referring to the difference equation, if there is a past and present output, the filter is an IIR filter.
The advantage of a Butterworth and elliptic filters is that their roll off is faster but suffers from passband ripple.
It is a process which converts a continuous time signal to discrete-time form.
A positive exponent of z denotes that the shift is
The value of yNo in DFT can be determined using Inverse Fourier Transform
An advanced impulse is placed before the reference 0.
A _______ is used to convert digital signals consisting of 0s and 1s to varying analog signals (such as a voltage signal).
The passband region of the filter is described by following specifications has a gain of
What type of filter is described by following specifications
The angular frequency is equal to the frequency multiplied by a factor of 2.
The typical unwanted result of upsampling in images is in the form of debris and artifacts
Which of the following signals is continuous time, deterministic, aperiodic?
Common among characteristics of both Butterworth and Chebyshev Type II filters are having wide transition bands and flat pass bands.
A mathematical operation that closely resembles convolution by measuring the degree to which the two signals are similar.
Which among the signals is equivalent to u(n-1)?
The location of the poles and zeros provide the characterization of the filter's response.
Where are the zeros of H(z) = 1+ z-2 located?
The approximate mainlobe width of a Bartlett window is:
The values of the new samples when employing linear interpolation is computed by
Interpolation is a method similar to
____________ can be used to applications in communications such as band limiting the signal for transmission
Which of the following Z-transforms is equivalent to xNo = u(n-1)
______________ by an integer factor of L means inserting L-1 zeros for every sample in the data sequence xNo.
The zero padding extends the non-zero value without changing the value of theentire signal
A marginally stable digital filter have pole/s which are located at
It is distinguished from other areas in computer science by the unique type of data it uses.
The zero padding extends the non-zero value by changing the value of the entire signal through rounding off
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?
Signal which exhibits symmetry in the vertical axis is classified as
A____ is used to convert digital signals consisting of 0s and 1s to varying analog signals such as a voltage signal.
Property of DFT which shows additivity and scaling.
If the signal has a sampling rate of 48kHz, to produce a 44.1kHz signal, using integer factors, the signal has to be
In audio signal processing, a microphone acts as the filter of the system.
Signal which cannot be expressed in simple mathematical form and are often expressed using probability.
Which of the following represents the process of upsampling?
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.
A property which shows that xNo in time domain is X(k) in frequency domain
Find the DFT of the discrete time signal xNo = [j, 1, -j, 1]
What is the resulting sampling rate if the original sampling rate is 6000 Hz, up-sampled by 10 and down-sampled by 3?
In audio signal processing, a _____acts as the transducer in the system. In communications, an _____converts electromagnetic waves.
Find the DFT of xNo = [ -1 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.
The synthesis and analysis equation for Fourier Series of continuous periodic signals is equivalent to N point DFT and the inverse DFT, respectively
In DFT multiplication in the time domain is multiplication in the frequency domain
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.
Time reversal property is similar to the
H(z) or the design of the filter is as easy as finding the ratio of the input over the output.
Find the inverse z
The value of z is the equivalent to the complex value in Fourier Transform if r = 0.
The Fourier transform of a Rectangular window is?
Which of the following represents the process of downsampling
In multirate digital signal processing, the factors must be any positive number.
____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.
Signal which exhibits periodicity or can complete a certain pattern in one cycle.
The value of the second sample after upsampling using linear interpolation would be xNo = [ 0 1 2 3 4] by 3 would be:
The magnitude response of a rectangular pulse is a sinc function.
To reduce a 192kHz to 44.1 kHz, resampling must be employed.
DFT provides a discrete frequency representation of a finite-duration sequence in the frequency domain
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.
____allows a complex plane representation of the signal or the system.
The output of Discrete Time Fourier transform is discreteand finite.
In audio signal processing, a microphone acts as the transducer in the system.
For higher-order IIR filters, ______ form can be used for more practical realization.
Find the unit impulse response of yNo= xNo+0.7 x(n-1)
Signal input is a real world signal which is in____.
____ is a discipline that spans electrical engineering, computing, mathematics and the physical sciences.
In practice, audio signals are sampled at 8 bits and below.
The insertion of a replica of the value in between samples is an example of upsampling.
The order of the digital filter given by H(z)= 1 + z-2
How many bits are needed to represent 1,000,000?
Compute for the minimum sampling resolution that could be represented by a 4 bit ADC?
Discrete time signal is represented mathematically by a sequence of numbers x.
What is the sequence representation of the discrete signal described by the functional representation given below?
A causal signal will neglect the values at the negative side.
Given a specification for filter requirements, IIR can be implemented with less order than the FIR filters.
It is the process which involves rounding off discrete values from the sampled signal.
The exponent of z of an advanced impulse is negative.
The magnitude frequency response represents the ________ of the digital filter.
___________ 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 Z-transform has limits from 0 to positive infinity is called unilateral Z-transform.
Digital signal samples are represented by their amplitude versus
In audio, after downsampling, the signal is compressed both in time and frequency domain.
If up sampling a signal involves by inserting non-zero values in between, this multirate DSP is referred to as
During downsampling, information is added with the values inserted in between.
Which of the following in not a discrete time signal?
What is the z-transform of the signal
Downsampling in the time-domain is _________ in frequency domain
When ω = π this corresponds to the____ possible rate of oscillation.
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 typical unwanted result of downsampling in images is in the form of debris and artifacts
The analysis and synthesis equation for Fourier Series of continuous periodic signals is equivalent to N point DFT and the inverse DFT, respectively
Find the DTFT of the discrete time signal xNo = [1, 0, -1, 0]
If the value inserted in between samples is just a replica value of a neighboring sample, it is not considered upsampling.
The ROC of
A converter used to change dc voltage into ac voltage.
A digital filter is considered stable if the poles lie ___________ unit circle.
It is non-invasive test that record the electrical activity of the heart.
Classify the signals with the notation given below:
Linearity in DSP systems states that the principle of _____________ exists.
Which of the following is true about a random signal?
The magnitude response of a rectangular pulse is a sine function.
The notation uNo refers to a __________
Signals are primarily classified into two: continuous time signal and discrete time.
From the given analog signal: xa(t) = 4 sin 4000πt + 8 cos 3000πt. Determine the minimum sampling frequency to avoid aliasing.
The transfer function of the LTI causal system given by yNo = xNo + 2x(n-1) + x(n-2) is?
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.
To keep up this site, we need your assistance. A little gift will help us alot.
Donate- The more you give the more you receive.
Related SubjectElectronics Engineering Technology
Electronics: Electronic Devices and Circuits
Principles of Communication Systems
Fundamentals of Mixed Signals and Sensors
Quantitative Methods
Research in Psychology 2
Psychological Statistics
Accounting Research Methods
Shopee Cashback Voucher
Temu $0 Shipping Fee
Amazon 75% Off Discounts