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