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