http://electronicsbus.com/dsp-matlab-programming-jobs-interview-questions/
http://www.mofeel.net/288-comp-dsp/4202.aspx
http://www.dspguru.com/book/export/html/8
http://maria-college-of-engineering.org/Digital%20Signal%20Processing.pdf
http://www.scribd.com/doc/23537576/dsp-quiz-soln
http://www.ece.cmu.edu/~ee791/homework/q2_2004.pdf
http://www.edaboard.com/thread71741.html
http://en.wikipedia.org/wiki/Surround_sound
http://drewdaniels.com/
http://www.scribd.com/doc/39908949/Dsp-Question-Bank-With-Solutions
http://allinterviewquestions.page.tl/DSP-Interview-Questions.htm
http://www.edaboard.com/thread71741.html
Some questions:
1. Differentiate between IIR and FIR filters.
2. What is the algorithm for a simple IIR filter, FIR filter?
3. What is bilinear transformation? Why is it required to be done.
4. If I average the height of 10 students to get a mean, and keep doing for the next 10 student set, what sort of filtering is involved in this?
5. How does 4 differ from a moving average filter?
6. What is aliasing? How can that be reduced?
7. What is the bandwidth for voice? What is the standard sampling frequency used in sampling voice?
8. If voice is added to a 9kHz tone and sampled and recovered, what will be the effect seen of the 9kHz tone?
9. What is the processor used for DSP operations? What is MAC?
10. AT what rate should I sample a vibration signal obatined from monitoring vibration in a car?
1. Differentiate between IIR and FIR filters.
2. What is the algorithm for a simple IIR filter, FIR filter?
3. What is bilinear transformation? Why is it required to be done.
4. If I average the height of 10 students to get a mean, and keep doing for the next 10 student set, what sort of filtering is involved in this?
5. How does 4 differ from a moving average filter?
6. What is aliasing? How can that be reduced?
7. What is the bandwidth for voice? What is the standard sampling frequency used in sampling voice?
8. If voice is added to a 9kHz tone and sampled and recovered, what will be the effect seen of the 9kHz tone?
9. What is the processor used for DSP operations? What is MAC?
10. AT what rate should I sample a vibration signal obatined from monitoring vibration in a car?
Q. Why should we go for digital signal processing where as the most of the real world data is in analog mode?
Q. What are the differences between a microprocessor and a DSP processor?
Q. What is the convolution?
Q. Why do we need Forrier transform in DSP?
Q. What is use of windowing in digital filters?
Q. Tell some thing about Interpolation and decimation?
Q. What is the need of FFT ?
Q. What’s the difference between FFT and DFT?
Q. What is the advantage of a Direct form II FIR over fom I?
Q. What is the difference between equiripple filter and FIR filter?
Q. What is the application fo Cross correlation and Auto Correlation?
Q. Explain using convolution the effects of taking an FFT of a sampe with no windowing (rectangular window).
Q. What are basis vectors in a transofrm?
Q. In signal processing, why we are much more interested in orthogonal transform?
Q. How does polyphase filtering save computations in a decimation filter?
Q. Why IIR filters doesnt have Linear phase?
Q. Whats basic difference b/w winer filter and kalman filter and lms filter
Q. What is the use of windowing in digital filters
Q. What are the pros and cons of Discrete Cosine Transform?
Q. What is Interpolation and decimation filters and why we need it?
Q. What is the simplest high pass filter ? write the equation?
Q. What is the difference between ProtoPlus and ProtoPlus Lite?
Q. What is Auto Regressive Model? How is the order of auto regressive model is decided?
Q. What is the basic difference between FIR and IIR filters?
Q. What two PSK modulation orders differ exactly by a factor of two in spectral efficiency?
Q. Under what conditions is the available bandwidth of a digital system Fs Hz instead of Fs/2 Hz?
Q. What is the difference between DFT and DTFT?
Q. FFT is in complex domain how to use it in real life signals optimally?
Q. What is Gibbs phenomenon?
Q. Can we create a table with out primary key?
Q. Suppose we have a system with transfer function H(z) = 1 / ((z – 1.1)*(z – 0.9)). Is the system stable or unstable?
Q. Differences b/w butterworth chebyshev?
Q. How do you reduce spectral leakage?
Q. Why is FFT faster than DFT? what is the actual concept behind this?
Q. Is the Gibbs phenomenon ever a factor?
Q. What is the concept of stability of an LTI system? How to check if a given system is stable?
Q. How does polyphase filtering save computations in an interpolation filter?
Q. If a have two vectors how will i check the orthogonality of those vectors?
Q. Can IIR filters be Linear phase? how to make it linear Phase?
Q. How can you compute fourier transform form Z-transform ?
Q. How is the non-periodic nature of the input signal handled?
Q. What is aliasing and how do we prevent it?
Q. How can you determine the stability of an LTI system?
Q. What is the need of Digital Signal Processing?
Q. Why do we need I&Q signals?
Q. What do you mean by spectral resolution?
Q. What is the special about minimum phase filter?
Q. How can you compute fourier transform form Z-transform ?
Q. Why after DCT we use a zig zag manner for run length coding?
Q. Why we use DCT extensilvely in compression?
Q. Can you write assembly language programs for DSP?
Q. What is your proficiency level of C-language for DSP applications?
########################################
A4d56 eng5neer5ng
What is the bit-depth of CD quality audio? And what is its dynamic
range?
- Explain 'dither'
- What is dBSPL?
- Differences between FIR and IIR filters
- Explain the auditory masking phenomenon.
1a - what is the general relationship between bit-depth and dynamic
range?
1b - what is the relationship between S/N, headroom, and dynamic
range?
1c - when is floating-point superior to fixed and when isn't it?
2a - rectangular dither
2b - triangular dither
2c - high-pass dither
3a - what's the physical reference level for dB_SPL?
4a - what are some common FIR design techniques (there are 2 that i
know)?
4b - what are some common IIR design techniques?
5a - how is auditory masking useful for audio data compression?
6 - What is white noise and pink noise and how might one generate
either?
7 - Level compression
7a - what are compression ratio, attack and release time parameters?
7b - what is soft knee vs. hard knee?
7c - what is "pumping"?
8 - FFT or DFT, what is the relationship between data length, sampling-
rate, and frequency resolution?
Does upsampling an existing 44100Hz/16bit signal to 96000Hz/24bit
increase "resolution"?
Why might a linear phase filter be undesirable in audio?
Why might FFT-based filtering be a bad idea in equipment that is used
for monitoring live performances?
What is the difference between "lossy" compression, like MP3, and
lossless compression, like MLP or FLAC?
Many modern dynamic range compressor/limiters, when operating upon
44100Hz input signals, always upsample to 88200Hz or 96000Hz before
compressing or limiting. Why?
Under what circumstances would a digital volume control be a bad idea?
Why might two pieces of equipment that use exactly the same analog to
digital converters, DSPs, and digital to analog converters, sound very
different?
1. Differentiate between IIR and FIR filters.
2. What is the algorithm for a simple IIR filter, FIR filter?
3. What is bilinear transformation? Why is it required to be done.
4. If I average the height of 10 students to get a mean, and keep doing for the next 10 student set, what sort of filtering is involved in this?
5. How does 4 differ from a moving average filter?
6. What is aliasing? How can that be reduced?
7. What is the bandwidth for voice? What is the standard sampling frequency used in sampling voice?
8. If voice is added to a 9kHz tone and sampled and recovered, what will be the effect seen of the 9kHz tone?
9. What is the processor used for DSP operations? What is MAC?
10. AT what rate should I sample a vibration signal obatined from monitoring vibration in a car?
#######################
how to classify signals?
discreet time/continous time, periodic/nonperiodic
what is the use of random signals?
for testing systems dynamic response statistically for very small amplitudes and time durations.
how to classify systems?
linear, stable, time invariant
what's the difference between correlation & convolution? how to obtain convolution from correlation operator *?
conv(h,x) = h*(-x)
what are the characteristics of a transient response of a system?
decay time, rise time, peak time, max overshoot, settling time
whats sampling thereom? who proposed it? whats importance?
Why is the need of FFT ?
What is the need of Digital Signal Processing?
what is the convolution?
What is aliasing and how do we prevent it?
Aliasing is frequency shifting of content of input signal above half the sample frequency. We avoid it by using antialiasing filters to limit signal content below 1/2 sample frequency and/or by sampling at high enough frequency to avoid antialiasing.
What is the application fo Cross correlation and Auto Correlation?
what do u mean by spectral resolution?
i think its separetion of two signal in frequency domain.
how do u reduce spectral leakage?
By incresing the window lenth in time domain we can reduce spectral leakage.
What is the concept of stability of an LTI system? How to check if a given system is stable?
Determine the roots of the system. If all roots have real part <> 0, the system is unstable.
What is the importance of time frequency resolution? Can this be explained with respect to some application eg when we need to determine the stationary and dynamic characteristic of a signal, how do transform with different time frequency resolution, affects it.
Time and frequency are two important aspects in which time and frequency are the inverse of each other.Hence if u have a small frequency it is spread out in large time and vice versa. Just u can analyze a time frequency plot.
So when we try to analyze non stationary signals these values continuously change and the properties of FFT do not hold for non stationary signals for which u need to go to either Short term Fourier trnasform or the wavelet analysis.
Each has it's own adv and disadv.
Tell some thing about Interpolation and decimation?
Intepolation basically filling missing samples,
decimation basically reducing the sample rate.
Explain using convolution the effects of taking an FFT of a sampe with no windowing (rectangular window).
The signal is multiplied by a rectangular window in the time domain. This corresponds to convolution by a sync function in the frequency domain. The wide center lobe of the sync function spreads frequencies and the side lobes shift frequencies.
What is Gibbs phenomenon?
The Gibbs phenomenon in time is the result of truncating its
corresponding discrete Fourier Transform, or, Fourier Series. The Gibbs
phenomenon in frequency is the result of truncating its sampled time series.
If a have two vectors, how will i check the orthogonality of those vectors?
Why do we need I&Q signals?
what care should be taken when we doing camulative decimation?
Transforms
In signal processing, why we are much more interested in orthogonal transform?
What are the pros and cons of Discrete Cosine Transform?"
What are basis vectors in a transofrm?
how can you compute fourier transform form Z-transform ?
z =r*e(-iw) when r=1 Z-transform converges to discrete Fourier tranform.
Why is FFT faster than DFT? what is the actual concept behind this?
FFT is an algorithm to implement DFT , FFT is not at all a tranform,
if u implement Directly it take (N^2) complex Multiplications and (N^2 -N) complex additions , where as FFT will Nlog2(N) complex Mul & additions
what is the difference between DFT and DTFT?
DFT = spectrum of DFT also discrete in nature,where as spectrum of DTFT is continuous.
why we use DCT extensilvely in compression?
DCT has excellent energy compaction property, for highly correlated input it give excellent energy compaction similar to KL transform.
why after DCt we use a zig zag manner for run length coding?
to convert to 2d to 1d data for run lenth coding, zig-zag is prefered because it scan the 2d data from low frequency to high frequency coef.
FFT is in complex domain, how to use it in real life signals optimally?
use two diiferent buffers for real & complex
why we use Forrier transform in DSP
how to find energy density (enrgy per unit bandwidth) from fourier transform?
| F(jw)| pow 2
why do we use laplace transform?
analysis & design of linear time invariant systems
can u compare laplace & fourier transforms? any disadvantages for fourier?
in a strict sense, fourier is not applicable for unit step, unit ramp and sinusoidal functions. this is overcome in laplace.
what are advantages of laplace transform?
simplifies operations, transforms frequently occuring exponential function to simpler easier to handle algebraic functions.
what's the fourier transform of a auto correlation for a periodic waveform ?
power spectral density
Filters
what is use of windowing in digital filters?
How toget a magnitude response of a quantized filter?
Any noise estimation has to be done??
What is the difference between equiripple filter and FIR filter?
whats basic difference b/w winer filter and kalman filter and lms filter ?
What is the advantage of a Direct form II FIR over fom I?
Why IIR filters doesnt have Linear phase?.
Liner phase IIR filter cannot physically realizable , its unstable.
Can IIR filters be Linear phase? how to make it linear Phase?
Idealy Physically realizable doesnt have linear phase, but we can implement IIR filter with linear phase response in passband. (refer Bessel series Approx)
What is the advantage of a Direct form II FIR over fom I?
i think Direct form -2 take less memory (not sure)
What is the basic difference blw FIR and IIR filters?
basic difference is FIR has liner phase response,where has IIR filter has nonlinear phase response.
what is the special about minimum phase filter ?
minimum phase system has both poles and zeors reside in side unit circle, these are basically used for
compensating channel impairments.
differences b/w butterworth, chebyshev, elliptical filter and advantages/disadvantages of each?
butter worth -monotonic response in pass band and stop band.
chebyshev-1 ripple in pass band, and monotonic in stop band.
chebyshev-2 monotonic in passband ripple in stop band.
elliptic- ripple response both in passband &stop band.
for given set of parameters we can achive minimum transistion band width with elliptic filter , other way
we can implement with fewer number of coef than other methods
What is Interpolation and decimation filters and why we need it?
interpolation for reconstruction and for sample rate up conversion.
deciamtion filter sample rate down conversion.
what is the simplest high pass filter ? write the equation?
Hp= 1- Lp
DSP Question bank
UNIT 1
1. Determine the energy of the discrete time sequence (2)
x(n) = (½)n, n≥0 . {W}^3 dot aim for high dot blogspot dot com
=3 n, n<0 2. Define multi channel and multi dimensional signals (2)
3. Define symmetric and anti symmetric signals. (2)
4. Differentiate recursive and non recursive difference equations. (2)
5. What is meant by impulse response? (2)
6. What is meant by LTI system? (2)
7. What are the basic steps involved in convolution? (2)
8. Define the Auto correlation and Cross correlation? (2)
9. What is the causality condition for an LTI system? (2)
10. What is zero padding? What are it uses? (2)
11. State the Sampling Theorem. (2) {W}^3 dot aim for high dot blogspot dot com
12 What is an anti imaging and anti aliasing filter? (2) 13. Determine the signals are periodic and find the fundamental period (2) in√ 2 πt i) sin 20πt+ sin5πt 14. Give the mathematical and graphical representations of a unit sample, unit step sequence. (2) 15. Sketch the discrete time signal x(n) =4 δ (n+4) + δ(n)+ 2 δ (n-1) + δ (n-2) -5 δ (n-3) (2) 16. Find the periodicity of x(n) =cost(2πn / 7) (2) 17. What is inverse system? (2) 18. Write the relationship between system function and the frequency response. (2) 19. Define commutative and associative law of convolutions. (2) 20. What is meant by Nyquist rate and Nyquist interval? (2) 21. What is an aliasing? How to overcome this effect? (2) 22. What are the disadvantages of DSP? (2) 23. State initial value theorem of Z transform. (2) 24 What are the different methods of evaluating inverse z transform? (2) 25 What is meant by ROC? (2) 26 What are the properties of ROC?(2) 27 What is zero padding? What are it uses?(2) 28 State convolution property of Z transform. (2) 29 State Cauchy residue theorem. (2) 30 Define fourier transform. (2) 31 Define discrete fourier series. (2) 32 Compare linear and circular convolution. (2) 33 Distinguish between Fourier series and Fourier transform. (2) 34 What is the relation between fourier transform and z transform. (2) 35 What is the use of Fourier transform? (2) 36. Define system function. (2) 37. State Parseval relation in z transform (2) CLASSIFICATION OF SYSTEMS: 1. Determine whether the following system are linear, time-invariant (16) i)y(n) = Ax(n) +B ii)y(n) =x(2n) iii)y(n) =n x2 (n) iv)y(n) = a x(n) 2. Check for following systems are linear, causal, time in variant, stable, static (16) i) y(n) =x(2n) ii) y(n) = cos (x(n)) iii) y(n) = x(n) cos (x(n) iv) y(n) =x(-n+2) v) y(n) =x(n) +n x (n+1) 3.a) For each impulse response determine the system is i) stable ii) causal (8) i) h(n)= sin (π n / 2) ii) h(n) = δ(n) + sin π n iii) h(n) = 2 n u(-n) . b)Find the periodicity of the signal x(n) =sin (2πn / 3)+ cos (π n / 2) (8) 4. Explain in detail about A to D conversion with suitable block diagram and to reconstruct the signal. (16) 5 a) State and proof of sampling theorem. (8) b)What are the advantages of DSP over analog signal processing? (8) 6 a)Explain successive approximation technique. (8) b)Explain the sample and hold circuit. (8) Z TRANSFORM: 1. a)State and proof the properties of Z transform. (8) b)Find the Z transform of (8) i) x(n) =[ (1/2)n – (1/4)n ] u(n) ii) x(n) = n(-1)n u(n) iii) x(n) (-1)n cos (πn/3) u(n) iv) x(n) = (½) n-5 u(n-2) +8(n-5) 2 a) Find the Z transform of the following sequence and ROC and sketch the pole zero diagram (8) i) x(n) = an u(n) +b n u(n) + c n u(-n-1) , |a| <|b| <| c| ii) x(n) =n2 an u(n) b)Find the convolution of using z transform (8) x1(n) ={ (1/3) n, n>=0
(1/2) - n n<0 } x2(n) = (1/2) n INVERSE Z TRANSFORM: 5. Find the inverse z transform (16) X(z) = log (1-2z) z < |1/2 | X(z) = log (1+az-1) |z| > |a|
X(z) =1/1+az-1 where a is a constant
X(z)=z2/(z-1)(z-2)
X(z) =1/ (1-z-1) (1-z-1)2
X(z)= Z+0.2/(Z+0.5)(Z-1) Z>1 using long division method.
X(z) =1- 11/4 z-1 / 1-1/9 z-2 using residue method.
X(z) =1- 11/4 z-1 / 1-1/9 z-2 using convolution method.
6.. A causal LTI system has impulse response h(n) for which Z transform is given by H(z)
1+ z -1 / (1-1/2 z -1 ) (1+1/4 z -1 ) (16)
i) What is the ROC of H (z)? Is the system stable?
ii) Find THE Z transform X(z) of an input x(n) that will produce the output y(n) = - 1/3
(-1/4)n u(n)- 4/3 (2) n u(-n-1)
iii) Find the impulse response h(n) of the system.
ANALYSIS OF LTI SYSTEM:
7. a)The impulse response of LTI system is h(n)=(1,2,1,-1).Find the response of the system to
the input x(n)=(2,1,0,2) (8)
b). Determine the response of the causal system y(n) – y(n-1) =x(n) + x(n-1) to inputs
x(n)=u(n) and x(n) =2 –n u(n).Test its stability (8)
8. Determine the magnitude and phase response of the given equation
y(n) =x(n)+x(n-2) (16)
9. a)Determine the frequency response for the system given by
y(n)-y3/4y(n-1)+1/8 y(n-2) = x(n)- x(n-1) (8)
b). Determine the pole and zero plot for the system described difference equations
y(n)=x(n)+2x(n-1)-4x(n-2)+x(n-3) (8)
10. Find the output of the system whose input- output is related by the difference equation
y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the step input. (16)
11. Find the output of the system whose input- output is related by the difference equation
y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the x(n) =4 n u(n). (16)
CONVOLUTION:
12. Find the output of an LTI system if the input is x(n) =(n+2) for 0≤ n≤ 3
and h(n) =an u(n) for all n (16)
13. Find the convolution sum of x(n) =1 n = -2,0,1
= 2 n= -1
= 0 elsewhere
and h(n) = δ (n) – δ (n-1) + δ( n-2) - δ (n-3) (16).
14. Find the convolution of the following sequence x(n) =(1,2,-1,1) , h(n) =(1, 0 ,1,1) (16)
15.Find the output sequence y(n) if h(n) =(1,1,1) and x(n) =(1,2,3,1) using a circular
Convolution. (16)
16. Find the convolution y(n) of the signals (16)
x(n) ={ α n, -3 ≤ n ≤ 5 and h(n) ={ 1, 0 ≤ n ≤ 4
0, elsewhere } 0, elsewhere }
UNIT 2
1. How many multiplication and additions are required to compute N point DFT using
radix 2 FFT? (2)
2. Define DTFT pair. (2)
3. What are Twiddle factors of the DFT? (2)
4. State Periodicity Property of DFT. (2)
5. What is the difference between DFT and DTFT? (2)
6. Why need of FFT? (2)
7. Find the IDFT of Y (k) = (1, 0, 1, 0) (2)
8. Compute the Fourier transform of the signal x(n) = u(n) – u(n-1). (2)
9. Compare DIT and DIF? (2)
10. What is meant by in place in DIT and DIF algorithm? (2)
11. Is the DFT of a finite length sequence is periodic? If so, state the reason. (2)
12. Draw the butterfly operation in DIT and DIF algorithm? (2)
13. What is meant by radix 2 FFT? (2)
14. State the properties of W N
k ? (2)
15. What is bit reversal in FFT? (2)
16. Determine the no of bits required in computing the DFT of a 1024 point sequence
with SNR of 30dB. (2)
17. What is the use of Fourier transform? (2)
18. What are the advantages FFT over DFT?
FOURIER TRANSFORM:
1. a) Determine the Fourier transform of x (n) =a |n|; -1 (8)
b) Determine the Inverse Fourier transform H (w) = (1-ae-jw) -1 (8)
2. State and proof the properties of Fourier transform (16)
FFT:
3. Determine the Discrete Fourier transform x (n) = (1, 1, 1, 1) and
Proof x(n)*h(n) =X(z) H(z (16)
4. Derive and draw the 8 point FFT-DIT butterfly structure. (16)
5. Derive and draw the 8 point FFT-DIF butterfly structure. (16)
6.Compute the DFT for the sequence.(0.5,0.5,0.5,0.5,0,0,0,0) (16)
7.Compute the DFT for the sequence.(1,1,1,1,1,1,0,0) (16)
8.Find the DFT of a sequence x(n)=(1,1,0,0) and find IDFT of Y(k) =(1,0,1,0) (16)
9. If x (n) = sin (nΠ/2), n=0, 1, 2, 3 (16)
10. h (n) = 2 n , n=0,1,2,3.Find IDFT and sketch it. (16)
11.Find 4 point DFT using DIF of x(n) =(0,1,2,3) (16)
12.a)Discuss the properties of DFT. (10)
b).Discuss the use of FFT algorithm in linear filtering. (6)
UNIT 3& 4
1. Define canonic and non canonic form realizations. (2)
2. Draw the direct form realizations of FIR systems? (2)
3. Mention advantages of direct form II and cascade structure? (2)
4. Define Bilinear Transformation. (2)
5. What is prewar ping? Why is it needed? (2)
6. Write the expression for location of poles of normalized Butterworth filter. (2)
7. Distinguish between FIR and IIR Filters. (2)
8. What is linear phase filter? (2)
9. What are the design techniques available for IIR filter? (2)
10. What is the main drawback of impulse invariant mapping? (2)
11. Compare impulse invariant and bilinear transformation. (2)
12. Why IIR filters do not have linear phase? (2)
13. Mention the properties of Butterworth filter? (2)
14. Mention the properties of Chebyshev filter? (2)
15. Why impulse invariant method is not preferred in the design of high pass IIR filter? (2)
16. Give the transform relation for converting LPF to BPF in digital domain. (2)
17. What are Gibbs oscillations? (2)
18. Explain briefly Hamming window (2).
19. If the impulse response of the symmetric linear phase FIR filter of length 5 is h(n) =
{2, 3, 0, x, y), then find the values of x and y. (2)
20. What are the desirable properties of windowing technique? (2)
21. Write the equation of Bartlett window. (2)
22.Why IIR filters do not have linear phase? (2)
23.Why FIR filters are always stable? (2)
24.Why rectangular window are not used in FIR filter design using window method? (2)
25.What are the advantages of FIR filter? (2)
26.What are the advantages and disadvantages of window? (2)
27.What is the necessary condition and sufficient condition for the linear phase characteristic of a
FIR filter? (2)
28.Compare Hamming and Hanning window? (2)
29.Why triangular window is not a good choice for designing FIR Filter? (2)
30.Why Kaiser window is most used for designing FIR Filter? (2)
31.What is the advantages in linear phase realization of FIR systems? (2)
Structures of IIR systems:
1. Obtain the cascade and parallel form realizations for the following systems (16)
Y (n) = -0.1(n-1) + 0.2 y (n-2) + 3x (n) +3.6 x (n-1) +0.6 x (n-2)
2.a) Obtain the Direct form II
y (n) = -0.1(n-1) + 0.72 y(n-2) + 0.7x(n) -0.252 x(n-2) (8)
b) .Find the direct form II
H (z) =8z-2+5z-1+1 / 7z-3+8z-2+1 (8)
3. Obtain the i) Direct forms ii) cascade iii) parallel form realizations for the following systems
y (n) = 3/4(n-1) – 1/8 y(n-2) + x(n) +1/3 x(n-1) (16)
4.Find the direct form –I, cascade and parallel form for (16)
H(Z) = z -1 -1 / 1 – 0.5 z-1+0.06 z-2
IIR FILTER DESIGN:
6. Explain the method of design of IIR filters using bilinear transform method. (16)
7. a)Derive bilinear transformation for an analog filter with system function H(s) = b/ s + a (8)
b) For the analog transfer function H(s) = 2 / (s+1) (s+3) .
Determine H (z) using bilinear transformation. With T=0.1 sec (8)
8. a)Convert the analog filter H(s) = 0.5 (s+4) / (s+1)(s+2) using impulse invariant transformation
T=0.31416s (8)
b)The normalized transfer function of an analog filter is given by H a (sn) = 1/ sn
2 +1.414 s n +1.
Convert analog filter to digital filter with cut off frequency of 0.4 π using bilinear transformation.
(8)
9. Design a single pole low pass digital IIR filter with -3db bandwidth of 0.2Π by using bilinear
transformation. (16)
10. For the constraints
0.8 ≤ |H (e jw)| ≤1, 0 ≤ ω ≤ 0.2π
|H (e jw)| ≤0.2, 0.6π ≤ ω ≤π with T= 1 sec .Determine system function H(z) for a Butterworth
filter using Bilinear transformation. (16)
11.Design a digital Butterworth filter satisfying the following specifications
0.7 ≤ |H (e jw)| ≤1, 0 ≤ ω ≤ 0.2π
|H (e jw)| ≤0.2, 0.6π ≤ ω ≤π with T= 1 sec .Determine system function H(z) for a Butterworth
filter using impulse invariant transformation. (16)
12. Design a digital Chebyshev low pass filter satisfying the following specifications 0.707 ≤ |H (e
jw)| ≤1, 0 ≤ ω ≤ 0.2π
|H (e jw)| ≤0.1 0.5 ≤ ω ≤π with T= 1 sec using for bilinear transformation. (16)
13.Design a digital Butterworth High pass filter satisfying the following specifications
0.9 ≤ |H (e jw)| ≤1, 0 ≤ ω ≤ π/2
|H (e jw)| ≤0.2, 3π/4 ≤ ω ≤π with T= 1 sec. using impulse invariant transformation (16)
14. Design a realize a digital filter using bilinear transformation for the following specifications
i) Monotonic pass band and stop band
ii) -3.01 db cut off at 0.5 π rad
iii) Magnitude down at least 15 db at ω = 0.75 π rad. (16)
FIR FILTER
15.a) Prove that an FIR filter has linear phase if the unit sample response satisfies the condition h(n)
= ± h(M-1-n), n =0,1,….. M-1.Also discuss symmetric and anti symmetric cases of FIR filter.
(8)
b) Explain the need for the use of window sequence in the design of FIR filter. Describe the
window sequence generally used and compare the properties. (8)
16. Design a HPF of length 7 with cut off frequency of 2 rad/sec using Hamming window. Plot the
magnitude and phase response. (16)
17. Explain the principle and procedure for designing FIR filter using rectangular window (16)
18. Design a filter with
H d (e
jώ) = e - 3 jώ , π/4 ≤ ω ≤ π/4
0. π/4 ≤ ω ≤ π using a Hamming window with N=7. (16)
19. H (w) =1 for | ω | ≤ π/3 and | ω | ≥2 π/3
0 otherwise for N=11. and find the response. (16)
20.Design a FIR filter whose frequency response (16)
H (e jώ) = 1 π/4 ≤ ω ≤ 3π/4
0. | ω | ≤3 π/4.
Calculate the value of h(n) for N=11 and hence find H(z).
21.Design an ideal differentiator with frequency response H (e jώ) = jw -π ≤ ω ≤ π
using hamming window for N=8 and find the frequency response. (16)
22.Design an ideal Hilbert transformer having frequency response
H (e jώ) = j -π ≤ ω ≤ 0
-j 0 ≤ ω ≤ π for N=11 using rectangular window. (16)
FIR structures:
23.a) Determine the direct form of following system (8)
H (z) =1+2z-1 - 3z-2 + 4z-3 - 5z-4
b). Obtain the cascade form realizations of FIR systems (8)
H (z) = 1+5/2 z-1+ 2z-2 +2 z-3
UNIT 5
1) Give some example of DSP
2) Explain Interpolation
3) Explain decimation
4) What is know to be subband coding
5) Define sampling rate conversion
6) How the image enchancement is achieved using DSP
7) Define compression
8) What are various compression technique
9) Explain subband coding
10) Define vocoders
11) Explain adaptive filtering
1) Explain the concept of deciation by a factor D and interpolation by factor I
2) With help of equation explain sampling rate conversion by a rational factor I/D
3) Explain the following application
i) speech compression
ii) sound processing
4) With help diagram explain adaptive filtering process
5) Explain speech vocoders and subband coding
6) Explain how image enchancement was achieved
click here to read more: http://aimforhigh.blogspot.com/2011/07/cs2403-question-bank-download-cse.html#ixzz2CERhbIIe
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Ref
http://aimforhigh.blogspot.in/2011/07/cs2403-question-bank-download-cse.html
http://www.abdn.ac.uk/piprg/wiki/images/6_DSP_Answer.pdf
http://www.scribd.com/doc/39908949/Dsp-Question-Bank-With-Solutions
http://www.prosoundweb.com/article/the_real_world_stage_monitoring_quiz/
http://www.atrochatro.com/quiz_electronics.html
http://drewdaniels.com/audioquiz.htm
http://www.proprofs.com/quiz-school/story.php?title=audio-engineering-chapter-2
http://flashcarddb.com/cardset/59098-intro-to-audio-quiz-1-flashcards
http://www.sacet.edu.in/IT/6th%20Sem/6th%20sem.-EC%201358-DSP.pdf
http://www.indiastudychannel.com/forum/93819-Digital-signal-processing-question-bank.aspx
https://5d342805-a-62cb3a1a-s-sites.googlegroups.com/site/eeecubebg/storehouse/1-dsp%202m_opt.pdf?attachauth=ANoY7cq0WDVZGPWWJvtwwKXXaiL3KKw-Oc4Sdeaw77IgkKM1p0RO45kx-DnU_i_XfFCXm9RxD66fiE_tR8V94OXaRs4EzE6w9XRWqPFgjcwFXdHlZ_Ryn3M0aOC4Hv8qwH_e9FGXqtYz-ucpSMPyPLto62ygx916bYVZyi4Z27Fghza0Enc42PucIKNCL1JKu_E1DkSMwEG7N6Y_NRiNyY0Awmc1YaTfo2FfJG7ypnxfAVG-ASthZBI%3D&attredirects=0
http://www.eeecube.com/2012/03/ec2312-digital-signal-processing.html#axzz2CCtvTPaG
http://www.iannauniversity.com/2011/07/ec2314-digital-signal-processing.html
http://www.mofeel.net/288-comp-dsp/4202.aspx
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