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Two Marks Questions & Answers
DIGITAL SIGNAL PROCESSING Sem/Year: 05/III
UNIT – I –FFT & DFT
Discrete Signals and Systems- A Review – Introduction to DFT – Properties of DFT –
Circular Convolution - Filtering methods based on DFT – FFT Algorithms –Decimation in
time Algorithms, Decimation in frequency Algorithms – Use of FFT in Linear Filtering.
3. How many multiplication and additions are required to compute N point DFT
using radix 2 FFT? (NOV/DEC 2004)
The number of multiplications and additions required to compute N-point DFT
using radix-2 FFT are N log2N and N/2 log2N
4. Define DFT of a sequence.
N-1
X -j2πkn/N
(k) = ∑ x(n) e
n=0 k = 0,1…..N-1
5. State Periodicity Property of DFT.
If X(n) is N- point DFT of a finite duration sequence x(n), then
x(n+N) = x(n) for all n
X(k+N) = X(k) for all k
6. What is the difference between DFT and DTFT? (MAY/JUNE 2009)
DFT
* Obtained by performing sampling operation in both the time and frequency
domain
* Discrete frequency spectrum
DTFT
* Sampling is performed only in time domain
* Continous function of frequency spectrum
7. Why need of FFT
for reasonably large values of N direct evaluation of the DFT requires an inordinate amount
of computations. By using FFT algorithms the number of computations can be reduc
8. What is meant by radix 2 FFT?
The FFT algorithm is most efficient in calculating N-point DFT. If the number of
M
output points N can be expressed as a power of 2, that is N = 2 where M is the integer,
then this algorithm is known as radix 2 FFT algorithm.
9. What is the main advantage of FFT?
FFT reduces the computation time required to compute discrete fourier transform.
10. State the properties of DFT.
1) Periodicity
2) Linearity and symmetry
3) Multiplication of two DFTs
4) Circular convolution
5) Time reversal
6) Circular time shift and frequency shift
7) Complex conjugate
8) Circular correlation
11. How to obtain the output sequence of linear convolution through
circular convolution?
Consider two finite duration sequences x(n) and h(n) of duration L samples and M
samples. The linear convolution of these two sequences produces an output sequence of
duration L+M-1 samples, whereas, the circular convolution of x(n) and h(n) give N
samples where N=max(L,M).In order to obtain the number of samples in circular
convolution equal to L+M-1, both x(n) and h(n) must be appended with appropriate
number of zero valued samples. In other words by increasing the length of the sequences
x (n) and h(n) to L+M-1 points and then circularly convolving the resulting sequences we
obtain the same result as that of linear convolution.
12. What is zero padding? What are its uses? (NOV 2006,DEC 2009)
Let the sequence x (n) has a length L. If we want to find the N-point DFT(N>L)
of the sequence x(n), we have to add (N-L) zeros to the sequence x(n). This is known as
zero padding.
The uses of zero padding are
1) We can get better display of the frequency spectrum.
2)With zero padding the DFT can be used in linear filtering.
13. Define sectional convolution.
If the data sequence x(n) is of long duration it is very difficult to obtain the output
sequence y(n) due to limited memory of a digital computer. Therefore, the data sequence
is divided up into smaller sections. These sections are processed separately one at a time
and controlled later to get the output
14. What are the two methods used for the sectional convolution?
The two methods used for the sectional convolution are
1) overlap-add method
2)overlap-save method.
15. What is overlap-add method?
In this method the size of the input data block xi(n) is L. To each data block we
append M-1 zeros and perform N point cicular convolution of xi(n) and h(n). Since each
data block is terminated with M-1 zeros the last M-1 points from each output block must
be overlapped and added to first M-1 points of the succeeding blocks.This method is
called overlap-add method.
16. What is overlap-save method?
In this method the data sequence is divided into N point sections xi(n).Each
section contains the last M-1 data points of the previous section followed by L new data
points to form a data sequence of length N=L+M-1.In circular convolution of xi(n) with
h(n) the first M-1 points will not agree with the linear convolution of xi(n) and h(n)
because of aliasing, the remaining points will agree with linear convolution. Hence we
discard the first (M-1) points of filtered section xi(n) N h(n). This process is repeated for
all sections and the filtered sections are abutted together.
17.Why FFT is needed?
The direct evaluation DFT requires N2 complex multiplications and N2 –N
complex additions.Thus for large values of N direct evaluation of the DFT is difficult.By
using FFT algorithm the number of complex computations can be reduced. So we use
FFT.
18.What is FFT? NOV/DEC 2006
The Fast Fourier Transform is an algorithm used to compute the DFT. It makes
use of the symmetry and periodicity properties of twiddle factor to effectively reduce
the DFT computation time.It is based on the fundamental principle of decomposing the
computation of DFT of a sequence of length N into successively smaller DFTs.
19.What is DIT algorithm?
Decimation-In-Time algorithm is used to calculate the DFT of a N point
sequence. The idea is to break the N point sequence into two sequences, the DFTs of
which can be combined to give the DFt of the original N point sequence.This algorithm is
called DIT because the sequence x(n) is often splitted into smaller sub- sequences.
20.What DIF algorithm?
It is a popular form of the FFT algorithm. In this the output sequence X(k) is
divided into smaller and smaller sub-sequences , that is why the name Decimation In
Frequency.
21.What are the applications of FFT algorithm?
The applications of FFT algorithm includes
1) Linear filtering
2) Correlation
3) Spectrum analysis
22. Why the computations in FFT algorithm is said to be in place?
Once the butterfly operation is performed on a pair of complex numbers (a,b) to
produce (A,B), there is no need to save the input pair. We can store the result (A,B) in the
same locations as (a,b). Since the same storage locations are used troughout the
computation we say that the computations are done in place.
23 .What are the differences and similarities between DIF and DIT algorithms?
(NOV/DEC 2006)(MAY/JUNE 2009)
Differences:
1)The input is bit reversed while the output is in natural order for DIT, whereas for DIF
the output is bit reversed while the input is in natural order.
2)The DIF butterfly is slightly different from the DIT butterfly, the difference being
that the complex multiplication takes place after the add-subtract operation in DIF.
Similarities:
Both algorithms require same number of operations to compute the DFT. Both
algorithms can be done in place and both need to perform bit reversal at some place
24. Distinguish between linear convolution and circular convolution of two sequences
MAY/JUNE 2006
25. What are differences between overlap-save and overlap-add methods.
26. Establish the relation between DFT and z transform
The z transform of a sequence is X(z) is n=∞ _1
X(z) = ∑ x(n) z
n=-∞
with a ROC that includes the unit circle.
If X(z) is sampled at the N equally spaced points on the unit circle j2Πk/N
,k=0,1,2...N-1
zk=e
We obtain X(k)=X(z) at z= j2Πk/N ,k=0,1,2...N-1
e
27. Define twiddle factor of FFT. NOV/DEC 2009
The complex number WN is called phase factor or twiddle factor. The WN represent
a
complex number e-j2Π/N.It is used to reduce the computational complexity.
UNIT II INFINITE IMPULSE RESPONSE DIGITAL FILTERS
Structures of IIR – Analog filter design – Discrete time IIR filter from analog
filter – IIR filter design by Impulse Invariance, Bilinear transformation,
Approximation of derivatives – (LPF, HPF, BPF, BRF) filter design using
frequency translation.
1. What is filter?
Filter is a frequency selective device ,which amplify particular range of
frequencies and attenuate particular range of frequencies.
2. What are the types of digital filter according to their impulse response?
IIR (Infinite impulse response )filter
FIR (Finite Impulse Response)filter.
3. Define IIR filter?
The filter designed by considering all the infinite samples of impulse
response are called IIR filter.
4. What do you understand by backward difference?
One of the simplest methods of converting analog to digital filter is
to approximate the differential equation by an equivalent difference
equation. d/dt(y(t)/t=nT=(y(nT)-y(nT-T))/T
5. What are the properties of chebyshev filter? NOV/DEC 2006
1. The magnitude response of the chebyshev filter exhibits ripple either in the stop
band or the pass band.
2. The poles of this filter lies on the ellipse
6. Give the equation for the order N, major, minor axis of an ellipse in case of
chebyshev filter?
The order is given by N=cosh-1(((10.1_p)-1/10.1_s-1)1/2))/cosh-1_s/_p
7. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the following methods
1. Approximation of derivatives
2. Impulse invariant method
3. Bilinear transformation
8. Write down bilinear transformation.
s=2/T (z-1/z+1)
9. List the Butterworth polynomial for various orders.
N Denominator polynomial
1 S+1
2 S2+.707s+1
3 (s+1)(s2+s+1)
4 (s2+.7653s+1)(s2+1.84s+1)
5 (s+1)(s2+.6183s+1)(s2+1.618s+1)
6 (s2+1.93s+1)(s2+.707s+1)(s2+.5s+1)
10. Differentiate Butterworth and Chebyshev filter. MAY/JUNE 2006
Butterworth damping factor 1.44 chebyshev 1.06
Butterworth flat response chebyshev damped response.
11. How phase distortion and delay distortion are introduced?
The phase distortion is introduced when the phase characteristics of a filter
is nonlinear with in the desired frequency band.
The delay distortion is introduced when the delay is not constant with in the
desired frequency band.
12. Distinguish IIR and FIR filters. NOV/DEC 2007
Impulse response is finite
They have perfect linear phase
Impulse Response is infinite
They do not have perfect linear phase
Non recursive.
Greater flexibility to control the shape of magnitude response
Less flexibility
13. Distinguish analog and digital filters
Constructed using active or passive components and it is described by a
differential equation Consists of elements like adder, subtractor and delay units and it is
described by a difference equation. Frequency response can be changed by changing the
Components Frequency response can be changed by changing the filter Coefficients It
processes and generates analog output Processes and generates digital output. Output
varies due to external conditions Not influenced by external conditions
14. Write the steps in designing chebyshev filter?
1. Find the order of the filter.
2. Find the value of major and minor axis.
3. Calculate the poles.
4. The numerator polynomial value depends on the value of
n. If n is odd: put s=0 in the denominator polynomial.
If n is even put s=0 and divide it by (1+e2)1/2
15. Write down the steps for designing a Butterworth filter?
1. From the given specifications find the order of the filter
2 find the transfer function from the value of N
3. Find c
4 find the transfer function ha(s) for the above value of c by by that value.
16. State the equation for finding the poles in chebyshev filter
sk=acos¢k+jbsin¢k,where ¢k=/2+(2k-1)/2n)
17. State the steps to design digital IIR filter using bilinear method
Substitute s by 2/T (z-1/z+1), where T=2/_ (tan (w/2) in h(s) to get h (z)
18. What is warping effect? NOV/DEC 2003
For smaller values of w there exist linear relationship between w and .but for
larger values of w the relationship is nonlinear. This introduces distortion in the frequency
axis. This effect compresses the magnitude and phase response. This effect is called
warping effect.
19. Write a note on pre warping. NOV/DEC 2008, MAY/JUNE 2009
The effect of the non linear compression at high frequencies can be
compensated. When the desired magnitude response is piecewise constant over
frequency, this compression can be compensated by introducing a suitable rescaling or
prewarping the critical frequencies.
20. Give the bilinear transform equation between s plane and z plane
s=2/T (z-1/z+1)
21. Why impulse invariant method is not preferred in the design of IIR filters
other than low pass filter?
In this method the mapping from s plane to z plane is many to one. Thus there are
an infinite number of poles that map to the same location in the z plane, producing an
aliasing effect. It is inappropriate in designing high pass filters. Therefore this method is
not much preferred.
22. What is meant by impulse invariant method? MAY/JUNE 2004
In this method of digitizing an analog filter, the impulse response of the resulting
digital filter is a sampled version of the impulse response of the analog filter. For e.g. if the
transfer function is of the form, 1/s-p, then
H (z) =1/1-e-pTz-1
23. Define IIR
This type of system has an impulse response of infinite time interval
24. Mention the features of IIR filters
No linear phase
Have desired characteristics for magnitude response only
25. Mention the advantages of digital filters
High thermal stability
Accurate
Easily programmable
Filtering is possible
UNIT III - FINITE IMPULSE RESPONSE DIGITAL FILTERS
Structures of FIR – Linear phase FIR filter – Fourier Series - Filter design
using windowing techniques (Rectangular Window, Hamming Window,
Hanning Window), Frequency sampling techniques – Finite word length
effects in digital Filters: Errors, Limit Cycle, Noise Power Spectrum.
1. What is mean by FIR filter?
The filter designed by selecting finite number of samples of impulse response h(n)
obtained from inverse Fourier transform of desired frequency response. H (w)) are called
FIR filters
2. Write the steps involved in FIR filter design
Choose the desired frequency response Hd (w)
Take the inverse Fourier transform and obtain Hd (n)
Convert the infinite duration sequence Hd (n) to h (n)
Take Z transform of h(n) to get H(Z)
3. What are advantages of FIR filter? (NOV/DEC 2004)
Linear phase FIR filter can be easily designed.
Efficient realization of FIR filter exists as both recursive and non-recursive
structures.
FIR filter realized non-recursively stable.
The round off noise can be made small in non recursive realization of FIR filter.
4. What are the disadvantages of FIR FILTER?(NOV/DEC 2006)
The duration of impulse response should be large to realize sharp cutoff filters. The
non integral delay can lead to problems in some signal processing applications.
5. What is the necessary and sufficient condition for the linear phase characteristic of
a FIR filter?
The phase function should be a linear function of w, which in turn requires constant
group delay and phase delay.
6. List the well-known design technique for linear phase FIR filter design?
Fourier series method and window method
Frequency sampling method.
Optimal filter design method.
7. For what kind of application, the symmetrical impulse response can be used?
The impulse response, which is symmetric having odd number of samples can be
used to design all types of filters, i.e. low pass, high pass, band pass and band reject.
The symmetric impulse response having even number of samples can be used to
design low pass and band pass filter.
8. What is the reason that FIR filter is always stable?
FIR filter is always stable because all its poles are at the origin.
9. What condition on the FIR sequence h(n) are to be imposed n order that this filter
can be called a liner phase filter? NOV/DEC 2005
The conditions are
(i) Symmetric condition h(n)=h(N-1-n)
(ii) Antisymmetric condition h(n)=-h(N-1-n)
12. What are the disadvantages of Fourier series method?
In designing FIR filter using Fourier series method the infinite duration impulse
response is truncated at n= (N-1/2).Direct truncation of the series will lead to fixed
percentage overshoots and undershoots before and after an approximated discontinuity in
the frequency response .
13. What is Gibbs phenomenon?
One possible way of finding an FIR filter that approximates H(ej)would be to truncate
the infinite Fourier series at n= (N-1/2).Abrupt truncation of the series will lead to
oscillation both in pass band and is stop band .This phenomenon is known as Gibbs
phenomenon.
14. What are the desirable characteristics of the windows?(APRIL/MAY 2007)
The desirable characteristics of the window are
1. The central lobe of the frequency response of the window should contain most of the
energy and should be narrow.
2. The highest side lobe level of the frequency response should be small.
3. The side’s lobes of the frequency response should decrease in energy rapidly as w tends
to π.
15. Compare Hamming window with Kaiser Window.
Hamming window Kaiser Window
1. The main lobe width is equal to8π/N and the peak side lobe level is –41dB.
2. The low pass FIR filter designed will have first side lobe peak of –53 dB. The
main lobe width, the peak side lobe level can be varied by varying the parameter π
and N.
3. The side lobe peak can be varied by varying the parameter π.
16. What is the necessary and sufficient condition for linear phase characteristics in
FIR filter?
The necessary and sufficient condition for linear phase characteristics in FIR filter is
the impulse response h(n) of the system should have the symmetry property,i.e,
H (n) = h(N-1-n)
where N is the duration of the sequence.
17. What is the advantage of Kaiser Widow?
1. It provides flexibility for the designer to select the side lobe level and N .
2. It has the attractive property that the side lobe level can be varied Continuously
from the low value in the Blackman window to the high value in the rectangle
window.
18. What is the principle of designing FIR filter using frequency sampling method?
In frequency sampling method the desired magnitude response is sampled and a linear
phase response is specified .The samples of desired frequency response are defined as DFT
coefficients. The filter coefficients are then determined as the IDFT of this set of samples.
19. what type of filters frequency sampling method is suitable? (NOV/DEC 2005)
Frequency sampling method is attractive for narrow band frequency selective
filters where only a few of the samples of the frequency response are non-zero.
20. What are the requirements of analog filters.
Transfer function should be rational function
Coefficient should be real
Poles in the left half
21. Mention the disadvantages of digital filters
Bandwidth is limited
Performance depend on hardware.
22. What do you understand by backward difference?
One of the simplest methods of converting analog to digital filter is to
approximate the differential equation by an equivalent difference equation.
d/dt(y(t)/t=nT=(y(nT)-y(nT-T))/T
23. Mention the techniques for digitizing analog filter
Bilinear transformation
Impulse invariant
24. Describe the features of IIR filters.
Physically realizable
Desired characteristics for magnitude
25. Compare Rectangular and Hanning windows.
Rectangular window Hanning window
(a) The width of the mainlobe in window (a) The width of the mainlobe in window
spectrum is 4Π/N. spectrum is 8Π/N.
(b) The maximum sidelobe magnitude in (b) The maximum sidelobe magnitude in
window spectrum is -13dB. window spectrum is -31dB.
(c) In window spectrum the sidelobe (c) In window spectrum the sidelobe
magnitude slightly decreases with magnitude decreases with increasing ω.
increasing ω.
(d) The minimum stop band attenuation (d) The minimum stop band attenuation
is 22 dB. is 44 dB.
26. How phase and delay distortions are introduced?
The phase distortion is introduced when the phase characteristics of a filter
is not linear within the desired frequency band.
The delay distortion is introduced when the delay is not a constant within
the desired frequency.
27. Compare Hanning and Hamming windows.
Hanning window Hamming window
(a) The width of the mainlobe in window (a) The width of the mainlobe in window
spectrum is 8Π/N. spectrum is 8Π/N.
(b) The maximum sidelobe magnitude in (b) The maximum sidelobe magnitude in
window spectrum is -31dB. window spectrum is -41dB.
(c) In window spectrum the sidelobe (c) In window spectrum the sidelobe
magnitude slightly decreases with magnitude remains constant.
increasing ω.
(d) The minimum stop band attenuation (d) The minimum stop band attenuation
is 44 dB. is 51 dB.
28. Compare Hamming and Blackman windows. APRIL/MAY 2007
Hamming window Blackman window
(a) The width of the mainlobe in window (a) The width of the mainlobe in window
spectrum is 8Π/N. spectrum is 12Π/N.
(b) The maximum sidelobe magnitude in (b) The maximum sidelobe magnitude in
window spectrum is -41dB. window spectrum is -58dB.
(c) In window spectrum the sidelobe (c) In window spectrum the sidelobe
magnitude remains constant. magnitude slightly decreases with
increasing ω.
(d) The minimum stop band attenuation (d) The minimum stop band attenuation
is 51 dB. is 78 dB.
(e) The higher value of sidelobe (e) The higher value of sidelobe
attenuation is achieved at the expense of attenuation is achieved at the expense of
constant attenuation at higher frequencies. increased mainlobe width.
29. State the condition for a digital filter to be causal and stable?
A digital filter is causal if its impluse response h(n)=0 for n<0.
A digital filter is stable if its impulse response is absolutely summable ,i.e,
30. What are the features of FIR filter?
1. FIR filter is always stable.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response
UNIT IV FINITE WORD LENGTH EFFECTS
Fixed point and floating point number representations – ADC –Quantization-
Truncation and Rounding errors - Quantization noise – coefficient quantization
error – Product quantization error - Overflow error – Roundoff noise power -
limit cycle oscillations due to product round off and overflow errors – Principle
of scaling
1. What do you understand by a fixed-point number? MAY/JUNE 2004
In fixed point arithmetic the position of the binary point is fixed. The bits to the right
represent the fractional part of the number & those to the left represent the integer part.
For example, the binary number 01.1100 has the value 1.75 in
decimal.
3. What is meant by block floating point representation? What are its advantages?
In block point arithmetic the set of signals to be handled is divided into blocks. Each
block has the same value for the exponent. The arithmetic operations with in the block uses
fixed point arithmetic & only one exponent per block is stored thus saving memory. This
representation of numbers is more suitable in certain FFT flow graph & in digital audio
applications.
4. What are the advantages of floating point arithmetic?
1. Large dynamic range
2. Over flow in floating point representation is unlike.
5. What are the three-quantization errors to finite word length registers in digital
filters?
1. Input quantization error
2. Coefficient quantization error
3. Product quantization error
6. What do you understand by input quantization error?
In digital signal processing, the continuous time input signals are converted into
digital using a b-bit ACD. The representation of continuous signal amplitude by a fixed
digit produce an error, which is known as input quantization error.
7. How the multiplication & addition are carried out in floating point arithmetic?
In floating point arithmetic, multiplication are carried out as follows, Let
f1 = M1*2c1 and f2 = M2*2c2. Then f3 = f1*f2 = (M1*M2) 2(c1+c2)
That is, mantissa is multiplied using fixed-point arithmetic and the exponents are
added. The sum of two floating-point number is carried out by shifting the bits of the
mantissa of the smaller number to the right until the exponents of the two numbers are
equal and then adding the mantissas.
8. What is the relationship between truncation error e and the bits b for representing
a decimal into binary?
For a 2's complement representation, the error due to truncation for both positive
and negative values of x is 0>=xt-x >-2-b
Where b is the number of bits and xt is the truncated value of x. The equation
holds good for both sign magnitude, 1's complement if x>0 If x<0, then for sign
magnitude and for 1's complement the truncation error satisfies.
9. What is meant rounding?
Rounding a number to b bits is accomplished by choosing the rounded result as the b bit
number closest to the original number unrounded.
10. What is meant by A/D conversion noise?
A DSP contains a device, A/D converter that operates on the analog input x(t) to
produce xq(t) which is binary sequence of 0s and 1s.
At first the signal x(t) is sampled at regular intervals to produce a sequence x(n) is
of infinite precision. Each sample x(n) is expressed in terms of a finite number of bits
given the sequence xq(n). The difference signal e(n)=xq(n)-x(n) is called A/D conversion
noise.
11. What is the effect of quantization on pole location?NOV/DEC 2004
Quantization of coefficients in digital filters lead to slight changes in their value.
This change in value of filter coefficients modifies the pole-zero locations. Some times the
pole locations will be changed in such a way that the system may drive into instability.
12. What is meant by quantization step size? NOV/DEC 2006, 2008
Let us assume a sinusoidal signal varying between +1 and -1 having a dynamic range 2. If
the ADC used to convert the sinusoidal signal employs b+1 bits including sign bit, the
number of levels available for quantizing x(n) is 2b+1. Thus the interval between
successive levels q= 2 =2-b -------- 2b+1
Where q is known as quantization step size.
13. How would you relate the steady-state noise power due to quantization and the b
bits representing the binary sequence?
Steady state noise power
Where b is the number of bits excluding sign bit.
15. What is overflow oscillation?
The addition of two fixed-point arithmetic numbers cause over flow the sum
exceeds the word size available to store the sum. This overflow caused by adder make the
filter output to oscillate between maximum amplitude limits. Such limit cycles have been
referred to as over flow oscillations
16. What are the methods used to prevent overflow?
There are two methods used to prevent overflow
1. Saturation arithmetic
2. Scaling
17. What are the two kinds of limit cycle behavior in DSP?
1. Zero input limit cycle oscillations
2. Overflow limit cycle oscillations
18. What is meant by "dead band" of the filter?
The limit cycle occur as a result of quantization effect in multiplication. The
amplitudes of the output during a limit cycle are confined to a range of values called the
dead band of the filter.
19. Explain briefly the need for scaling in the digital filter implementation.
To prevent overflow, the signal level at certain points in the digital filter must be
scaled so that no overflow occurs in the adder.
20. What is meant by autocorrelation?
The autocorrelation of a sequence is the correlation of a sequence with its shifted
version, and this indicates how fast the signal changes.
21. Define noise transfer function (NTF)?
The NTF is defined as the transfer function from the noise source to the filter output.
The NTF depends on the structure of the digital network.
22. What are the two types of quantization employed in digital system?
The two types of quantization in digital system are truncation and rounding.
24. What is truncation?
The process of reducing the size of binary number by discarding all bits less
significant than the least significant bit that is retained.
25. What is the drawback in saturation arithmetic?
The saturation arithmetic introduces non-linearity in the adder which creates signal
distortion.
26. What are the types of arithmetic used in digital computers?
The floating point arithmetic and two’s complement arithmetic are the two types.
UNIT-V - DSP APPLICATIONS
Multirate signal processing: Decimation, Interpolation, Sampling rate
conversion by a rational factor – Adaptive Filters: Introduction, Applications
of adaptive filtering to equalization.
1. Define sampling rate conversion.
Sampling rate conversion of digital signal can be obtainedby converting the digital
signal
into analog form (D/A) and then resampling the resulting (A/D) conversion.
2. Define decimation.
The process of reducing the sampling rate by a factor D is known as
decimation or down sampling.
3. What is interpolator?
Interpolator is also known as upsampler.
The process of sampling rate conversion in digital domain can be viewed as
linear filtering operation.
4. What is a decimator?
Decimator is also known as downsampler. It reduces sampling rate.
5. Mention two applications of multirate signal processing.
Phase shifter
Transmultiplexers
Vocoder
6. What do you mean by sub band coding?
Sub band coding is a method where speech signal is sub-divided into several
frequency bands and each band is digitally encoded and separated by
allocating different bits per sample to the signal of different sub-bands.we can
achieve a reduction in a bit rate of the digitalized bit signal. It is very useful to
reproduce speech signal efficiently.
7. What is an anti-imaging filter?
The filter which is used to remove the image spectra is known as anti-
imaging filter.
8. What is transmultiplexers?
Transmultiplexers is used to convert frequency division multiplexed signals
into time division multiplexed signals and vice versa.
9. Mention the two stages of musical sound processing.
First stage: the sound from the singer or the sound from the instrument is recorded
on a single track of multitrack tape.
Second stage: the special audio effects are added to this sound. The special
audio effects can be generated by DSP.
10. What are the steps to be followed for the reproduction of the recorded
signal?
Decoding and demodulation
Error correction and demultiplexin.
11. Define interpolation.
The process of increasing the sampling rate by a factor I is known as interpolation.
12. What is linear filter?
Linear filter process time varying input signals to produce output signals subject to
the constraint of linearity. It is also used in statistics and data analysis.
13. Define multirate signal processing systems.
The systems that employ multiple sampling rates in the processing of digital
signals are known as multi rate signal processing.
14. What are the two methods used for sampling rate conversion?
First method:
The digital signal is converted into analog signal by using DAC. Then analog
signal is converted into digital signal using ADC.
Second method:
sampling rate conversion is performed in digital domain.
15. Define aliasing.
Aliasing refers to an effect that causes different signals to become
indistinguishable when sampled. It also refers to the distortion when the signal
reconstructed from samples is different from the original continuous.
16. Define Adaptive Filter.
An adaptive filter is a system with a linear filter that has a transfer function
controlled by variable parameters and a means to adjust those parameters according
to an optimization algorithm. Because of the complexity of the optimization
algorithms, most adaptive filters are digital
