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TWO MARKS
UNIT I
2. Define IDFT.
The IDFT is used to convert the N point frequency domain sequence X (K) to an
N point time sequence. The IDFT of the sequence X (K) of length N is defined as
N-1
x (n) =1/N X (K) e+j2 nk/N for n=0, 1,2,N-1
K=0
Let DFT{x (n)} =X (K), DFT{x1 (n)} =X1 (K), DFT{x2 (n)} =X 2(K)
6. What is FFT?
The FFT (Fast Fourier Transform) is a method for computing the DFT with
reduced number of calculations. The computational efficiency is achieved by employing
divide and conquers approach. This is based on the decomposition of an N point DFT
into successively smaller DFT.
In an N point sequence if N can be expressed as N= r m then the sequence can be
decimated into r- point sequences. For each r- point sequence, r point DFT is computed.
From the results of r- point DFTs the r 2 - point DFTs are computed. From the results of
r 2- point DFT the r3- point DFT are computed and so on, until we get r m point DFT.
Hence the number of stage of computation is m. The number r is called the radix of the
FFT algorithm.
WN=e-j2 /N
11. Draw and explain the basic butterfly diagram (or) flow graph of DIT radix-2
FFT?
The basic butterfly diagram of DIT radix-2 FFT is shown below. It performs the
following operations .Here a & b are the input complex number and A&B are output
complex number.
i) Input complex number b is multiplied by the phase factor WN K.
ii) The product bWNK is added to the input complex number a to from a new
complex number A.
iii) The product bWNK is subtracted to the input complex number a to from a new
complex number B.
14. Draw and explain the basic butterfly diagram or flow graph of dif radix2 FFT?
The basic butterfly diagram of DIF radix2 FFT is shown below. It performs the
following operations .Here a & b are the input complex number and A&B are output
complex number.
i) The sum of the complex number a and b are computed to form a new complex
number A.
ii) The complex number b is subtracted from a to get the difference a-b.
iii) The difference term a-b is multiplied with phase factor WNK to from a new complex
number B.
17. Calculate the number of multiplications needed in calculation of DFT and FFT
with 64 point Sequence.
DFT
The number of complex multiplications required using direct computation is
N2=642=4096
FFT
The number of complex multiplication required using FFT is
N/2logN=64/2log64=192
18. Calculate the number of addition needed in calculation of DFT and FFT with
16 point Sequence.
DFT
The number of complex addition required using direct computation is
N (N-1) =16(16-1) =240
FFT
The number of complex addition required using FFT is
N log N=16log16=64
N-1
X (K) = (n) e-j2 nk/N for K=0, 1, 2, N-1
n=0
=1
1. What are the types of digital filter according to their impulse response?
4. What is the necessary and sufficient condition for the linear phase characteristic
of a FIR filter?
They have perfect linear phase They do not have perfect linear
phase
Non recursive Recursive
6. List the well known design technique for linear phase FIR filter design?
One possible way of finding an FIR filter that approximates H d(e j)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.
9. Under what conditions a finite duration sequence h(n) will yield constant group
delay in its frequency response characteristics and not the phase delay?
The frequency response of FIR filter will have constant group delay and not the
phase delay .
WR (n) = {1 -(N-1)/2n(N-1)/2
0 otherwise
1.The main lobe width is equal to 8/N 1.The main lobe width is equal to 4/N
and the peak side lobe level is 41dB. and the peak side lobe level is 13dB.
2.The low pass FIR filter designed will 2.The low pass FIR filter designed will
have minimum stop band attenuation have minimum stop band attenuation
of 53 dB of 21dB
1.The main lobe width is equal to 8/N The main lobe width, the peak side lobe
and the peak side lobe level is 41dB. level can be varied by varying the
parameter and N.
2.The low pass FIR filter designed will The side lobe peak can be varied by
have first side lobe peak of 53 dB varying the parameter .
FIR filter is always stable because all its poles are at the origin.
16. What condition on the FIR sequence h (n) are to be imposed N order that this
filter can be called a liner phase filter?
The conditions are
(i) Symmetric condition h(n)=h(N-1-n)
(ii) Anti symmetric condition h(n)=-h(N-1-n)
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?
To design a filter means to select the coefficients such that the system has specific
characteristics. The required characteristics are stated in filter specifications. Most of the
time filter specifications refer to the frequency response of the filter. There are different
methods to find the coefficients from frequency specifications:
FIR filters are one of two primary types of digital filters used in Digital Signal Processing
(DSP) applications (the other type being IIR).
The impulse response is "finite" because there is no feedback in the filter; if you put in an
impulse (that is, a single "1" sample followed by many "0" samples), zeroes will
eventually come out after the "1" sample has made its way in the delay line past all the
coefficients.
Some people say the letters F-I-R; other people pronounce as if it were a type of tree. We
prefer the tree. (The difference is whether you talk about an F-I-R filter or a FIR filter.)
25. What is the alternative to FIR filters?
DSP filters can also be "Infinite Impulse Response" (IIR). (See dspGuru's IIR FAQ.) IIR
filters use feedback, so when you input an impulse the output theoretically rings
indefinitely.
Each has advantages and disadvantages. Overall, though, the advantages of FIR filters
outweigh the disadvantages, so they are used much more than IIRs.
UNIT III
1.. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the following methods
1. Impulse invariant method
2. Bilinear transformation
2. 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).
s=2/T (1-z-1/1-z+1)
The relation between the analog and digital frequencies in bilinear transformation
is given by
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
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 by using the formula
7. 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. i.e. ,all the
poles in the s plane between the intervals (2k-1)/T to (2k+1)/T .Thus there are an
infinite number of poles that map to the same location in the z plane, producing an
aliasing effect. Due to spectrum aliasing the impulse invariant method is inappropriate in
designing high pass filters. That is why the impulse method is not much preferred. in the
design of IIR filters other than low pass filter
8. By impulse invariant method obtain the digital filter transfer function and the
differential equation of the analog filter H(s) =1/s+1
S- pK=1-epKTz-1
S+1=0
S= -1= pK
Substitude the value of pK
H (z) =1/1-e-Tz-1
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,
N
H(S)= 1/s-pk, then
K=1
N
H (z) =1/1-epKTz-1
K=1
10. Give the magnitude function of Butterworth filter. What is the effect of varying
order N on magnitude and phase response?
|H(j)=1/[1+(/c)2N]1/2 ,N=1,2,3,4,.
Where N is the order of the filter and c is the cutoff frequency. The magnitude
response of Butterworth filter closely approximates the ideal response as the order N
increases. The phase response becomes more non linear as N increases.
Where
k= p/ s
= (100.1s-1)0.5
= (100.1p-1)0.5
IIR filters are one of two primary types of digital filters used in Digital Signal Processing
(DSP) applications (the other type being FIR). "IIR" means "Infinite Impulse Response".
The impulse response is "infinite" because there is feedback in the filter; if you put in an
impulse (a single "1" sample followed by many "0" samples), an infinite number of non-
zero values will come out (theoretically).
DSP filters can also be "Finite Impulse Response" (FIR). FIR filters do not use feedback,
so for a FIR filter with N coefficients, the output always becomes zero after putting in N
samples of an impulse response.
16. What are the advantages of IIR filters (compared to FIR filters)?
IIR filters can achieve a given filtering characteristic using less memory and calculations
than a similar FIR filter.
17. What are the disadvantages of IIR filters (compared to FIR filters)?
They are more susceptable to problems of finite-length arithmetic, such as
noise generated by calculations, and limit cycles. (This is a direct
consequence of feedback: when the output isn't computed perfectly and is
fed back, the imperfection can compound.)
They are harder (slower) to implement using fixed-point arithmetic.
They don't offer the computational advantages of FIR filters for multirate
(decimation and interpolation) applications.
- no group delay ripple, no gain ripple in both bands, slow gain cutoff
First, if your filter has zero-valued coefficients, you don't actually have to calculate those
taps; you can leave them out. A common case of this is "half-band" filter, which have the
property that every-other coefficient is zero.
Second, if your filter is "symmetric" (linear phase), you can "pre-add" the samples which
will be multiplied by the same coefficient value, prior to doing the multiply. Since this
technique essentially trades an add for a multiply, it isn't really useful in DSP
microprocessors which can do a multiply in a single instruction cycle. However, it is
useful in ASIC implementations (in which addition is usually much less expensive than
multiplication); also, some newer DSP processors now offer special hardware and
instructions to make use of this trick.
The spectrum of an ideal impulse is flat and so the frequency shape of the resulting
output of the network is the frequency response of the network
UNIT IV
1. What are the different types of arithmetic in digital systems?
There are three types of arithmetic used in digital system
1. Fixed point arithmetic
2. Floating point arithmetic
3. Block floating arithmetic.
(-0.5625)10 = 1.011011
0.000001
-------------
(1.011100)2
5. 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.
8. 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 numbers 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.
13. What is the relationship between truncation error e(n) 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
-2-b e (n) 0
Where b is the number of bits
14. What is meant rounding? Discuss its effect on all types of number
representation?
Rounding a number to b bits is accomplished by choosing the rounded result as the b
bit number closest to the original number unrounded.
For fixed point arithmetic, the error made by rounding a number to b bits satisfy the
inequality
-2-b 2-b
----- e (n) --------
2 2
for all three types of number systems, i.e., 2's complement, 1's complement & sign
magnitude.
For floating point number the error made by rounding a number to b bits satisfy the
inequality
-2-be (n) 2-b where e (n) =xT-x
q = 2 / 2 b+1 =2-b
20. How would you relate the steady-state noise power due to quantization and the b
bits representing the binary sequence?
Steady state noise power
e 2 =2-2b/12
Where b is the number of bits excluding sign bit.
21. What are the two kinds of limit cycle behavior in DSP?
1. Zero input limit cycle oscillations
2. Overflow limit cycle oscillations
26. 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.
27. Why the limit cycle problem does not exist when FIR digital filter is realized in
direct form or cascade form?
In case of FIR filter there are no limit cycle oscillations, if the filter is realized in
direct form or cascade form since these structures have no feedback.
29. What is the steady state variance of the noise in the output due to quantization of
the input for the first order filter?
y(n)=ay(n-1)+x(n)
e 2 =(2-2b/12)(1/1-a2)
31. Express the fraction 7/8 and -7/8 in sign magnitude, 2s complement and 1s
complement.
Fraction (7/8) = (0.111)2 in sign magnitude, 2s complement and 1s complement.
32. Give the expression for signal to quantization noise ratio and calculate the
improvement with an increase of 2 bit to the existing bit.
SNR=6.02b+10.79+10log10 x 2
With an increase of 2 bits, increase in SNR is approximately 12 Db
UNIT V
1. What is the energy density spectrum?
Sxx(F)=|X(F)|2
Sxx(f)= | x(n)e-j2Kf |2
n= -
6. Define Periodogram.
The effect of reducing the length of data from N point to M=N/K, result in a
window whose spectral width has been increased by a factor of K. consequently, the
frequency resolution has been reduced by a factor K. in return in resolution we reduced
the variance.
i) First difference is that welchs method allows overlapping of data sequence. The
overlap 50% or 75%.
ii) The second difference is the data with in a sequence are windowed prior to computing
the periodogram.
Qa=E[Pxx(f)]2/var[Pxx(f)]
Quality factor=10
Length of sample sequence (N) =1000
Overlap in Welch method=50%
Bartlett method
QB=1.11Nf
f= (Qbart/1.11N)=0.009
Welch method
Qw=1.39Nf
f=0.0072
Blackman-turkey
QBT=2.34 N f
f=0.0042
16. Limitation of non parametric methods for power spectrum estimation? (What
are the disadvantages of non-parametric methods of power spectral estimation?)
i) It requires long data sequence to obtain the necessary frequency resolution.
ii) Spectral leakage effect because of windowing.
iii)The assumption of the autocorrelation estimator x(m) to be zero for mN. This
assumption limits the frequency resolution and quality of power spectrum estimate.
iv)Assumption that the data are periodic with period N. These assumptions may not be
realistic.
i) Bias
ii) Variance
Bias of estimator is defined as the true value of the parameter minus the expected
value of the estimator.
An unbiased estimator is one for which the bias is 0. This then means that the
expected value of the estimator is the true value so that the probability density is
symmetrical then its center would be at the true value.
Deterministic signals are functions that are completely specified in time. The
nature and amplitude of such signal at any time can be predicted. The Patten of the signal
is a regular and can be characteristics mathematically.
Examples
i) x (t)=ct this is the ramp whose amplitude of this signal increases linearly
with time and slope is c.
A non deterministic signal is one whose occurrence is random in nature and its
Patten is quite irregular.
Example:
i) A typical example of a non deterministic signal is thermal noise in an electrical
circuit.
26. What is zero padding? Does zero padding improve the frequency resolution in
Spectral estimate?
Zero padding is increase the length of sequence by adding zero to the given
sequence.
Note that zero padding does not change the resolution but it does have the effect
of interpolating the spectrum pxx(f)