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Anna University, Chennai.
Prepared by,
Prof. U. Vinothkumar, AP/ECE/Dr.N.G.P.IT
UNIT - 1
DISCRETE FOURIER TRANSFORM
Syllabus:
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.
Two mark questions:
1. Define Signal.
Signal is a physical quantity that varies with respect to time, space or any
other independent variable.
(Or)
It is a mathematical representation of the system
Eg y(t) = t. and x(t)= sin t.
2. Define system.
(NOV-2004)
A set of components that are connected together to perform the particular
task. E.g. Filters
(Or)
A System is defined as a physical device that generates a response or an output
signal, for a given input signal.
3. State the classification of discrete time signals.
(APR-2006)
(APR-2008)
The signal that are defined at discrete instants of time are known as discretetime signals. The discrete-time signals are continuous in amplitude and discrete in
time. They are denoted by x(n).
7. Give some applications of DSP?
A continuous time signal can be represented in its samples and recovered back
if the sampling frequency Fs 2B. Here Fs is the sampling frequency and B is
the maximum frequency present in the signal.
9. What are the properties of convolution?
(APR-2006)
It is a finite duration discrete frequency sequence, which is obtained by
sampling one period of Fourier transform. Sampling is done at N equally spaced
points over the period extending from w=0 to 2.
DFT is defined as X(w)= x(n)e-jwn. Here x(n) is the discrete time sequence
X(w) is the fourier transform of x(n).
11. Define Twiddle factor.
The method of appending zero in the given sequence is called as Zero padding.
(DEC-2006)
Consider the complex valued sequences x(n) and y(n).If x(n)y*(n)=1/N
X(k)Y*(k)
15. List the properties of DFT.
computation?
Number of arithmetic operations involved in the computation of DFT is
greatly reduced by using different FFT algorithms as follows.
1. Radix-2 FFT algorithms. -Radix-2 Decimation in Time (DIT) algorithm. Radix-2 Decimation in Frequency (DIF) algorithm.
2. Radix-4 FFT algorithm.
18. What is the computational complexity using FFT algorithm?
(DEC-2006)
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. Initially
the N-point sequence is divided into two N/2-point sequences xe(n) and x0(n), which
have the even and odd members of x(n) respectively. The N/2 point DFTs of these
two sequences are evaluated and combined to give the N point DFT. Similarly the
N/2 point DFTs can be expressed as a combination of N/4 point DFTs. This process
is continued till we left with 2-point DFT. This algorithm is called Decimation-intime because the sequence x(n) is often splitted into smaller sub sequences.
21. What are the differences and similarities between DIF and DIT algorithms?
Differences: 1. For DIT, the input is bit reversal while the output is in natural
order, whereas for DIF, the input is in natural order while the output is bit reversed.
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. Bot algorithms can be done in place and both need to perform bit reversal
at some place during the computation.
22. What are the applications of FFT algorithms?
1. Linear filtering
2. Correlation
3. Spectrum analysis
23. What is a decimation-in-frequency algorithm?
In this the output sequence X (K) is divided into two N/2 point sequences and
each N/2 point sequences are in turn divided into two N/4 point sequences.
24. Distinguish between DFT and DTFT.
(NOV-2008)
S.No.
DFT
DTFT
1.
Obtained by performing sampling Sampling is performed only in time
operation in both the time and domain.
frequency domains.
2.
Continuous function of
(NOV-2008)
( )
order.
The x(n) in bit reversed order= {1,-1,3,-3,2,-2,4,-4}.
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
UNIT - 2
IIR Filter Design
Syllabus:
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.
Two mark questions:
1. Define IIR filter?
IIR filter has Infinite Impulse Response.
(NOV-2010)
iv)
(ARP-2008)
Analog filter
i)
ii)
iii)
iv)
10
18. What are the requirements for an analog filters to be stable and causal?
i.
The analog filter transfer function H(s) should be a rational function of s
and the coefficients of s should be real.
ii.
The poles should lie on the left half of s-plane.
iii.
The number of zeros should be less than or equal to number of poles.
19. Distinguish between IIR and FIR filters.
(DEC-2004)
The filter design starts from ideal frequency response. By taking inverse
fourier transform of ideal frequency response, the desired impulse response is
obtained, which consists of infinite number of samples.
The digital filter design by selecting only N samples of the impulse response
are called FIR filters. The digital filters designed by considering all the infinite
samples of impulse response are called IIR filters.
20. Compare IIR and FIR filters.
IIR Filter
i.
All the infinite samples of
impulse
response
are
considered.
ii.
The impulse response cannot
be directly converted to digital
filter transfer function.
iii.
The design involves design of
analog filter and then
transforming analog filter to
digital filter.
iv.
The specifications include the
desired characteristics for
magnitude response only.
v.
Linear phase characteristics
cannot be achieved.
i.
ii.
iii.
iv.
v.
FIR Filter
Only N samples of impulse
response are considered.
The impulse response can be
directly converted to digital
filter transfer function.
The digital filter can be
directly designed to achieve
the desired specification.
The specifications include the
desired characteristics for
both magnitude and phase
response.
Linear phase filter can be
easily designed.
11
22. Mention any two techniques for digitizing the transfer function of an analog
filter.
The bilinear transformation and the impulse invariant transformation are the
two techniques available for digitizing the analog filter transfer function.
23. What are the properties that are maintained same in the transformation of
analog to digital filer?
The analog filter should be stable and causal for effective transformation to
digital filters. While transforming the analog filer to digital filters these two
properties (i.e. stability and causality) are maintained same, which means that the
transformed digital filer should also be stable and causal.
24. What is aliasing?
(NOV-2010)
The phenomena of high frequency sinusoidal components acquiring the
identity of low frequency sinusoidal components after sampling is called aliasing.
The aliasing problem will arise if the sampling rate does not satisfy the Nyquist
sampling criteria.
25. What is frequency warping?
In bilinear transformation the relation between analog and digital frequencies
is non-linear. When the s-plane is mapped in to z-plane using bilinear
transformation, this non-linear relationship introduce distortion in frequency axis,
which called frequency warping.
26. What is butterworth approximation?
(NOV-2009)
In butterworth approximation, the error function is selected such that the
magnitude is maximally flat in the origin (i.e., at = 0) and monotonically
decreasing with increasing .
27. How the poles of butterworth transfer function are located in s-plane?
The poles of the normalized butterworth transfer function symmetrically lies
on a unit circle in s-plane with angular spacing of /N.
28. What is the properties of butterworth filter?
i.
The butterworth filters are pole design.
ii.
At the cutoff frequency c, the magnitude of normalized butterworth filter
is 1/2.
iii.
The filter order N, completely specifies the filter and as the value of N
increases the magnitude response approaches the ideal response.
12
iv.
13
UNIT - 3
FIR Filter Design
Syllabus:
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.
Two mark questions:
1. What are FIR filters?
(DEC-2008)
The specifications of the desired filter will be given in terms of ideal
frequency response Hd(w). The impulse response hd(n) of the desired filter can be
obtained by inverse fourier transform of Hd(w), which consists of infinite samples.
The filters designed by selecting finite number of samples of impulse response are
called FIR filters.
2. What are the different types of filters based on impulse response?
Based on impulse response the filters are of two types 1. IIR filter 2. FIR filter
The IIR filters are of recursive type, whereby the present output sample
depends on the present input, past input samples and output samples.
The FIR filters are of non- recursive type, whereby the present output
sample depends on the present input, and previous output samples.
3. What are the different types of filter based on frequency response?
The filters can be classified based on frequency response. They are,
i)
Low pass filter
ii)
High pass filter
iii) Band pass filter
iv) Band reject filter.
4. What are the techniques of designing FIR filters?
(NOV-2004)
There are three well-known methods for designing FIR filters with linear
phase. These are 1) windows method 2) Frequency sampling method 3) Optimal or
mini-max design.
5. What is the reason that FIR filter is always stable?
FIR filter is always stable because all its poles are at origin.
14
(NOV-2011)
15
12. What are the conditions to be satisfied for constant phase delay in linear
phase FIR filters?
(APR-2009)
The conditions for constant phase delay are
Phase delay, = (N-1)/2 (i.e., phase delay is constant)
Impulse response, h(n) = -h(N-1-n) (i.e., impulse response is
antisymmetric)
13. How constant group delay & phase delay is achieved in linear phase FIR
filters?
The following conditions have to be satisfied to achieve constant group delay
& phase delay. Phase delay, = (N-1)/2 (i.e., phase delay is constant) Group delay,
= /2 (i.e., group delay is constant) Impulse response, h(n) = -h(N-1-n) (i.e.,
impulse response is antisymmetric)
14. What are the possible types of impulse response for linear phase FIR filters?
There are four types of impulse response for linear phase FIR filters
Symmetric impulse response when N is odd.
Symmetric impulse response when N is even.
Antisymmetric impulse response when N is odd.
Antisymmetric impulse response when N is even.
15. List the well-known design techniques of linear phase FIR filters.
There are three well-known design techniques of linear phase FIR filters. They
are
Fourier series method and window method
Frequency sampling method.
Optimal filter design methods.
16. What are the desirable characteristics of the frequency response of window
function?
The desirable characteristics of the frequency response of window function are
The width of the main lobe should be small and it should contain as much
of the total energy as possible.
The side lobes should decrease in energy rapidly as w tends to .
16
17
18
Hd(e )= hd(n)e-jn
n= -
If h(n) is absolutely summable (i.e., Bounded Input Bounded Output Stable). So, it
is in stable.
j
19
UNIT - 4
FINITE WORDLENGTH EFFECTS
Syllabus:
Fixed point and floating point number representations ADC QuantizationTruncation and Rounding errors -Quantization noise coefficient quantization
error Product quantization error - Overflow error Round-off noise power - limit
cycle oscillations due to product round off and overflow errors Principle of scaling
Two mark questions:
1. What do finite word length effects mean?
(DEC-2008)
The effects due to finite precision representation of numbers in a digital
system are called finite word length effects.
2. List some of the finite word length effects in digital filters.
Errors due to quantization of input data.
Errors due to quantization of filter co-efficient
Errors due to rounding the product in multiplications
Limit cycles due to product quantization and overflow in addition.
3. What are the different formats of fixed-point representation? (APR-2004)
Sign magnitude format
Ones Complement format
Twos Complement format.
In all the three formats, the positive number is same but they differ only in
representing negative numbers.
4. Explain the floating-point representation of binary number.
The floating-point number will have a mantissa part. In a given word size the
bits allotted for mantissa and exponent are fixed. The mantissa is used to represent
a binary fraction number and the exponent is a positive or negative binary integer.
The value of the exponent can be adjusted to move the position of binary point in
mantissa. Hence this representation is called floating point.
5. What are the types of arithmetic used in digital computers?
The floating point arithmetic and twos complement arithmetic are the two
types of arithmetic employed in digital systems.
20
6. What is truncation?
The truncation is the process of reducing the size of binary number by
discarding all bits less significant than the least significant bit that is retained. In
truncation of a binary number of b bits all the less significant bits beyond bth bit are
discarded.
7. What is rounding?
(DEC-2009)
Rounding is the process of reducing the size of a binary number to finite word
size of b-bits such that, the rounded b-bit number is closest to the original unquantized number.
8. Explain the process of upward rounding?
In upward rounding of a number of b-bits, first the number is truncated to bbits by retaining the most significant b-bits. If the bit next to the least significant bit
that is retained is zero, then zero is added to the least significant bit of the truncated
number. If the bit next to the least significant bit that is retained is one then one is
added to the least significant bit of the truncated number.
9. What are the errors generated by A/D process?
(APR-2008)
The A/D process generates two types of errors. They are quantization error
and saturation error. The quantization error is due to representation of the sampled
signal by a fixed number of digital levels. The saturation errors occur when the
analog signal exceeds the dynamic range of A/D converter.
10. What is quantization step size?
In digital systems, the numbers are represented in binary. With b-bit binary
we can generate 2b different binary codes. Any range of analog value to be
represented in binary should be divided into 2b levels with equal increment. The 2b
levels are called quantization levels and the increment in each level is called
quantization step size. If R is the range of analog signal then, Quantization step size,
q = R/2b
11. How the digital filter is affected by quantization of filter coefficients?
The quantization of the filter coefficients will modify the value of poles &
zeros and so the location of poles and zeros will be shifted from the desired location.
This will create deviations in the frequency response of the system. Hence the
resultant filter will have a frequency response different from that of the filter with
un-quantized coefficients.
21
22
23
Binary point
Sign bit
24
UNIT - 5
DSP APPLICATIONS
Syllabus:
Multi-rate signal processing: Decimation, Interpolation, Sampling rate
conversion by a rational factor Adaptive Filters: Introduction, Applications of
adaptive filtering to equalization.
Two mark questions:
1. What is multi-rate signal processing?
(APR-2012)
The theory of processing signals at different sampling rates is called multirate signal processing.
2. Define down sampling. (or Decimation)
Down sampling a sequence x(n) by a factor M is the process of picking every
th
M sample and discarding the rest.
3. What is mean by up-sampling? (or interpolation)
Up-sampling by a factor L is the process of inserting L-1 zeros between two
consecutive samples.
4. If the spectrum of sequence x(n) is X(ejw), then what is the spectrum of a
signal down-sampled by factor 2?
(DEC-2013)
jw
jw/2
jw((w/2)-
Y(e )=(1/2)[X(e )+ X(e
)]
5. If the Z-transform of a sequence x(n) is X(z) then what is the Z-transform of
a sequence down-sampled by a factor M?
Y(z)= (1/M)
(z(1/M)e(-j2k/M))
6. If the z-transform of a sequence x(n) is X(z) then what is the z-transform of
a sequence up-sampled by a factor L?
Y(z)= X(zL)
7. What is the need for anti-imaging filter after up-sampling a signal?
The frequency spectrum of up-sampled signal with a factor L, contains (L-1)
additional images of the input spectrum. Since we are not interested in image spectra,
a low-pass filter with a cutoff frequency wc = (/L) can be used after up-sampler.
This filter is known as anti-imaging filter.
25
26
(APR-2004)
y(n)
v(n)
20. Draw the frequency domain representation of downsampler.
x(n)
y(n) = x(Dn)
D
x(ejw)
Y(ejw)=[1/D] x(ejw/D)
27
23. Draw the Multirate signal processing system with analysis and synthesis
filter banks.
(APR-2009)
28
(DEC-2010)
yD(n)
yD(n)= x(Mn)
For an input sequence x(n), select only the samples which occur at integer
multiples of M. The other samples are thrown away.
Aliasing will occur in yD(n) unless x(n) is sufficiently bandlimited loss of
information.
28. Draw the structure of L-folder expander.
For an input sequence x(n), insert L 1 zeros between each sample.
x(n) can always be recovered from yE(n) no loss of information, no aliasing.
29
X(n)
yE(n)
YE(n)= x(Mn)
29. Develop an expression for the output y(n) as a function of the input x(n)
for the multirate structure of below fig.
(NOV-2010)