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DSP 2

The document provides contact information for a research organization called Expert Systems and Solutions located in Paiyanoor, Chennai, India. It calls for final year students and Ph.D students in electrical and electronics fields to submit research projects that can be assembled in their research labs with guidance from experts.

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0% found this document useful (0 votes)
37 views

DSP 2

The document provides contact information for a research organization called Expert Systems and Solutions located in Paiyanoor, Chennai, India. It calls for final year students and Ph.D students in electrical and electronics fields to submit research projects that can be assembled in their research labs with guidance from experts.

Uploaded by

setsindia3735
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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You are on page 1/ 53

EXPERT SYSTEMS AND SOLUTIONS

Email: [email protected]
[email protected]
Cell: 9952749533
www.researchprojects.info
PAIYANOOR, OMR, CHENNAI
Call For Research Projects Final
year students of B.E in EEE, ECE,
EI, M.E (Power Systems), M.E
(Applied Electronics), M.E (Power
Electronics)
Ph.D Electrical and Electronics.
Students can assemble their hardware in our
Research labs. Experts will be guiding the
projects.

1
Multirate Digital Signal
Processing

Basic Sampling Rate Alteration Devices


• Up-sampler - Used to increase the sampling
rate by an integer factor
• Down-sampler - Used to decrease the
sampling rate by an integer factor

2
Up-Sampler
Time-Domain Characterization
• An up-sampler with an up-sampling factor
L, where L is a positive integer, develops an
output sequence xu [n] with a sampling rate
that is L times larger than that of the input
sequence x[n]
• Block-diagram representation
x[n] L xu [n ]

3
Up-Sampler
• Up-sampling operation is implemented by
inserting L  1 equidistant zero-valued
samples between two consecutive samples
of x[n]
• Input-output relation
 x[n / L], n  0,  L,  2 L,
xu [n]  
 0, otherwise

4
Up-Sampler
• In practice, the zero-valued samples
inserted by the up-sampler are replaced
with appropriate nonzero values using some
type of filtering process
• Process is called interpolation and will be
discussed later

5
Down-Sampler
Time-Domain Characterization
• An down-sampler with a down-sampling
factor M, where M is a positive integer,
develops an output sequence y[n] with a
sampling rate that is (1/M)-th of that of the
input sequence x[n]
• Block-diagram representation
x[n] M y[n]

6
Down-Sampler
• Down-sampling operation is implemented
by keeping every M-th sample of x[n] and
removing M  1 in-between samples to
generate y[n]
• Input-output relation
y[n] = x[nM]

7
Basic Sampling Rate
Alteration Devices
• Sampling periods have not been explicitly
shown in the block-diagram representations
of the up-sampler and the down-sampler
• This is for simplicity and the fact that the
mathematical theory of multirate systems
can be understood without bringing the
sampling period T or the sampling
frequency FT into the picture
8
Down-Sampler
• Figure below shows explicitly the time-
dimensions for the down-sampler

x[ n ]  xa ( nT ) M y[ n ]  xa ( nMT )

Input sampling frequency Output sampling frequency


1 ' FT 1
FT  FT  
T M T'

9
Up-Sampler
• Figure below shows explicitly the time-
dimensions for the up-sampler
x[ n ]  xa ( nT ) L y[n]
 x ( nT / L ), n 0,  L, 2 L,
 a
 0 otherwise

Input sampling frequency Output sampling frequency


1 ' 1
FT  FT  LFT 
T T'
10
Basic Sampling Rate
Alteration Devices
• The up-sampler and the down-sampler are
linear but time-varying discrete-time
systems
• We illustrate the time-varying property of a
down-sampler
• The time-varying property of an up-sampler
can be proved in a similar manner
11
Basic Sampling Rate
Alteration Devices
• Consider a factor-of-M down-sampler
defined by y[n] = x[nM]
• Its output y1[n] for an input x1[n]  x[n  n0 ]
is then given by
y1[n]  x1[ Mn]  x[ Mn  n0 ]
• From the input-output relation of the down-
sampler we obtain
y[n  n0 ]  x[ M (n  n0 )]
12
 x[ Mn  Mn0 ]  y1[n]
Up-Sampler
Frequency-Domain Characterization
• Consider first a factor-of-2 up-sampler
whose input-output relation in the time-
domain is given by
 x[n / 2], n  0,  2,  4,
x u [n ]  
 0, otherwise

13
Up-Sampler
• In terms of the z-transform, the input-output
relation is then given by
 
n n
X u ( z)   u
x [ n ] z   x[ n / 2] z
n   n  
n even

 
m 
x[m] z 2 m  X ( z 2 )

14
Up-Sampler
• In a similar manner, we can show that for a
factor-of-L up-sampler
L
X u ( z)  X ( z )
• On the unit circle, for z  e j , the input-
output relation is given by
j j L
X u (e )  X (e )

15
Up-Sampler
• Figure below shows the relation between
j j
X (e ) and X u (e ) for L = 2 in the case of
a typical sequence x[n]

16
Up-Sampler
• As can be seen, a factor-of-2 sampling rate
j
expansion leads to a compression of X (e )
by a factor of 2 and a 2-fold repetition in
the baseband [0, 2]
• This process is called imaging as we get an
additional “image” of the input spectrum

17
Up-Sampler
• Similarly in the case of a factor-of-L
sampling rate expansion, there will be L  1
additional images of the input spectrum in
the baseband
• Lowpass filtering of xu [n] removes the L  1
images and in effect “fills in” the zero-
valued samples in xu [n] with interpolated
sample values
18
Down-Sampler
Frequency-Domain Characterization
• Applying the z-transform to the input-output
relation of a factor-of-M down-sampler

y[n]  x[Mn]
we get

n
Y ( z)   x[Mn] z
n  
• The expression on the right-hand side cannot be
directly expressed in terms of X(z)
19
Down-Sampler
• To get around this problem, define a new
sequence xint [n] :
 x[n], n  0,  M ,  2 M ,
xint [n]  
 0, otherwise
• Then
 
n n
Y ( z)   x[Mn] z   xint [Mn] z
n   n  

k / M 1/ M
  xint [k ] z  X int ( z )
20 k  
Down-Sampler
• Now, xint [n] can be formally related to x[n]
through
xint [n]  c[n]  x[n]
where
1, n  0,  M ,  2 M ,
c[n]  
0, otherwise
• A convenient representation of c[n] is given
by 1 M 1 kn
c[n]  
M k 0
WM

where WM  e  j 2 / M
21
Down-Sampler
• Taking the z-transform of xint [n]  c[n]  x[n]
and making use of
1 M 1
c[n]   WMkn
M k 0
we arrive at
   M 1
n 1 kn  n
X int ( z )   c[n]x[n] z    M
 W  x[ n ] z
n   M n    k 0 
1 M 1 
 
 1 M 1
kn  n k
  
  x
M k 0  n  
[ n ]WM z 
   X z W M
22  M k 0
Down-Sampler
• Consider a factor-of-2 down-sampler with
an input x[n] whose spectrum is as shown
below

• The DTFTs of the output and the input


sequences of this down-sampler are then
related as
1
Y (e )  { X (e j / 2 )  X (e j / 2 )}
j 
23 2
Down-Sampler
• Now X (e j / 2 )  X (e j ( 2 ) / 2 ) implying
that the second term X (  e j  / 2 ) in the
previous equation is simply obtained by
shifting the first term X (e j / 2 ) to the right
by an amount 2 as shown below

24
Down-Sampler
• The plots of the two terms have an overlap,
and hence, in general, the original “shape”
of X (e j ) is lost when x[n] is down-
sampled as indicated below

25
Down-Sampler
• This overlap causes the aliasing that takes
place due to under-sampling
• There is no overlap, i.e., no aliasing, only if
X ( e j )  0 for    / 2
• Note: Y (e j )is indeed periodic with a
period 2, even though the stretched version
of X (e j ) is periodic with a period 4

26
Down-Sampler
• For the general case, the relation between the
DTFTs of the output and the input of a factor-
of-M down-sampler is given by
1 M 1
Y ( e j )   X ( e j ( 2 k ) / M )
M k 0
• Y (e j )is a sum of M uniformly shifted
and stretched versions of and
X (scaled
e j )
by a factor of 1/M
27
Down-Sampler
• Aliasing is absent if and only if
X (e j )  0 for    / M
as shown below for M = 2
X (e j )  0 for    / 2

28
Cascade Equivalences
• A complex multirate system is formed by an
interconnection of the up-sampler, the
down-sampler, and the components of an
LTI digital filter
• In many applications these devices appear
in a cascade form
• An interchange of the positions of the
branches in a cascade often can lead to a
computationally efficient realization
29
Cascade Equivalences
• To implement a fractional change in the
sampling rate we need to employ a cascade
of an up-sampler and a down-sampler
• Consider the two cascade connections
shown below
x[n ] M L y1 [ n ]

x[n ] L M y2 [ n]

30
Cascade Equivalences
• A cascade of a factor-of-M down-sampler
and a factor-of-L up-sampler is
interchangeable with no change in the
input-output relation:
y1[n]  y2[n]
if and only if M and L are relatively prime,
prime
i.e., M and L do not have any common
factor that is an integer k > 1

31
Cascade Equivalences
• Two other cascade equivalences are shown
below
Cascade equivalence #1
x[n ] M H (z ) y1 [ n ]

 x[n ] H (z M ) M y1 [ n ]
Cascade equivalence #2

x[n ] L H (z L ) y2 [ n]

 x[n ] H (z ) L y2 [ n]
32
Filters in Sampling Rate
Alteration Systems
• From the sampling theorem it is known that
a the sampling rate of a critically sampled
discrete-time signal with a spectrum
occupying the full Nyquist range cannot be
reduced any further since such a reduction
will introduce aliasing
• Hence, the bandwidth of a critically
sampled signal must be reduced by lowpass
filtering before its sampling rate is reduced
by a down-sampler
33
Filters in Sampling Rate
Alteration Systems
• Likewise, the zero-valued samples
introduced by an up-sampler must be
interpolated to more appropriate values for
an effective sampling rate increase
• We shall show next that this interpolation
can be achieved simply by digital lowpass
filtering
• We now develop the frequency response
specifications of these lowpass filters
34
Filter Specifications
• Since up-sampling causes periodic
repetition of the basic spectrum, the
unwanted images in the spectra of the up-
sampled signal xu [n] must be removed by
using a lowpass filter H(z), called the
interpolation filter,
filter as indicated below
xu [n]
x[n ] L H (z) y[n ]

• The above system is called an interpolator


35
Filter Specifications
• On the other hand, prior to down-sampling,
the signal v[n] should be bandlimited to
  /M by means of a
lowpass filter, called the decimation filter,
filter
as indicated below to avoid aliasing caused
by down-sampling
x[n ] H (z) M y[n ]

• The above system is called a decimator

36
Interpolation Filter
Specifications
• Assume x[n] has been obtained by sampling
a continuous-time signalxa (t ) at the Nyquist
rate
j
• If X a ( j) and X (e ) denote the Fourier
transforms of xa (t ) and x[n], respectively,
then it can be shown
1   j  j 2 k 
j
X (e )   Xa
To k    To


37
• where To is the sampling period
Interpolation Filter
Specifications
• Since the sampling is being performed at the
Nyquist rate,
rate there is no overlap between the
shifted spectras of X ( j / To )
• If we instead sample xa (t ) at a much higher
rate T  L  To yielding y[n], its Fourier
j
transform Y (e ) is related to X a ( j)
through
j 1   j  j 2 k  L   j  j 2 k 
Y (e )  Xa
T k   
  Xa 
T  To k    To / L 
38
Interpolation Filter
Specifications
• On the other hand, if we pass x[n] through a
factor-of-L up-sampler generating xu [n] ,
the relation between the Fourier transforms
of x[n] andxu [n] are given by
X u ( e j )  X ( e j L )
• It therefore follows that if xu [n] is passed
through an ideal lowpass filter H(z) with a
cutoff at /L and a gain of L, the output of
the filter will be precisely y[n]
39
Interpolation Filter
Specifications
• In practice, a transition band is provided to
ensure the realizability and stability of the
lowpass interpolation filter H(z)
• Hence, the desired lowpass filter should
have a stopband edge at  s   / L and a
passband edge  p close to  s to reduce the
distortion of the spectrum of x[n]

40
Interpolation Filter
Specifications
• If  c is the highest frequency that needs to
be preserved in x[n], then
 p  c / L
• Summarizing the specifications of the
lowpass interpolation filter are thus given
by
j  L,   c / L
H (e )  
 0,  / L    
41
Decimation Filter
Specifications
• In a similar manner, we can develop the
specifications for the lowpass decimation
filter that are given by

j 1,   c / M
H (e ) 
0,  / M    

42
Filter Design Methods
• The design of the filter H(z) is a standard
IIR or FIR lowpass filter design problem
• Any one of the techniques outlined in
Chapter 7 can be applied for the design of
these lowpass filters

43
Filters for Fractional Sampling
Rate Alteration
• A fractional change in the sampling rate can
be achieved by cascading a factor-of-M
decimator with a factor-of-L interpolator,
where M and L are positive integers
• Such a cascade is equivalent to a decimator
with a decimation factor of M/L or an
interpolator with an interpolation factor of
L/M
44
Filters for Fractional Sampling
Rate Alteration
• There are two possible such cascade
connections as indicated below
H d (z) M L H u (z)

L H u (z) H d (z) M

• The second scheme is more computationally


efficient since only one of the filters, H u (z )
or H d (z ), is adequate to serve as both the
interpolation and the decimation filter
45
Filters for Fractional Sampling
Rate Alteration
• Hence, the desired configuration for the
fractional sampling rate alteration is as
indicated below where the lowpass filter
H(z) has a stopband edge frequency given
by
  
 s  min , 
L M 
L H (z) M
46
Computational Requirements
• The lowpass decimation or interpolation
filter can be designed either as an FIR or an
IIR digital filter
• In the case of single-rate digital signal
processing, IIR digital filters are, in general,
computationally more efficient than
equivalent FIR digital filters, and are
therefore preferred where computational
cost needs to be minimized

47
Computational Requirements
• This issue is not quite the same in the case
of multirate digital signal processing
• To illustrate this point further, consider the
factor-of-M decimator shown below
v[n ] y[n ]
x[n ] H (z) M
• If the decimation filter H(z) is an FIR filter
of length N implemented in a direct form,
then N 1
v[n]   h[m] x[n  m]
48 m 0
Computational Requirements
• Now, the down-sampler keeps only every M-
th sample of v[n] at its output
• Hence, it is sufficient to compute v[n] only
for values of n that are multiples of M and
skip the computations of in-between
samples
• This leads to a factor of M savings in the
computational complexity
49
Computational Requirements
• Now assume H(z) to be an IIR filter of
order K with a transfer function
V ( z) P( z )
 H ( z) 
X ( z) D( z )
where K
n
P( z )   pn z
n 0
K
n
D( z )  1   dn z
50 n 1
Computational Requirements
• Its direct form implementation is given by
w[n]  d1w[n  1]  d 2 w[n  2]  
 d K w[n  K ]  x[n]
v[n]  p0 w[n]  p1w[n  1]    pK w[n  K ]
• Since v[n] is being down-sampled, it is
sufficient to compute v[n] only for values of
n that are integer multiples of M
51
Computational Requirements
• However, the intermediate signal w[n] must
be computed for all values of n
• For example, in the computation of
v[M ]  p0 w[M ]  p1w[M  1]    pK w[M  K ]
K+1 successive values of w[n] are still
required
• As a result, the savings in the computation
in this case is going to be less than a factor
of M
52
Computational Requirements
• For the case of interpolator design, very
similar arguments hold
• If H(z) is an FIR interpolation filter, then the
computational savings is by a factor of L
(since v[n] has L  1 zeros between its
consecutive nonzero samples)
• On the other hand, computational savings is
significantly less with IIR filters
53

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