Unit 4 Random Signal

Processing
MULTIRATE SIGNAL
PROCESSING-LAST PART OF
UNIT 4
BASICS OF RANDOM SIGNAL PROCESSING

• Introduction to probability function, joint probability,

conditional probability – estimation parameters – joint
distribution function, probability density function,
ensemble average – mean squared value, variance,
standard deviation, moments, correlation, covariance,
orthogonality, auto-covariance, auto-correlation, cross-
covariance and cross-correlation – stationarity – ergodic
– white noise – energy density spectrum – power
density spectrum estimation – periodogram – direct
method, indirect method, Bartlett method – Welch
method. Decimator (down sampling) – frequency-
domain analysis of decimator – interpolation (up
sampling) – frequency-domain analysis of
interpolator
• HOD WILL HANDLE:
• power density spectrum estimation – periodogram –
direct method, indirect method, Bartlett method –
Welch method
• ProF.J.Valarmathi:VIT
• Introduction to probability function, joint probability,
conditional probability – estimation parameters – joint
distribution function, probability density function,
ensemble average – mean squared value, variance,
standard deviation, moments, correlation, covariance,
orthogonality, auto-covariance, auto-correlation, cross-
covariance and cross-correlation – stationarity – ergodic
– white noise – energy density spectrum
Definition of Multirate Signal
Processing
• The systems that uses single sampling
rate from A/D converter to D/A converter
are known as single rate systems.
• When the sampling rate of the signals is
unequal at various parts of the system-
Those discrete time systems that process
data at more than one sampling rate are
known as multirate systems.
Where Multirate signal processing
is used:
• 1.In high quality data acquisition and
storage systems.
• 2.In audio signal processing. For example
a CD is sampled at 44.1KHz but DAT is
sampled at 48 KHz. Conversion between
DAT and CD use multirate signal
processing technique.
• 3.Narrow band filtering for fetal ECG and
EEG
• 4.In video PAL and NTSC run at different
sampling rates. Therefore to watch an
American program in Europe one need a
sampling rate converter.
• 5.In speech processing to reduce the
storage space or the transmitting rate of
the speech data.
• 6.In transmultiplexers.
Basic operations: Interpolation and
decimation.
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.
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
Up-Sampler

Output sequence up-sampled by 3

Input Sequence 1
1

0.5
0.5

Amplitude
Amplitude

0 0

-0.5 -0.5

-1 -1
0 10 20 30 40 50 0 10 20 30 40 50
Time index n Time index n
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
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]
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]
Down-Sampler

Input Sequence Output sequence down-sampled by 3

1 1

0.5 0.5
Amplitude

0 Amplitude 0

-0.5 -0.5

-1 -1
0 10 20 30 40 50 0 10 20 30 40 50
Time index n Time index n
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 into the picture
FT
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'
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'
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
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 )]
 x[ Mn  Mn0 ]  y1[n]
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)
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 )
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
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
 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 
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
X ( e j / 2 )
obtained by shifting the first term
to the right by an amount 2 as
shown below
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
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
Y ( e j )
• Note: is indeed periodic with a
period 2, even though the stretched
version of X (e j ) is periodic with a period
4
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 X (e j )
and scaled by a factor of 1/M
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
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

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
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 )
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 )
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]
Up-Sampler

• As can be seen, a factor-of-2 sampling

rate expansion leads to a compression
of X (e j ) 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
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
• Low pass filtering of xu [n] removes the
images and in effect “fills in” the zero-L  1
valued samples in xu [n] with interpolated
sample values
Filter Specifications
• Since up-sampling causes periodic
repetition of the basic spectrum, the
unwanted images in the spectra of the up-
sampled signal xumust
[n] 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
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
Interpolation Filter Specifications
• On the other hand, if we pass x[n]
through a factor-of-L up-sampler xu [n]
generating xu [n] , the relation between
the Fourier transforms of x[n] and
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]