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Multirate Digital Signal Processing

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0% found this document useful (0 votes)
18 views31 pages

Multirate Digital Signal Processing

This is a documented notes of Multi-rate Digital Signal Processing.

Uploaded by

ayushjain6548
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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Multirate Signal Processing

1
Introduction
The process of converting or alterating a signal from a given
sampling rate to a different sampling rate is called sampling
rate conversion.
The system that process data at more than one sampling rate
are known as multirate systems.
In single-rate DSP systems, all data is sampled at the same
rate , no change of rate within the system.
 In multirate DSP systems, sample rates are changed (or are
different) within the system.

2
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

Two ways of multirate sampling


• Down sampling (Decimation)
• Up sampling (Interpolation)

3
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] that is L
times larger than that of the input sequence x[n]

• Block-diagram representation

x[n] L xu [n ]

4
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, 
y[n]  
 0, otherwise

5
6
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]

7
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[Mn]

8
9
Up-Sampler

Frequency-Domain Characterization
• Consider a factor-of-2 up-sampler whose input-
output relation in the time-domain is given by

x[n / 2], n  0,  2,  4,


xu [n]  
 0, otherwise

10
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 )

11
Up-Sampler
• In a similar manner, we can show that for a
factor-of-L up-sampler
X u(z)  X(z L
)
• On the unit circle, for z  e j , the input-
output relation is given by
j jL
X u (e )  X (e )

12
Up-Sampler

X u (e j )  X (e jL )

13
Frequency-Domain Characterization of Down Sampler
Applying z transform to the input output relation of a factor of M down sampler
y[n]  x[ Mn]
we get

Y ( z)  
n 
x[ Mn] z  n

Expression on the right handside cannot be directly expressed in terms of X ( z )


To solve this problem, defined a new sequence xint [ n]
 x[n], n  0,  M ,  2M ,
xint [n]  
 0, otherwise
 
Y ( z)  
n 
x[ Mn] z n
 
n 
xint [ Mn] z  n

 
k 
xint [k ] z  k / M  X int ( z1/ M )

14
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
M k 0
W where WM  e  j 2 / M

Taking the z  transform of


xint [n]  c[ n]  x[ n]
M 1
1
and making use of c[n] 
M
W
k 0
kn
M


X int ( z )   c[n]x[n] z
n 
n

1 
 M 1 kn 
15

M
   M 
n   k  0
W x[ n ] z n
1 M 1
  kn  n 
X int ( z ) 
M
 
k  0  n 
x[ n ]WM z 

1 M 1   n 
X int ( z )    
M k 0  n 
k
x[n](WM z ) 

1 M 1
X int ( z )  
M k 0
X  zW M 
k

subsituting in Y(Z)
Y(Z)  X int ( z1/ M )
1 M -1
Y(Z)= 
X
M k =0
z  W
1/ M -k
M 
On the unit circle, for z  e jw , output is given by
M 1
Y( e ) 
jw 1
M
 X  e jw/M
WM 
k

16 k 0
 
M 1
1
Y(e ) 
jw

M
 X
k 0
e jw/M
WM
k

 
M 1
1
Y(e ) 
jw

M
 X
k 0
e jw/M j2 k/M
e

 
M 1
1
Y(e ) 
jw

M
 X
k 0
e j(w - 2k)/M

17
Let x(n) be the input and y(n) be the output of a down sampler. For
down sampling by a factor of 2, determine the z-transform of the
output.

18
Cascade Equivalences
• Basic sampling rate converter devices can be used to
change the sampling rate of a signal only by an
integer factor.
• To implement a fractional change in the sampling
rate we need to employ a cascade of an up-sampler and
a down-sampler

• Upsampler and Downsampler Cascade

19
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Copyright © 2001, S. K. Mitra
23
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Subband Coding
• Subband Coding is an approach to compression which the
relies on separating the source output into different bands of
frequencies using digital filters.
• Divide the spectrum of the signal into several subbands then
allocate bits to each band appropriately.
• In practice, certain frequency bands are less important
perceptually because they contain less significant information
• Bands with less information may be quantized with fewer
bits.
• Consider quantizing the samples of a speech signal. How
many bits are required?
• In general, 16 bits per sample is normally used for audio. This
gives an adequate dynamic range.
28
 16 bits per sample, 10 kHz sampling frequency gives kbits/s
 Divide into 2 bands: high frequency and low frequency
subbands.
 High frequencies of speech are less important to intelligibility.
 Therefore use only 8 bits per sample
 Reconstructed signal has no noticeable reduction is signal
quality.
 The source output can be decomposed into its constituent parts
using digital filters.
 Each of these constituent parts will be different bands of
frequencies which make up the source. A compression approach
where digital filters are used to separate the source output into
different bands of frequencies.
Each part then can be encoded separately.
29
A filter is system that isolates certain frequencies.
(i) Low Pass Filters
(ii) High Pass Filters
(iii) Band Pass Filters
Sampling and Nyquist rule :
If fo is the highest frequency of the signal then the sampling rate > 2fo per
second can accurately represent the continuous signal in digital form.
Extension of Nyquist rule:
For signal with frequency components between frequencies f1and f2 then,
sampling rate = 2(f2 — f1) per second.
Violation of Nyquist rule:
Distortion due to aliasing.
Filters used in Subband Coding
 Quadrature Mirror Filters (QMF),
 Johnston Filter
 Smith-Barnwell Filters
 Daubechies Filters
30
….and so on
31

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