Multirate Digital Signal Processing
Multirate Digital Signal Processing
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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.
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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
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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 ]
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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
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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]
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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]
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Up-Sampler
Frequency-Domain Characterization
• Consider a factor-of-2 up-sampler whose input-
output relation in the time-domain is given by
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Up-Sampler
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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 jL
X u (e ) X (e )
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Up-Sampler
X u (e j ) X (e jL )
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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
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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
X int ( z ) c[n]x[n] z
n
n
1
M 1 kn
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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 - 2k)/M
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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.
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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
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Copyright © 2001, S. K. Mitra
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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.
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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.
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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
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….and so on
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