Multi Rate Digital Signal Processing

The process of employing multiple sampling rates

in the processing of digital signals is called ‘Multi
Rate Digital Signal Processing’.

The process of converting a signal from a given

rate or sampling frequency to a different rate or
sampling frequency is called ‘sampling rate
conversion’.
Sampling Rate Conversion Methods
There are two general methods are there for accomplishing
sampling rate conversion of a digital signal, one method
is,
Pass the digital signal through DAC, filter it if necessary,
and then resample the resulting analog signal at desired rate
i.e pass the analog signal through an ADC.

Disadvantages with this method

Signal distortion is introduced
i. by the DAC during signal reconstruction
ii. By the quantization effects during ADC
Another method is digital domain method.
In digital domain the sampling rate conversion can be
done,
By means of ‘Down sampling’ - Reducing the sampling rate
by an integer factor ‘D’.
By means of ’ Up sampling’ - Increasing the sampling rate
by an integer factor ‘I’.
The process of reducing the sampling rate by an integer
factor ‘D’ is called ‘Decimation’.
The process of increasing the sampling rate by an integer
factor ‘I’ is called ‘Interpolation’.
 Decimation by a factor D
• Let consider x(n) be a i/p sequence which is passed
through a LPF, characterized by the impulse response
denoted by h(n) and a frequency denoted by HD(ω) for
performing decimation process as shown below:

x(n) LPF v(n) Down sampler

h(n) ↓D
• The o/p of the filter is a sequence given as,

Which is then down sampled by the factor D to produce y(m). Thus

• The frequency domain characteristics of the o/p sequence
y(m) can be obtained by
• Now the Z-Transform of y(m) is
If the filter is properly designed then
c[n]

Standard Approach
Decimation by a Factor D
 Interpolation by a factor I
• The process of increasing the sampling rate of a digital
signal by an integer factor I is called Interpolation.
• In Interpolation process an I-1 new samples which are
zero’s are interpolated between the successive values of
the digital signal.
• Let v(m) be a sequence of sampling rate Fy=Ifx is obtained
from x(n) after adding I-1 zero’s between successive
values of x(n) and is expressed as, HI(ωy)
v(m) hI(m) y(m)
↑I
• The z-transform of v(m) is,

• The corresponding frequency spectra is obtained by

substituting
 HI(ωy)
v(m) hI(m) y(m)
↑I
 Sampling rate conversion by a rational
factor I/D
• Sampling rate conversion by a rational factor ‘I/D’ can be
achieved by first performing interpolation by the factor ‘I’
and then decimating the interpolator o/p by a factor ‘D’.
• In this process both the interpolator and decimator are
cascaded as shown in the figure below:

x(n) Upsampler LPF LPF Downsampler y(m)

Rate Fx ↑I hU(l) hd(l) ↓D

Interpolator Decimator

Rate=Fx(I/D)=Fy

Rate=IFx
• In this figure the two filters with impulse responses
hu(l),hd(l)are operated at the same frequency=IFx and
hence these two filters are combined in to a single LPF of
impulse response h(l) which is shown in the figure below:
H(ωv)
x(n) Upsampler v(l) LPF w(l) Downsampler y(m)
Rate Fx ↑I h(l) ↓D Rate=Fx(I/D)=Fy

Rate=IFx
• Let v(l) be the o/p of the interpolator and can be
represented as,

• Let w(l) be the o/p of the filter and can be obtined as,

• Finally the o/p of the sampling rate converter denoted by

y(m) can be obtained as,
• The corresponding frequency domain representation is,
=
 FIR Filter
• In general, a FIR system is described by the difference
equation

• or by the system transfer function

• According to the equ…(1)

y(n)=h(0)x(n)+h(1)x(n-1)+…….+h(M-1)x(n-M+1) and
can be realized as
 y(n)=h(0)x(n)+h(1)x(n-1)+…….+h(M-1)x(n-M+1) and
can be realized as
X(n) y(n)
 Filter Design and Implementation for
Sampling Rate Conversion
• Here sampling rate
conversion which is
‘Decimation’ and
‘Interpolation’ is
performed by direct form
FIR filter structures.
• The design and
implementation of FIR
filter for performing
decimation process as
shown in the figure:
• This realization is simple but
inefficient because,
1.up sampling process introduces
I-1 zero’s between successive
points of the signal.
2.If ‘I’ is large, most of the signal
components in the FIR filter are
zero.
3.The multiplications and additions
in the FIR filter result in zero’s
due to this large ‘I’.
• Therefore it is necessary
to develop a more
efficient structure.
• This can be achieved by
embedding the down
sampling operation
within the filter it self as
shown in the figure.
• In this structure all multiplications and additions are
performed at the lower sampling rate Fx/D.
• Thus desired efficiency can be achieved.
• Next consider the
interpolation process
which can be performed
by means direct form FIR
filter structures as shown
in the figure.
• This structure is realized
by first inserting I-1
zero’s between the
samples of x(n) and then
filtering the sequence.
• The major problem in this
realization is that the filter
computations are
performed at high
sampling rate Ifx.
• This problem is solved by
using transposed form of
FIR filter and embedding
the up sampler within the
filter as shown in the
figure.
• So all multiplications are
performed at the lower
rate Fx.
 Design and Implementation of Poly Phase Filter
Structures for Sampling Rate Conversion
• The sampling rate conversion which is ‘interpolation’
(‘decimation’) is also performed by means of poly phase
filter structures as shown in the figure below which results
in better computational efficiency than FIR systems.
• Here each sub filter is
defined with unit impulse
responses
pk(n)=h(k+nI) where
k=0,1,……I-1,
n=0,1,……K-1
• This structure is achieved
by reducing the large FIR
filter of length ‘M’ in to a
set of smaller filters of
length K=M/I where ‘M’ is
selected to be a multiple of
‘I’.
• Here all sub filters are
basically al pass filters of
different phase
characteristics and are
arranged in a parallel form.
• The o/p of each filter can
be selected by a
commutator.
• The rotation of the
commutator is in the
counter clockwise
direction.
• This filter structure
performs computations at
low sampling rate Fx.
• Next is the decimator which can be realized by
transposing the Interpolator structure as shown below:
• Here each sub filter is defined with unit impulse responses
pk(n)=h(k+nD) where k=0,1,……D-1, n=0,1,……K-1
• This structure is achieved by reducing the large FIR filter
of length ‘M’ in to a set of smaller filters of length K=M/D
is an integer and ‘M’ is selected to be a multiple of ‘D’.
Applications of Multi-rate Signal Processing
• Design of Phase Shifters
• Interfacing of Digital Systems with different sampling
rates
• Implementation of narrow band LPF’s
• Implementation of Digital filter banks
• Sub band coding of speech signals
• Quadrature mirror filters
• Transmultiplexers
• Oversampling A/D and D/A conversion
 Design of Phase Shifters
• Here a network is designed that delays the signal x(n) by a
rational fraction of a sampling interval Tx i.e d=(K/I)Tx,
where ‘d’ is the delay.
• In the frequency domain this delay corresponds to a linear
phase shift of the form Θ(ω)=-(K/I) ω.
• Let consider the system which performs both
‘interpolation’ and ‘decimation’ as shown below:
• The interpolator increases the sampling rate by a factor ‘I’.
• The LPF eliminates the images or frequency duplications
in the spectrum of the interpolated signal.
• Next the o/p of the filter is delayed by ‘k’ samples at the
sampling rate IFx.
• Then the delayed signal is decimated by factor D=I.
• Thus the desired delay of (K/I)Tx will be achieved.
 Interfacing of Digital Systems with different
sampling rates
• Let consider the interfacing of two digital systems ‘A’ and
‘B’ through a digital sample and hold block as shown below:

• The sapling rate of system ‘A’ is Fx and the sampling rate

of system ‘B’ is Fy .
• The o/p of system is fed to an interpolator which increases
the sampling rate by a factor ‘I’.
• The o/p of the interpolator which is sampling rate IFx is
fed to a digital sample-and-hold system.
• The signals from the digital sample-and-hold system are
read out in to system ‘B’ at a rate IFy of system ‘B’.
• Thus the desired interfacing is achieved and the o/p rate of
digital sample-and-hold system is not synchronized with
the i/p rate.
 Implementation of Digital filter banks
• Digital filter banks are categorized as two types based on
‘decimation’ and ‘interpolation’
1. Analysis filter banks
2. Synthesis filter banks
• An analysis filter bank consists of a set of filters with
system function {Hk(z)} are arranged in parallel with i/p
x(n) as shown below:
• The frequency response characteristics of this filter bank
splits the signal in to a corresponding number of sub bands.
• Next, a synthesis filter bank consists of a set of filters with
system function {Gk(z)} are arranged in parallel with i/p
yk(n) as shown below:

• The o/p this filter bank are summed to form a synthesized

signal x(n).
• These filter banks are often used for performing
spectrum analysis and signal synthesis.
• When a filter bank is used for computing DFT of a
sequence {x(n)} then the filter bank is called DFT
filter bank.
• The analysis filter bank for computing DFT consists
of ‘N’ filters of system function {Hk(z)} where
{k=0,1,2,…….N-1}.
• If {Hk(z)} where {k=1,2,…….N-1} then the
analysis filter bank is called uniform DFT filter bank
and the corresponding frequency domain
representation is
• The frequency response characteristics of the filters {Hk(z),
k=1,2,…….N-1} are obtained by uniformly shifting the
frequency response of the filter having system function
{H0(z)} by multiples of ‘2π/N’ .
• In the time domain the impulse responses are expressed as,
• The uniform DFT analysis filter bank can be
realized as shown below:
• In this structure the frequency components in the
sequence {x(n)} are translated in frequency to low
pass by multiplying x(n) with
and the resultant signals are passed through LPF’s
with impulse responses denoted by h0(n).
• The resulting decimated signal can be expressed
as

• Where Xk(m) are samples of DFT at frequencies

ωk=2πk/N .
• The corresponding synthesis filter for each element in the
filter bank can be viewed as shown below:

• The i/p signal sequences [Yk(m),k=0,1,……N-1] are up-

sampled by a factor of I=D, filtered to remove the images
or image frequency components and translated in
frequency by, multiplication by the complex exponentials
{exp(j2πnk/N), k=0,1,……N-1}.
• The resulting frequency translated signals from the N
filters are then summed for obtaining v(n) as shown below:

• Where the factor 1/N is a normalizing factor.

{yn(m)}represent samples of the inverse DFT sequence
corresponding to {yk(m)}.
{g0(n)} is the impulse response of the interpolation filter.
Sub band coding of speech signals
Quadrature mirror filters
Two channel QMF Bank