VISVESVARAYA TECHNOLOGICAL UNIVERSITY

BELGAUM-590014

For the academic year 2010-2011

A Technical Seminar Report on

MULTIRATE SIGNAL PROCESSING
In partial fulfillment of the requirement for the award of degree of

BACHELOR OF ENGINEERING
IN
TELECOMMUNICATION ENGINEERING

Presented By:
RAJESH PATIL H.S
[1MV07TE039]
Under the Guidance of
Internal Guide
Sri A.R.RAVINDRA
Lecturer
Department of Telecommunication Engineering

SIR M. VISVESVARAYA INSTITUTE OF TECHNOLOGY

Department of Telecommunication Engineering
Krishnadevarayanagar, Hunasamaranahalli, New Airport Road,
Via Yelahanka, Bangalore–562157
 SIR M. VISVESVARAYA INSTITUTE OF
TECHNOLOGY
Department of Telecommunication Engineering
Krishnadevarayanagar, Hunasamaranahalli, New Airport Road, Via
Yelahanka,
Bangalore-562157

CERTIFICATE
This is to certify that, the report of the seminar entitled
“MULTIRATE SIGNAL PROCESSING” is a bona-fide work
carried out by RAJESH PATIL H.S of 8th Semester bearing the
USN No. 1MV07TE039 in partial fulfillment for the award of
degree of Bachelor of Engineering in Telecommunication
Engineering as specified by Visvesvaraya Technological
University, Belgaum during the academic year 2011. It is
certified that all the corrections/suggestions indicated for
Internal Assessment have been incorporated in the report. The
report has been approved as it satisfies the academic
requirements of Seminar Work prescribed for the Bachelor of
Engineering degree.

Sri A.R. Ravindra Prof. K.R. Kini

Lecturer Professor and Head of the Department
Department of Telecommunication Department of Telecommunication
Engineering Engineering
Sir M.V.I.T. Sir M.V.I.T.
 ACKNOWLEDGEMENTS

I wish to extend my sincere and respectful gratitude to Prof. M.S. Indira, Principal, Sir
M.V.I.T. for permitting me to carry out my technical seminar and her encouragement
throughout the course.

I would like to take this opportunity to thank Prof. K.R. Kini, Head of the Department,
Department of Telecommunication Engineering, Sir M.V.I.T., Bangalore, for his cheerful
encouragement and valuable suggestions. His motivation, encouragement, guidance
and commitment to his belief in strengthening our fundamentals have been
instrumental to our performance in all aspects.

I would like to thank Sri A.R. Ravindra for his enlightening guidance and his rigorous
clarification and teaching of concepts without which this report would not be in its
present state.

I would like to thank all the staff members of our department for their unfledged
support and co-operation throughout the course as well as their dedication to imparting
quality knowledge.

Last but never the least I thank my family who have worked hard every single time
and helped me reach my goals without the slightest discomfort and to all my friends
helped me for the completion of the seminar successfully.

RAJESH PATIL H.S

[1MV07TE039]
 ABSTRACT

Digital signal processing has become one of the most important methods to handle information.
The rapid development of different types of large market products such as mobile phones, DVD’s , etc could
not have been possible without modern DSP methods.

Multi-rate processing and sample rate conversion, or interpolation and decimation as they are a
clever digital signal processing (DSP) techniques that broadband and wireless design engineers can employ
during the system design process. Using these techniques, design engineers can gain an added degree of
freedom that could improve the overall performance of system architecture.

Multi-rate processing finds use in signal processing systems where various sub-systems with
differing sample or clock rates need to be interfaced together. At other times multi-rate processing is used to
reduce computational overhead of a system. For example, an algorithm requires k operations to be completed
per cycle. By reducing the sample rate of a signal or system by a factor of M, the arithmetic bandwidth
requirements are reduced from kfs operations to kfs/M operations per second.

In single-rate systems, only one sampling rate is used throughout a digital signal processing
system, whereas in multirate systems the sampling rate is changed at least once. The rapid development of
multirate digital signal processing is complemented by the emergence of new applications. These include sub-
band coding of speech, audio, and video signals, multicarrier data transmission, fast transforms using digital
filter banks and discrete wavelet analysis of all types of signals.
 LIST OF CONTENTS

Topic Page No.

Certificate 2
Abstract 4
Contents 5
List of Figures 6
List of Abbreviations 7

I. Introduction 8
II. Need for MRSP 11
III. Basics of Signal processing and MRSP 13
IV. Up-Sampler used for MRSP 16
V. Down-Sampler used for MRSP 18
VI. Sampling Rate Conversion by Non-Integer Factor 20
L/M
VII. Filter Banks used for MRSP 21
VIII. Advantages and Disadvantages of MRSP 24
IX. Applications of MRSP 25
X. Summary 27
XI. References 28
 LIST OF FIGURES
FIGURE TITLE OF FIGURE PAGE
NO NO
1 Block diagram for analog signal processing using DSP 14
2 Up-sampler 16
3 Example of Up-sampler with factor L=3 16
4 Relation between and for L=2 17
5 Down-sampler 18
6 Example for down-sampling with factor M=3 18
7 Relation between and for M=2 19
8 Sampling rate conversion by a factor L/M 20
9 44.1 KHz to 48 KHz sampling rate conversion 20
10 Two channel multirate filter bank 21
11 Analysis filter banks 22
12 Synthesis filter bank 23
13 Two channel QMF structure 23
14 Magnitude response of Analysis and Synthesis filters 23
15 Digital to analogue conversion for a CD player using x8 26
oversampling
16 Oversampling in CD music signal reconstruction 26
 LIST OF ABBREVIATIONS
MRSP MultiRate Signal Processing
DSP Digital Signal Processing
DAT Digital Audio Tape
NTSC National Television Systems Committee
PAL Phase Alternate Line
SDR Software Defined Radio
SCA Software Communications Architecture
ADC Analog to digital converter
DAC Digital to analog converter
fs Sampling Frequency
QMF Quadrature Mirror Filter
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I. INTRODUCTION

Digital Signal Processing has become essential to the design and

implementation of high performance audio, video, multi-media, and communication
systems signal processing. An essential component of cost effective DSP algorithms is
multirate signal processing. Many of us (mistakenly) believe that digital filters are simply
sampled data counterparts of linear time-invariant analog prototype filters. The digital
world is richer than this and offers us easy access to filters with time varying coefficients
to perform DSP tasks that defy intuition, and many misconceptions related to aliasing.
Such filters offer extremely efficient structures to simultaneously perform digital
filtering, spectral translation, interpolation, and decimation in both non-recursive and
recursive structures.

Multirate systems have gained popularity since the early 1980. Multi-rate
signal processing studies digital signal processing systems which include sampling rate
conversion. Multirate signal processing techniques are necessary for systems with
different input and output sample rates, but may also be used to implement systems with
equal input and output rates. In multirate digital signal processing the sampling rate of a
signal is changed in order to increase the efficiency of various signal processing
operations. Decimation, or Down-sampling, reduces the sampling rate, whereas
expansion, or up-sampling, followed by interpolation increases the sampling rate.

In most applications multirate systems are used to improve the

performance, or for increased computational efficiency. A key characteristic of multirate

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algorithms are their high computational efficiency. In many cases, these algorithms are
the prime reason that an application can now be implemented economically using modern
digital signal processors.

Multirate signal processing is used for the practical applications in signal

processing to save costs, processing time, and many other practical reasons.

Some applications of multirate signal processing are:

• Up-sampling, i.e., increasing the sampling frequency, before D/A conversion
in order to relax the requirements of the analog low pass antialiasing filter.
This technique is used in audio CD, where the sampling frequency 44.1 kHz is
increased fourfold to 176.4 kHz before D/A conversion.

• Various systems in digital audio signal processing often operate at different

sampling rates. The connection of such systems requires a conversion of
sampling rate.

• Decomposition of a signal into M components containing various frequency

bands. If the original signal is sampled at the sampling frequency fs (with a
frequency band of width fs=2, or half the sampling frequency), every
component then contains a frequency band of width 1/2fs=M only, and can be
represented using the sampling rate fs=M. This allows for efficient parallel
signal processing with processors operating at lower sampling rates. The
technique is also applied to data compression in sub band coding, for example
in speech processing, where the various frequency band components are
represented with different word lengths.

• In the implementation of high-performance filtering operations, where a very

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narrow transition band is required. The requirement of narrow transition bands

leads to very high filter orders. However, by decomposing the signal into a
number of sub bands containing the pass band, stop band and transition bands,
each component can be processed at a lower rate, and the transition band will
be less narrow. Hence the required filter complexity may be reduced
significantly.

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II. NEED FOR MRSP

Communication systems make liberal use of multirate filters in several

ways. Multirate processing finds application in shaping filters, in channelizers, in
interpolators, in efficient bandwidth and sample rate reduction schemes, in anti-alias
filtering, and in many other applications. Multirate signal processing has had a significant
influence at the physical or hardware layer of modern communication systems. In
particular, multirate signal processing is found at the core of communication systems that
couple the Software Defined Radio (SDR) and Software Communications Architecture
(SCA) to reconfigure system resources for operation over a wide range of modulation
formats and waveforms.

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.
Multirate can offer several advantages:
• Reduced computational complexity
• Reduced transmission data rate.

The basic needs of multirate signal processing is that the need for different
sampling rates that are employed in a process. Some examples are as follows:
• Digital audio: Three different sampling rates are employed they are
• Broadcasting requires 32 KHz.
• Digital compact disc requires 44.1 KHz.
• Digital audio tape requires 48 KHz.

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• Digital video:
• Luminance signal is sampled at 13.5MHz.
• Colour difference signal is sampled at 6.75MHz.
• Sampling rate of
• NTSC composite signal is 14.31818MHz.
• PAL composite signal is 17.73447MHz.

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III. BASICS OF SIGNAL PROCESSING AND MRSP

SIGNALS:
A signal is a function of independent variables such as time, speech,
position, temperature, and pressure etc… For example, speech and music signal represent
air pressure as a function of time at a point in space. A black and white picture is a
representation of light intensity as a function of two spatial coordinates. The video signal
in television consists of a sequence of images, called frames and is a function of three
variables: two spatial coordinates and time.
A signal is formally defined as a function of one or more variables which
convey the information on the nature of physical phenomenon.

TYPES OF SIGNAL:
Signals are classified on different basis some of them are given bellow:
 Continuous and discrete time signals
 Periodic and non-periodic signals
 Deterministic and random signals
 Even and odd signals
 1D,2D,3D signals

PROCESSING AND ITS TYPES:

Processing in terms of signals can be defined as the alteration of signal by
operating different operations (such as scaling, addition, multiplication, integration,
differentiation, translation, filtering) on it for an better application.

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Mainly there are two types of signal processing they are:

• Analog signal processing
• Digital signal processing
Processing of continuous and discrete signals are respectively called as
analog and digital signal processing. In case of analog signals most signal processing
operations are done in time domain, whereas, in case of digital signal it can be done both
in time domain as well as in frequency domain. Processing of analog signals in frequency
domain is too difficult so the signals are converted into digital signals using sampling
theorem and then processed and converted back to analog form.

Figure 1: Block diagram for analog signal processing using DSP

MULTIRATE SIGNAL PROCESSING (MRSP):

In many practical applications of digital signal processing, one is faced

with the problem of changing the sampling rate of a digital signal, either increasing it or
decreasing it by some amount. For example, in telecommunication systems that transmit
and receive different types of signals (e.g., teletype, facsimile, speech, video, etc...), there
is a requirement to process the various signals at different rates commensurate with the
corresponding bandwidths of the signals. The process of converting a signal from a given
rate is called sampling rate conversion. In turn, systems that employ multiple sampling
rates in processing of digital signals are called multirate digital signal processing systems.

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Multirate signal processing techniques are necessary for systems with

different input and output sample rates, but may also be used to implement systems with
equal input and output rates. In multirate digital signal processing the sampling rate of a
signal is changed in order to increase the efficiency of various signal processing
operations. Decimation, or Down-sampling, reduces the sampling rate, whereas
expansion, or up-sampling, followed by interpolation increases the sampling rate.

As in many applications, cascade connections of the basic sampling rate

alteration devices and digital filters are employed. For sampling rate alterations, the basic
sampling rate alteration devices are invariably employed together with lowpass digital
filters.
To achieve different sampling rates at different stages, multirate digital
signal processing systems employ the down-sampler and up-sampler, the two basic
sampling rate alteration devices in addition to the conventional elements such as adders,
the multiplexers, and the delay. Discrete-time systems with unequal sampling rate at
various parts of the system are called multirate systems. Up-sampler and down-sampler’s
detailed discussions are done in the next chapter. Many multirate systems employ a bank
of filters with either a common input or a summed output. These filter banks are
discussed in subsequent chapter.

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IV. UP-SAMPLER USED FOR MRSP

Up-sampler is used to increase the sampling rate by an integer factor. An

up-sampler with a 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]. Up-sampler is linear but time-variant discrete time systems.

Figure 2: Up-sampler
Up –sampling operation is implemented by inserting L-1 equidistant zero-
valued samples between two consecutive samples of X[n].
The input-output relation is given by the following equation:
 x[n /L], n = 0, ± L, ± 2 L,
xu [n] = 
 0, otherwise
Figure below shows the up-sampling by a factor of 3

Figure 3: Example of Up-sampler with factor L=3

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In practice, the zero-valued samples inserted by the up-sampler are replaced

with appropriate nonzero values using some type of filtering process and Process is called
interpolation and will be discussed later.

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
In terms of the z-transform, the input-output relation is then given by
∞ ∞
X u ( z) = ∑ x [ n] z
n = −∞
u
−n
= ∑ x[n / 2] z
n = −∞
−n

n even

=
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 jω , the input-output relation is given by

X u (e jω ) = X (e jωL )
jω
Figure below shows the relation between X (e jω ) and X u (e ) for L = 2 in the case of a
typical sequence x[n]

jω
Figure 4: Relation between X (e jω ) and X u (e ) for L=2

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V. DOWN-SAMPLER USED FOR MRSP

Down-sampler is used to decrease the sampling rate by an integer factor. 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]. Down-sampler is linear but time-variant discrete time systems.

Figure 5: Down-sampler
Down–sampling operation is implemented by keeping every M-th smaple of
x[n] and removing M-1 in-between samples to generate y[n].
The input-output relation is given by the following equation:
y[n] = x[nM]
Figure below shows the down-sampling by a factor of 3

Figure 6: Example for down-sampling with factor M=3

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Frequency-Domain Characterization:
Applying the Z-transform to the input-output relation of a factor M down-
sampler Y[n]=x[Mn]
We get
Y(z)=
The expression on the right-hand side cannot be directly expressed in terms
of X(z). To get around this problem, define a new sequence xint [ n.]Where

x[ n], n = 0, ± M , ± 2 M , 
xint [ n] = 
 0, otherwise
Now xint [ n] can formally related to x[n] by the equation xint [n] = c[ n] ⋅ x[ n]
Where 1, n = 0, ± M , ± 2 M , 
c[ n] = 
0, otherwise

then applying Z-transform and reducing finally we get the following equation:

∑ X ( zW )
M −1
1 −k
X int ( z ) = M
M k =0

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

1
related as Y (e jω ) = { X (e jω / 2 ) + X (−e jω / 2 )}
2
jω / 2
Figure below shows the relation between X (e jω ) and X (e ) for M = 2

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jω / 2
Figure 7: Relation between X (e jω ) and X (e ) for M = 2

VI. SAMPLIN RATE CONVERSION BY A NON INTEGER

FACTOR L/M

With an understanding of the down-sampling and up-sampling processes, we

now study the sampling rate conversion by a non-integer factor of L/M. This can be
viewed as two sampling conversion processes. In step 1, we perform the up-sampling
process by a factor of integer L following application of an interpolation filter H1(z); in
step 2, we continue filtering the output from the interpolation filter via an anti-aliasing
filter H2(z), and finally operate down-sampling. The entire process is illustrated in the
below figure.

Figure 8: Sampling rate conversion by a factor L/M

Since the interpolation and anti-aliasing filters are in a cascaded form and
operate at the same rate, we can select one of them. We choose the one with the lower
stop frequency edge and choose the most demanding requirement for passband gain and
stopband attenuation for the filter design. A lot of computational saving can be achieved
by using one lowpass filter. Let us see one example of CD to DAT form conversion. The
sampling rate in CD is 44.1 kHz and in DAT is48 KHz.
Now the question is how to convert 44.1 KHz data to 48 KHz data? This is as
shown below: 48 / 44.1 = 160 / 147

Figure 9: 44.1 KHz to 48 KHz sampling rate conversion

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VII. FILTER BANKS USED FOR MRSP

Introduction:

In signal processing, a filter bank is an array of band-pass filters that

separate the input signal into multiple components, each one carrying a single frequency
sub-band of the original signal. One application of a filter bank is a graphic equalizer,
which can attenuate the components differently and recombine them into a modified
version of the original signal. The process of decomposition by the filter banks is called
as analysis, the output reconstruction process is called synthesis. Some other applications
of filter banks are found in signal compression, vocoder etc. . . .

A basic multirate filter bank is shown in below Figure. Multirate filter banks
are so named because they effectively alter the sampling rate of a digital system, as
indicated by the decimators (down-samplers) following the analysis filters, A0 and A1,
and the expanders (up-samplers) preceding the synthesis filters, S0 and S1. Properly
designed analysis and synthesis filters combined with the properties of decimation and
expansion allow filter banks to partition a wideband input signal into multiple frequency
bands (often called sub-bands or channels) and to recombine these sub-band signals back
into the original signal.

Figure 10: Two channel multirate filter bank

The detailed study of the analysis and synthesis filter and qudrature mirror
filters (QMF) are discussed briefly in the next sections.

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Analysis and Synthesis filters:

A basic operation in multirate signal processing is to decompose a signal
into a number of sub-band components, which can be processed at a lower rate
corresponding to the bandwidth of the frequency bands. Down-sampling mixes frequency
components in the original signal by aliasing and frequency folding. Therefore, the signal
should be filtered before decimation. The below figure shows the decomposition of a
signal into two sub-band components. The purpose of the filters H1 and H2 is to extract
the low- and high-frequency components of the signal x before decimation. The set of
filters shown in below figure is called an analysis filters or analysis filter banks. If
required, the signals may be decimated further into narrower sub-band components.

A convenient way to implement the decimation is to use stages with the

decimation factor M= 2 as shown in figure. Then only one low-pass and one high-pass
filter is required. For M>2, band-pass filters with different Pass-bands are required as
well. The sub-band components obtained from analysis filter are then allowed for
processing.

Figure 11: Analysis filter banks

After processing of the separate sub-band components, they are combined to
reconstruct (a properly processed version of) the original signal at the original, higher
sampling rate. Up-sampling generates aliasing frequencies. Therefore, the expanded
signals should be filtered in order to extract the correct frequency components. The set of

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filters used to reconstruct the desired signal is called synthesis filter or synthesis filter
banks.

The below figure shows the block diagram of synthesis filters:

Figure 12: Synthesis filter bank

Qudrature Mirror Filter (QMF):
The basic building block in applications of Qudrature mirror filters (QMF) is
the two-channel QMF bank as shown in the below figure. This is a multirate digital filter
structure that employs two decimators in the “signal analysis” section and two
interpolators in the “signal synthesis” section. The low-pass and the high-pass filters in
analysis section have impulse responses H0(z) and H1(z), respectively. Similarly, the
low-pass and high-pass filters contained in the synthesis section have impulse response
F0(z) and F1(z), respectively.

Figure 13: Two channel QMF structure

The analysis and the synthesis filters in the above figure are typically
complementary low-pass and high-pass filters that mirror each other about the digital

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frequency, /2, as shown in below figure. Such filters are often called quadrature mirror

filters (QMF), since /2 correspond to one fourth the sampling frequency.

VIII. ADVANTAGES AND DISADVANTAGES OF MRSP

Advantages of MRSP are as follows:

• Multistage design yields significant reductions in computation & storage
requirements compared to single stage.
• Reductions are due to wide transition bands of filters at early stages (even though
sampling rates are high) leading to small values of N.
• In the last stage though the transition band is small, the sampling rate is also low,
hence filter order (N) is also small as compared to single stage decimation.
• Sub-band coding allows parallel processing.
• Improve/increase convergence speed in adaptive filtering applications (e.g. echo
cancellation and adaptive equalization).

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IX. APPLICATIONS OF MRSP

• Sub-band coding of speech signals.
• Implementation of wavelets.
• Used in source coding and compression for contemporary communication
applications.
• Used in digital transmultiplexer.
• Over sampling A/D and D/A conversion.
• CD to DAT format change.
• Found applied in image compression.
• Used in adaptive equalization.
• Used in echo cancellation.
• Used in adaptive beamforming.
• Digital audio and vedio.
• Applied in code division multiple access.
• Found applied in demodulation in vision based navigation system sensors.

Let us study one of application in brief i.e. .Multirate systems used in a CD

player when the music signal is converted from digital to analog (DAC). Digital data (16-
bit words) are read from the disk at a sampling rate of 44.1 kHz. If this data were
converted directly into an analog signal, image frequency bands centred on multiples of

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the sampling-rate would occur, causing amplifier overload, and distortion in the music
signal. To protect against this, a common technique called oversampling is often
implemented nowadays in all CD players and in most digital processing systems of music
signals. Figure below shows a CD player and how oversampling is utilized.It is
customary to oversample (or expand) the digital signal by a factor of x8, followed by an
interpolation filter to remove the image frequencies. The sampling rate of the resulting

signal is now increased up to 352.8 kHz. The digital signal is then converted into an
analogue waveform by passing it through a 14-bit DAC. Then the output from this device
is passed through an analogue low-pass filter before it is sent to the speakers.

Figure 15: Digital to analogue conversion for a CD player using x8 oversampling.

Below figures illustrates the procedure of converting a digital waveform into
an analogue signal in a CD player using x8 oversampling. As an example, Figure (a)
illustrates a 20 kHz sinusoidal signal sampled at 44.1 kHz, denoted by x[n]. The six
samples of the signal represent the waveform over two periods. If the signal x[n] was
converted directly into an analogue waveform, it would be very hard to exactly
reconstruct the 20 kHz signal from this diagram. Now, Figure (b) shows x[n] with an x8
interpolation, denoted by y[n]. Figure (c) shows the analogue signal y(t), reconstructed
from the digital signal y[n] by passing it through a DAC. Finally, Figure (d) shows the
waveform of z(t), which is obtained by passing the signal y(t) through an analogue low-
pass filter.

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Figure 16: Oversampling in CD music signal reconstruction

X. SUMMARY

Interpolators, in their various forms, are used in most signal processing

applications. The obvious example is the estimation of a sequence of missing samples.
However, the use of an interpolator covers a much wider range of applications, from low-
bit-rate speech coding to pattern recognition and decision-making systems. Conventional
theories of digital signal processing assert that the ideal filter is the best for interpolation
or decimation. However, as we have shown above, a sharp filter characteristic
approximating the ideal filter does not necessarily behave well. In particular, such a filter
often exhibits a large amount of ringing as illustrated in the previous section. The ringing
is due to the Gibbs phenomenon, which is caused by the sharp characteristic of the filter.
On the other hand, our filter shows a slow decay. The reason is that due to the underlying
analog characteristic, there is important information content beyond the Nyquist
frequency, and such a slow decay is necessary to recover such information. Moreover,
conventional design requires us to give a filter order in advance. The higher the order is,
the closer to the ideal characteristic the filter is, and hence filters of a very high order are
often used.
Interpolators are used to increase the sampling rate. Whereas decimators used
to decreases the sampling rate. Non-integer sampling rates are also obtained easily as

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discussed above. Different types of multirate filters are utilized to change the sampling
rate as it is required for the applications.

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XI. REFERENCES

• P.P Vaidyanathan. Multirate digital filters, filter banks, polyphase networks, and
applications: A tutorial. Proc. IEEE, 78(1):56_93, January 1990.
• Digital signal processing, fundamentals and application by Li Tan.
• Digital signal processing, Principles, Algorithms, and Applications by Jhon G.
Proakis and Dimitris G. Manolakis.
• Digital signal processing by sanjit K. mitra.
• Crochiere, Ronald E. and Rabiner, Lawrence R., “Multirate Digital Signal
Processing”, Prentice-Hall, Inc., 1983.
• R. Ansari and B. Liu, “Multirate signal processing,” in Handbook for Digital
Signal Processing, S. K. Mitra and J. F. Kaiser, Eds., chapter 14, pp. 981–1084.
New York: JohnWiley and Sons, 1993.
• E. C. Ifeachor and B. W. Jervis, Digital Signal Processing, A Practical Approach,
Addison-Wesley, 1993.

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