On Filter Bank Based MIMO Frequency Multiplexing and

Demultiplexing
Master thesis performed in Electronics Systems division
by
Amir Eghbali
Report number:LiTH-ISY-EX--06/3911--SE
September 2006

ii

Title
On Filter bank Based MIMO Frequency Multiplexing and
Demultiplexing
Master thesis in Electronics Systems
at Linkping Institute of Technology
by
Amir Eghbali
LiTH-ISY-EX--06/3911--SE

Supervisor: Prof. Hkan Johansson

Examiner: Prof. Hkan Johansson
Linkping: 26 September 2006

iii

iv

Presentation Date
2006-09-26
Publishing Date (Electronic
version)
2006-10-02
Language

English

Division of Electronics
Systems
Department of Electrical
Engineering

Type of Publication

Number of Pages

95

Licentiate thesis
Degree thesis
Thesis C-level
Thesis D-level
Report
Other (specify below)

ISBN (Licentiate thesis)

ISRN: LiTH-ISY-EX--06/3911SE
Title of series (Licentiate thesis)

Series number/ISSN (Licentiate thesis)

URL, Electronic Version

http://www.ep.liu.se

Publication Title
On Filter Bank Based MIMO Frequency Multiplexing and Demultiplexing
Author
Amir Eghbali
Abstract
The next generation satellite communication networks will provide multimedia services supporting high bit rate, mobility,
ATM, and TCP/IP. In these cases, the satellite technology will act as the internetwork infrastructure of future global systems and
assuming a global wireless system, no distinctions will exist between terrestrial and satellite communications systems, as well as
between fixed and 3G mobile networks. In order for satellites to be successful, they must handle bursty traffic from users and
provide services compatible with existing ISDN infrastructure, narrowcasting/multicasting services not offered by terrestrial
ISDN, TCP/IP-compatible services for data applications, and point-to-point or point-to-multipoint on-demand compressed video
services. This calls for onboard processing payloads capable of frequency multiplexing and demultiplexing and interference
suppression.
This thesis introduces a new class of oversampled complex modulated filter banks capable of providing frequency
multiplexing and demultiplexing. Under certain system constraints, the system can handle all possible shifts of different user
signals and provide variable bandwidths to users. Furthermore, the aliasing signals are attenuated by the stopband attenuation of
the channel filter thus ensuring the approximation of the perfect reconstruction property as close as desired. Study of the system
efficient implementation and its mathematical representation shows that the proposed system has superiority over the existing
approaches for Bentpipe payloads from the flexibility, complexity, and perfect reconstruction points of view. The system is
analyzed in both SISO and MIMO cases. For the MIMO case, two different scenarios for frequency multiplexing and
demultiplexing are discussed.
To verify the results of the mathematical analysis, simulation results for SISO, two scenarios of MIMO, and effects of the
finite word length on the system performance are illustrated. Simulation results show that the system can perform frequency
multiplexing and demultiplexing and the stopband attenuation of the prototype filter controls the aliasing signals since the filter
coefficients resolution plays the major role on the system performance. Hence, the system can approximate perfect reconstruction
property by proper choice of resolution.

Number of pages: 95
Keywords
Frequency Band Reallocation, Filter Bank, Multirate Signal Processing, MIMO, Satellite Communications

vi

Abstract
The next generation satellite communication networks will provide multimedia
services supporting high bit rate, mobility, ATM, and TCP/IP. In these cases, the
satellite technology will act as the inter-network infrastructure of future global systems
and assuming a global wireless system, no distinctions will exist between terrestrial and
satellite communications systems, as well as between fixed and 3G mobile networks. In
order for satellites to be successful, they must handle bursty traffic from users and
provide
services
compatible
with
existing
ISDN
infrastructure,
narrowcasting/multicasting services not offered by terrestrial ISDN, TCP/IP-compatible
services for data applications, and point-to-point or point-to-multipoint on-demand
compressed video services. This calls for onboard processing payloads capable of
frequency multiplexing and demultiplexing and interference suppression.
This thesis introduces a new class of oversampled complex modulated filter
banks capable of providing frequency multiplexing and demultiplexing. Under certain
system constraints, the system can handle all possible shifts of different user signals and
provide variable bandwidths to users. Furthermore, the aliasing signals are attenuated by
the stopband attenuation of the channel filter thus ensuring the approximation of the
perfect reconstruction property as close as desired. Study of the system efficient
implementation and its mathematical representation shows that the proposed system has
superiority over the existing approaches for bentpipe payloads from the flexibility,
complexity, and perfect reconstruction points of view. The system is analyzed in both
Single Input single Output (SISO) and Multiple Input Multiple Output (MIMO) cases.
For the MIMO case, two different scenarios for frequency multiplexing and
demultiplexing are discussed.
To verify the results of the mathematical analysis, simulation results for SISO,
two scenarios of MIMO, and effects of the finite word length on the system
performance are illustrated. Simulation results show that the system can perform
frequency multiplexing and demultiplexing and the stopband attenuation of the
prototype filter controls the aliasing signals since the filter coefficients resolution plays
the major role on the system performance. Hence, the system can approximate perfect
reconstruction property by proper choice of resolution.

vii

viii

Acknowledgments
First, I would like to thank my supervisor Prof. Hkan Johansson
for the invaluable guidance and incredible patience in answering my
questions. I could ask any questions at any time.
Special thanks go to my family for all the support they provided. I
will never forget their kindness.
I would also like to thank all the apples and bananas that kept me
alive during the time I was working on my thesis!

ix

Foreword
The next generation information society will include
telecommunications, computing, video, TV, videoconferencing, and
consumer electronics in every building and requires wideband services to
provide multi-application networks at rates around 2 Mbps accessible to
everybody everywhere [1]. The terrestrial networks, even with the large
bandwidth available due to optical fiber technology, cannot meet these
requirements. However, satellites play an important role since if a
satellite is in orbit, the subscriber only has to install a satellite terminal
and subscribe to the service. To solve the problem of the next generation
networks, network technicians suggest asynchronous transfer mode
(ATM) comprised of a multiplexer with a high-rate output having every
possible lower rate at the input side. On the other hand,
telecommunications managers try to provide temporary solutions such as
asynchronous digital subscriber line (ADSL) and high-rate DSL (HDSL)
[1].
One of the disadvantages of geostationary communications
satellites, is the large delay for one up- and downlink, which is disturbing
for voice. However, the terrestrial copper, optical fiber, and the cellular
radio networks carry most voice traffic. Thus, the satellites can be a
suitable choice for interactive data services and delivery of a large
amount of data on request. In addition, low earth orbit (LEO) systems
such as GLOBALSTAR and ICO are competing with the terrestrial
networks for voice applications. Therefore, for wideband multimedia
applications, geostationary satellites with several high-gain spot-beam
antennas, OnBoard Processing (OBP), and switching seem to be a logical
step in migration from pure TV broadcast to interactive multimedia
services. The functionality that the OBP system offers is suited to
provide the services required by the information society. The elements
making up the OBP system are [1]:
The User Station (UTS): The UTS consists of an outdoor unit and
an indoor unit with a capability of being equipped with Integrated
Services Digital Network (ISDN), Electronic Network Systems
(ENS), and packet switch (TCP/IP) interface.
The Master Control Center (MCC): The MCC translates the
subscriber terminals protocols and algorithms into commands. It
also controls the communication flow inside the broadband
satellite communications network. This block is also responsible
xi

for the compatibility of the new networks with the existing

protocols and algorithms.
The switching payload: The payload consists of DSP functions
such as digital beamforming, frequency multiplexing and
demultiplexing, interference suppression, signal level control and,
in a regenerative system, modems [2].
This thesis focuses on digital signal processing of satellite payloads
which has two major categories as [2]:
1. Onboard regeneration and baseband processing: Examples
for this type are data buffering and multiple access
reformatting, data rate conversion, coding, and encryption.
These systems decouple noise and interference on the
uplink and downlink and are able to optimize access,
modulation, and coding techniques for the uplink and
downlink.
2. Onboard non-regenerative processing: Here, signals are
sampled with appropriate precision and sampling rate.
Subsequent processing is performed as arithmetic
operations on the signal samples. In particular, such
techniques allow the digital demultiplexing of narrowband
channels and processing of individual channels to include
level control and beamforming. Hence, we need a
transparent payload architecture where signals are not
regenerated onboard. The system level advantages of this
system are power efficiency, frequency reuse, flexibility
in response to changing traffic, reproducibility, and lack
of sensitivity to temperature changes [2].
This thesis proposes a bentpipe payload architecture that handles all
possible frequency shifts and all possible user data rates, has low
complexity, achieves high level of parallelism, and is easy to analyze and
design. The system uses a new class of oversampled complex modulated
Filter Banks (FB), which brings superiority over previously proposed
architectures. In particular, it outperforms the regular modulated FB
based networks from the flexibility point of view and has better
performance over the tree-structured FB based networks in terms of
flexibility and complexity. Furthermore, the proposed system
outperforms the overlap/save DFT/IDFT based networks if perfect
reconstruction property is important.
The key features of the proposed system are:

xii

Use of oversampled filter banks: This choice makes the

suppression of aliasing easier and allows the combination of
smaller subbands into wider subbands without introducing large
aliasing distortion. This property brings full flexibility to the
system.
More FB channels than granularity bands: This feature brings the
ability to generate all possible frequency shifts and reduces the
complexity of the system.
Complex modulated filter banks: These filter banks result in very
low complexity and simplicity in terms of analysis, design, and
implementation.
The report is organized in three chapters. In the first chapter, basic
building blocks of multirate systems i.e. interpolators, decimators, and
polyphase decomposition are introduced. Since the proposed structure
uses filter banks, building blocks of filter banks and their mathematical
representations are derived. Based on the structure and parameters, the
maximally decimated and oversampled filter banks are discussed. Next,
the concept of paraunitariness followed by DFT and cosine modulated
filter banks in maximally decimated systems is covered. The distinction
between uniform and non-uniform filter banks is treated mathematically
but the focus is on the uniform filter banks. The oversampled filter bank
analysis starts with the definition of frame theory followed by the
example on oversampled DFT modulated filter banks. Having discussed
the time invariant systems, basics and properties of the time varying filter
banks are investigated. The chapter ends with common issues in design
of filter banks from a system point of view described as constraints in a
hierarchical manner.
The second chapter discusses the basics of transmultiplexers, as
duals of filter banks, and derives their mathematical representation. Next,
perfect reconstruction, cancelling of multiuser and interblock
interference, and channel equalization are discussed. As special cases of
transmultiplexers, the multiple access schemes such as Code Division
Multiple Access (CDMA), Time Division Multiple Access (TDMA), and
Frequency Division Multiple Access (FDMA) are introduced. Having
discussed the transmultiplexers, different architectures of payloads i.e.,
bentpipe, partial processing, full processing, and hybrid systems used in
satellite applications and their features are studied. The chapter ends with
a review of filter bank applications in payload systems.
The third chapter illustrates the proposed system for frequency
multiplexing and demultiplexing. First, the problem of frequency
xiii

multiplexing and demultiplexing is formulated followed by the

introduction to a new class of online variable oversampled complex
modulated filter banks. Based on the problem formulation and the filter
bank definition, the constraints of the architecture are derived. Next, the
characteristics of the filter bank blocks namely analysis/synthesis banks
and channel switch are defined. In order to decrease the implementation
complexity, the polyphase decomposition is applied to derive the new
system architecture. In reality, there are several users in the uplink which
must be multiplexed to different downlink spot beams. This calls for a
MIMO system capable of performing the multiplexing and
demultiplexing and satisfying the defined Perfect Reconstruction (PR)
properties. The extension of the proposed system to a MIMO case is
covered in two scenarios.
The last part of the chapter illustrates simulation results of the proposed
architecture from the functionality and performance points of view. To
do so, the system test setup and the error measurement algorithm which
is Mean Square Error (MSE) are described. Next, examples on SISO and
MIMO cases verifying the system functionality to multiplex and
demultiplex signals are presented. To evaluate the system performance,
the finite word length effects are introduced and examples for a 64-QAM
signal with different resolutions are illustrated. The chapter ends with
conclusion and topics for future research.

xiv

Outline of Tasks
The tasks assigned in this thesis work were as follows:
1.
2.
3.
4.

Study of the multirate signal processing basics.

Literature review on different filter bank architectures.
Literature review on different satellite payload systems.
Implementation of a MIMO polyphase Frequency Band
Reallocation network in MATLAB including the finite word
length effects.
5. Evaluation of the MIMO polyphase network from the BER point
of view.

xv

xvi

Table of Contents
ABSTRACT ................................................................................................................VII
ACKNOWLEDGMENTS........................................................................................... IX
FOREWORD ............................................................................................................... XI
OUTLINE OF TASKS ...............................................................................................XV
LIST OF ABBREVIATIONS .................................................................................. XXI
CHAPTER ONE: OVERVIEW OF MULTIRATE SYSTEMS AND FILTER
BANKS ............................................................................................................................1
1.

INTRODUCTION ..................................................................................3
1.1.
BASIC BUILDING BLOCKS OF MULTIRATE SYSTEMS ..................................... 3
1.1.1. Polyphase Decomposition........................................................................ 5
1.2.
DIGITAL FILTER BANKS ................................................................................ 9
1.2.1. Analysis Filter Bank................................................................................. 9
1.2.2. Downsamplers ....................................................................................... 10
1.2.3. Subband Processing............................................................................... 10
1.2.4. Upsamplers ............................................................................................ 11
1.2.5. Synthesis Filter Bank ............................................................................. 11
1.3.
GENERAL FILTER BANK ARCHITECTURE ..................................................... 11
1.4.
MAXIMALLY DECIMATED FILTER BANKS ................................................... 13
1.5.
PARAUNITARY FILTER BANKS ..................................................................... 16
1.5.1. Properties of Paraunitary PR Filter banks............................................ 17
1.6.
DFT MODULATED FILTER BANKS ............................................................... 18
1.6.1. Uniform and Non-uniform Filter Bank .................................................. 18
1.6.2. Uniform DFT Modulated Filter Banks .................................................. 19
1.7.
COSINE MODULATED FILTER BANKS .......................................................... 24
1.8.
OVERSAMPLED PR FILTER BANKS .............................................................. 29
1.9.
TIME VARYING FILTER BANKS .................................................................... 34
1.10.
DIFFERENCES BETWEEN TIME VARYING AND LTI FILTER BANKS ............... 38
1.11.
FILTER BANK DESIGN ISSUES ...................................................................... 39
1.11.1.
Filter Issues....................................................................................... 39
1.11.2.
Filter Bank Issues.............................................................................. 39
1.11.3.
Analysis/Synthesis Issues .................................................................. 39
1.11.4.
Total System Issues ........................................................................... 40

CHAPTER TWO: OVERVIEW OF TRANSMULTIPLEXERS AND

SATELLITE PAYLOAD SYSTEMS .........................................................................41
2.

INTRODUCTION ................................................................................43
2.1.
TRANSMULTIPLEXERS ................................................................................. 43
2.1.1. Mathematical Representation of Transmultiplexers .............................. 43
2.1.2. Perfect Reconstruction in Transmultiplexers......................................... 45
2.1.3. Canceling InterBlock Interference in Transmultiplexers....................... 46
2.1.4. Canceling Multi User Interference in Transmultiplexers ...................... 46
2.1.5. Time Frequency interpretation .............................................................. 48
2.1.6. CDMA System Based on Transmultiplexers .......................................... 49

xvii

2.1.7. TDMA System Based on Transmultiplexers........................................... 50

2.1.8. FDMA System Based on Transmultiplexers........................................... 51
2.2.
SATELLITE PAYLOAD ARCHITECTURES ....................................................... 51
2.2.1. Bentpipe Payload................................................................................... 52
2.2.2. Full Processing Payload........................................................................ 52
2.2.3. Partial Processing Payload ................................................................... 53
2.2.4. Hybrid Payload...................................................................................... 54
2.3.
FREQUENCY MULTIPLEXING/DEMULTIPLEXING USING FILTER BANKS ....... 54
CHAPTER THREE: PROPOSED BENTPIPE SYSTEM AND SIMULATION
RESULTS......................................................................................................................55
3.

INTRODUCTION ................................................................................57
3.1.
PROBLEM FORMULATION ............................................................................ 58
3.2.
CLASS OF ONLINE VARIABLE OVERSAMPLED COMPLEX MODULATED FILTER
BANKS 59
3.2.1. System Constraints................................................................................. 59
3.2.2. Constraints on Sampling Rate Converters and Number of Channels .... 60
3.2.3. Analysis Filters ...................................................................................... 61
3.2.4. Synthesis Filters..................................................................................... 62
3.2.5. Application of Switch in the FFBR Network .......................................... 63
3.2.6. Efficient Implementation........................................................................ 64
3.3.
MIMO FFBR NETWORK ............................................................................. 65
3.3.1. K-Input K-Output FFBR Networks ........................................................ 65
3.3.2. S-Input K-Output FFBR Networks......................................................... 66
3.4.
SIMULATION RESULTS ................................................................................. 66
3.4.1. System Parameters Selection ................................................................. 67
3.4.2. Transmitter/Receiver Filter Design ....................................................... 67
3.4.3. Implementation of the SISO System ....................................................... 69
3.4.4. Implementation of the MIMO System..................................................... 71
3.5.
FINITE WORD LENGTH EFFECTS ON THE FFBR NETWORK .......................... 74
3.6.
CONCLUDING REMARKS AND FUTURE TOPICS ............................................ 77

REFERENCES .............................................................................................................79
APPENDIXES...............................................................................................................83
APPENDIX A: MATLAB PROGRAM TO DESIGN THIRD AND SIXTH BAND
FILTERS .......................................................................................................................85
APPENDIX B: MATLAB PROGRAM TO GENERATE USER SIGNALS ..........87
APPENDIX C: MATLAB PROGRAM TO IMPLEMENT THE SYSTEM IN
FIGURE 29....................................................................................................................89
APPENDIX D: MATLAB PROGRAM TO IMPLEMENT THE SYSTEM IN
FIGURE 31....................................................................................................................91
APPENDIX E: MATLAB PROGRAM TO DESIGN PROTOTYPE FILTERS
USING MINIMAX ALGORITHM.............................................................................95

xviii

LIST OF FIGURES

Figure 1: Effect of Aliasing and Imaging in Upsamplers and

Downsamplers........................................................................................... 5
Figure 2: Noble Identities in Multirate Systems. ...................................... 6
Figure 3: Efficient Polyphase Decimator and Interpolator
Implementation. ........................................................................................ 7
Figure 4: Filer Realization Using Subband Decomposition. .................... 8
Figure 5: Typical Analysis and Synthesis Banks...................................... 9
Figure 6: Typical Frequency Responses of Analysis Filters. ................. 10
Figure 7: General Filter Bank Architecture. ........................................... 12
Figure 8: Realization of the Analysis and Synthesis Banks Based on
Polyphase Matrices. ................................................................................ 13
Figure 9: Simplified Realization of Filter Banks Using Noble Identities.
................................................................................................................. 14
Figure 10: Filter Characteristics for Uniform and Non-Uniform Filter
Banks....................................................................................................... 18
Figure 11: Analysis Bank Polyphase Realization of DFT Modulated
Filter Banks. ............................................................................................ 21
Figure 12: Analysis Bank Polyphase Realization of DFT Modulated
Filter Banks. ............................................................................................ 22
Figure 13: Simplest Case of the DFT Modulated Filter Banks.............. 23
Figure 14: Analysis Filters for the Cosine Modulated Filter Banks. ...... 26
Figure 15: Polyphase Realization Analysis Bank for the Cosine
Modulated Filter Banks........................................................................... 26
Figure 16: Architecture of Oversampled DFT Modulated Filter Bank. . 32
Figure 17: Polyphase Realization of the Oversampled DFT Modulated
Filter Bank............................................................................................... 33
Figure 18: General Architecture of Time Varying Filter Banks............. 35
Figure 19: Different Stages of a Time Varying Filter Bank. .................. 35
Figure 20: General Architecture of a Transmultiplexer.......................... 44
Figure 21: Architecture of Transmultiplexer with Transmit and receive
Filters. ..................................................................................................... 45
Figure 22: Modeling the Channel to Cancel InterBlock Interference..... 46
Figure 23: Time Frequency Tilde of a General Discrete Time Function.49
Figure 24: CDMA System Based on Transmultiplexer. ......................... 50
Figure 25: Simple TDMA System Based on Transmultiplexer.............. 50
Figure 26: Transmultiplexer Synthesis/Analysis Filter Characteristics for
FDMA System. ....................................................................................... 51
Figure 27: Illustration of Guard and Granularity Bands in the FFBR
System..................................................................................................... 58
xix

Figure 28: FFBR system with Fixed Analysis and Adjustable Synthesis
Bank. ....................................................................................................... 59
Figure 29: FFBR system with Fixed Analysis/Synthesis Banks and
Channel Switch. ...................................................................................... 63
Figure 30: Polyphase Implementation of the FFBR Network. ............... 64
Figure 31: K-Input K-Output MIMO FFBR with Fixed Analysis and
Synthesis FBs.......................................................................................... 65
Figure 32: S-Input K-Output MIMO FFBR with Fixed Analysis and
Synthesis FBs.......................................................................................... 66
Figure 33: Transmit and Receive Filter Characteristics to Evaluate the
FFBR Network........................................................................................ 68
Figure 34: Test Setup for FFBR Network Evaluation. ........................... 69
Figure 35: Example Channel Switch for SISO Case. ............................. 70
Figure 36: Input, Output, and Analysis Filters for SISO Polyphase FFBR
Network................................................................................................... 71
Figure 37: Example Channel Switch for Two-Input Two-Output MIMO
FFBR Network........................................................................................ 71
Figure 38: Inputs and Outputs for MIMO FFBR Network with two Inputs
and two Outputs. ..................................................................................... 72
Figure 39: Input and Outputs of the FFBR Network without Channel
Switch. .................................................................................................... 73
Figure 40: Example One-Input/Two-Output Channel Switch for MIMO
FFBR Network........................................................................................ 73
Figure 41: Input and Outputs of the FFBR Network with Channel Switch
of Figure 40............................................................................................. 74
Figure 42: Quantization in the Polyphase FFBR Network. .................... 75
Figure 43: Multiplexed 64-QAM Data Constellation for Three Filter
Coefficient Lengths................................................................................. 76
Figure 44: FFBR Network Noise Variance for Channels in Figure 38. . 77

xx

List of Abbreviations
Abbreviation
AFB
ATM
BER
CDMA
DCT
DFT
DSL
DSP
ENS
ESA
FB
FDM
FDMA
FFBR
FIR
GDFT
HDSL
IDFT
IIR
ISDN
ISI
ISP
LEO
LP
LTI
LTV
MCC
MIMO
MSE
MUI
OBP
PFBR
PR
PU
QAM
SFB
SISO
SNR
SS/TDMA
TCP/IP
TDMA
TM
TVFB
UTS
VPN

Comments
Analysis Filter Bank
Asynchronous Transfer Mode
Bit Error Rate
Code Division Multiple Access
Discrete Cosine Transform
Discrete Fourier Transform
Digital Subscriber Line
Digital Signal Processing
Electronic Network Systems
European Space Agency
Filter Bank
Frequency Division Multiplexing
Frequency Division Multiple Access
Flexible Frequency Band Reallocation
Finite Impulse Response
Generalized Discrete Fourier Transform
High bit rate Digital Subscriber Line
Inverse Discrete Fourier Transform
Infinite Impulse Response
Integrated Services Digital Network
Inter Symbol Interference
Internet Service Provider
Low Earth Orbit
Low Pass
Linear Time Invariant
Linear Time Variant
Master Control Center
Multiple Input Multiple Output
Mean Square Error
Multi User Interference
OnBoard Processing
Perfect Frequency Band Reallocation
Perfect Reconstruction
ParaUnitary
Quadrature Amplitude Modulation
Synthesis Filter Bank
Single Input Single Output
Signal to Noise Ratio
Satellite-Switched Time Division Multiple Access
Transmission Control Protocol/Internet Protocol
Time Division Multiple Access
TransMultiplexer
Time Varying Filter Banks
User Station
Virtual Private Network

xxi

xxii

Chapter One: Overview of Multirate Systems

and Filter Banks

1. Introduction
Multirate digital filters and filter banks find wide application in
areas such as speech processing, communications, analog voice privacy
systems, image compression, antenna systems, and digital audio industry.
This applicability has excited immense amount of research leading to a
substantial progress in multirate systems including decimation and
interpolation filters, polyphase structures, and several types of
analysis/synthesis filter banks with specific properties that suit some
applications. To analyze different systems mathematically, it is useful to
have some blocks that are common among the systems and furthermore,
ease the analysis process. In the analysis of the multirate systems and
filter banks, which is the subject of this chapter, the basic building blocks
are the interpolators and decimators, which used along with the concept
of the polyphase decomposition, reduce the implementation complexity.
In this chapter, we start with the definition of these building blocks,
and then we proceed to define the basics of filter bank theory. In this
context, different types of maximally decimated and oversampled filter
banks are discussed. Furthermore, a brief introduction to time varying
filter banks is provided.

1.1.

Basic Building Blocks of Multirate Systems

In the area of the multirate signal processing, interpolators and

decimators are the basic blocks that alter the sampling frequency at
different parts of the system leading to name Multirate. An interpolator
is a combination of an upsampler and a lowpass filter where the
upsampler inserts M 1 zeros between consecutive samples of the
original signal. Doing so, the output signal spectrum is a compressed
version of the input signal spectrum. In the mathematical representation
of an upsampler, we have [3]

Y ( z ) = X ( z M ) or

Y (e j ) = X (e jM ) ,

1.1

where y (n) and x(n) are the output and input sequences, respectively. If
X (e j ) is periodic with 2 , then Y (e j ) will be periodic with 2 [3].
M

On the other hand, a decimator is the combination of a lowpass

filter and a downsampler where the downsampler retains only the Mth
samples of the input signal. In the mathematical representation, assuming
the notations on interpolator, we have
Y ( z) =

1
M

M 1

X (z

1
M

W k ) or Y (e j ) =

k =0

1
M

M 1

X (e

j ( 2 k )
M

), W = e

j 2
M

1.2

k =0

Hence, Y (e j ) is a sum of M uniformly shifted versions of an M fold

stretched version of X (e j ) [3]. An important issue in the analysis of
these blocks is imaging and aliasing.
Looking at Equation (1.2), one can conclude that if x(n) is band
limited to < <
(more generally < < + 2 [3]), the
M

original signal can be recovered from y (n) by the use of a lowpass filter.
Otherwise, the problem of aliasing can occur damaging the information.
So, an interpolator can cause imaging due to compression of the input
signal spectrum, which must be removed by a lowpass filter following
the upsampler.
Similarly, a decimator can cause aliasing due to the stretching of
the input signal spectrum. To deal with this problem, a lowpass filter
must remove unnecessary signals before the downsampler. The imaging
and aliasing effects and the characteristics of the lowpass filters for a
system with a decimation and interpolation ratio of three are shown in
Figure 1.

X ( e j )
Decimation Filter

X (e )

Interpolation Filter

Images to be removed

Aliasing in the absence of the filter

Figure 1: Effect of Aliasing and Imaging in Upsamplers and Downsamplers.

It must be added that in reality, the brick wall filters can not be
realized, so the filters should have transition bands. This can be solved by
considering the fact that data signals are not also strictly band limited
which allows for filters to have transition bands. To reduce the
complexity of the interpolator and decimator implementation, the idea of
polyphase decomposition is used and will be discussed in the next
section.

1.1.1. Polyphase Decomposition

Polyphase decomposition realizes any lowpass filter as the sum of
polyphase components [3]. Any finite or infinite length sequence {h(n)}
with a z-transform H (z ) can be written as [4]

H ( z) =

M 1

n =

k =0

h(n) z n = z k H k ( z M ) = 1 z 1 ... z ( M 1)

H 0 (z M )

M
H1 (z )

.
.

M
H M 1 ( z )

1.3

The right hand side of the Equation (1.3) is called the polyphase
decomposition. In general, there are two types of polyphase

decompositions. As the first type, any lowpass filter with cutoff

frequency at can be written as
M

M 1

H ( z ) = z i H i ( z M ) ,

1.4

i =0

where H i (z ) are the polyphase components. In the time domain, the

impulse responses of the polyphase components can be derived as
hi ( n) = h(i + Mn), 0 i M 1 . It must be noted that the polyphase
components can have different lengths. As an example, the 2-fold and 3fold polyphase components of a 6th order filter with transfer function
H (z ) can be derived as
H ( z ) = h[0] + z 1h[1] + z 2 h[2] + z 3h[3] + z 4 h[ 4] + z 5 h[5] + z 6 h[6]
2

1.5

= (h[0] + z h[2] + z h[4] + z h[6]) + z (h[1] + z h[3] + z h[5]),

14444442444444
3
1444424444
3
E0 ( z 2 )

2 fold

E1 ( z 2 )

= (h[0] + z 3 h[3] + z 6 h[6]) + z 1 (h[1] + z 3 h[4]) + z 2 (h[2] + z 3 h[5]), 3 fold .

1444424444
3
144244
3
1442443
E0 ( z 3 )

E1 ( z 3 )

E2 ( z 3 )

The second type of the polyphase decomposition can be derived as

M 1
H ( z ) = z ( M 1 i ) Ri ( z M ) and is useful in the analysis of synthesis bank filters
i =0

[3]. The relationship between these types is Ri ( z ) = EM 1i ( z ) [5]. The

advantage of polyphase components can be better understood by the use
of two noble identities shown in Figure 2 whose properties are proved in
[5]. It must be added that these noble identities are different in the case of
time varying systems and are defined in [6].
x[n]

H (z )

x[n]

H (zM )

v2 [n]

v2 [n]

y[m] x[n]

y[m] x[n]

v1 [m]
v1 [m]

H (z L )

y[m]

H ( z)

y[m]

Figure 2: Noble Identities in Multirate Systems.

Having these tools, we can derive the efficient decimation and

interpolation filter implementations as shown in Figure 3.
6

x[n]

H 0 (z M )

y[m]

x[n]
H L1 ( z )

z 1

z 1
H L2 ( z )

H1 ( z M )

.
.
.

.
.
.

z 1

H 0 ( z)

H M 1 ( z M )

Polyphase Decimator

y[m]

Polyphase Interpolator

Figure 3: Efficient Polyphase Decimator and Interpolator Implementation.

In these structures, the filters run at lower sampling rates compared

to the input signal sampling rate i.e. 1 T . Since the samples across the
adders are phased by T seconds and hence they do not interact in the
adder, some commutator models as described in [7] can be used to avoid
the adders for easier implementation. A generalization of the polyphase
decomposition is called the structural subband decomposition given by
[4]

H ( z ) = 1 z 1

V0 ( z M )

M
V1 ( z )
,

.
... z ( M 1) T

M
VM 1 ( z )

1.6

where T = [tij ] is an M M non-singular matrix. A non-singular square

matrix is one that has a matrix inverse. In other words, a square matrix is
nonsingular if and only if its determinant is nonzero. The relationship
between the polyphase components and the generalized polyphase
components is as

 X 0 (zM )
H0 (zM )

M
M
X1( z )
H1 ( z )
.

.
.
1

=T

.
.

.
.

M
M
X M 1 ( z )
H M 1 ( z )

1.7

As with the case for the polyphase decomposition, the structural

decomposition can be used to realize an FIR filter. Suppose H (z ) is an
FIR filter with an impulse response of length N = P M , where P and
M are positive integers. One can apply the structural subband
decomposition and write the filter as [4]

H ( z ) = 1 z 1

V0 ( z M )

M
V1 ( z )

M 1
.
M
... z ( M 1) T
= I k ( z )Vk ( z ),
.

k =0

M
VM 1 ( z )

M 1

I k ( z ) = tk +1, j +1 z j , k = 0,1,..., M 1.

1.8

j =0

Finally, the filter can be realized as shown in Figure 4.

x(n)

F0 ( z )

y (n)

F1 ( z )
.
.
.

FM 1 ( z )

Figure 4: Filer Realization Using Subband Decomposition.

where Fi ( z ) = I i ( z )Vi ( z M ), i = 0,1,..., M 1 . It must be mentioned that by

choosing simple invertible transform matrices T , the complexity can
further be reduced. Polyphase decomposition reduces the complexity of
the filter realization and hence finds extensive use in the analysis and
implementation of filter banks, as discussed in the next sections.
8

1.2.

Digital Filter Banks

The idea of filter banks is to split the input signal x(n) into subband
signals xk (n) through the use of analysis filters H k (z ) . The subband
signals can then be processed which is usually called subband processing.
^
The last stage is the reconstruction to approximate the output signal x k (n)
by the use of synthesis filters Fk (z) to combine the subband signals [3].
The typical system diagram is shown in Figure 5.
x ( n)

H 0 ( z)
H1 ( z )

H M 1 ( z )

.
.
.

x0 ( n)

y0 ( n )

x1 (n)

y1 (n)
.
.
.

xM 1 (n)

Analysis Bank

yM 1 (n)

F0 ( z )
F1 ( z )

FM 1 ( z )

x ( n)

Synthesis Bank

Figure 5: Typical Analysis and Synthesis Banks.

In this section, we will introduce the main blocks of the filter banks
and their properties for specific types of filter banks namely maximally
decimated, oversampled, and time varying filter banks which will be
discussed in the later subsections. Generally, a filter bank has five main
blocks namely analysis bank, downsampler, subband processing,
upsampler, and synthesis bank. These blocks will be discussed in the next
subsections.

1.2.1. Analysis Filter Bank

This block is a collection of M so called analysis or decimation
filters with a common input signal. The typical frequency responses of
these filters can be overlapping, marginally overlapping, and nonoverlapping as shown in Figure 6 .

Overlapping

Marginally Overlapping

Non-Overlapping

Figure 6: Typical Frequency Responses of Analysis Filters.

1.2.2. Downsamplers
In order to increase the subband processing efficiency, the sampling
rate can be reduced. The choice of down sampling ratio leads to two
types of systems as:
Maximally decimated filter banks: In this case, the number of the
subband channels is equal to the down sampling ratio leading to
equal number of samples in the subband and full band signals.
Although this seems to bring maximum efficiency, but it causes
aliasing.
Oversampled filter banks: Contrary to the maximally decimated
case, one can choose the decimation ratio to be less than the
number of subband channels. The draw back here is that the
number of subband samples is larger than the number of full band
samples. This has some advantages though and will be discussed
later.

1.2.3. Subband Processing

In this block, the subband signals are processed according to the
requirements. Examples of the processing can be coding, decoding, etc.
In the design of filter banks, this part is usually ignored and the prefect
reconstruction properties are defined for the filter bank only. Throughout
this document, we will assume the frequency response of the subband
processing block to be unity for all frequencies.

10

1.2.4. Upsamplers
In order to have the data at the original sampling rate, upsampling
which simply inserts a number of zeros in between every two samples is
used.

1.2.5. Synthesis Filter Bank

As discussed in Section 1.1, upsampling causes imaging and must
be removed by an interpolation filter. The synthesis bank is a collection
of M so called synthesis or interpolation filters with a summed output
which is simply a combination of the subband signals. In order to have
perfect reconstruction, the frequency responses of the synthesis filters
must be matched to frequency responses of the analysis filters. The
waveform h(t ) is said to be matched to the waveform s (t ) if [8]
h(t ) = ks( ) or H ( j 2f ) = kS ( j 2f )e j 2f = kS * ( j 2f )e j 2f ,

1.9

where k and s are arbitrary constants.

In other words, ignoring the delay and amplitude factors, the
transfer function of a matched filter is the complex conjugate of the
spectrum of the filter to which it is matched. The use of a matched filter
gives the maximum Signal to Noise Ratio (SNR). However, in most
cases, the synthesis and filters are exactly the same as the analysis filters.

1.3.

General Filter Bank Architecture

As a conclusion of the previous discussion, the filter bank

architecture can be drawn as shown in Figure 7.

11

H 0 ( z)

R0

H1 ( z )

R1

X (z )

Y0

Y0

Processing

Y1

Processing

.
.
.

H M 1 ( z )

R1

F1 ( z )

.
.
.

Y M 1

Processing

X ( z)

YM 1

RM 1

F0 ( z )

R0

Y1

FM 1 ( z )

RM 1

Figure 7: General Filter Bank Architecture.

In general, the decimation and interpolation ratios Rm can be

different resulting in the aliased channel outputs as [9]

1
Ym (e ) =
Rm
j

R m 1

X (e

j(

Rm

2k
)
Rm

k =0

)H m (e

j(

Rm

2k
)
Rm

).

1.10

The set {ym (n)} forms a critically sampled time-frequency

representation of the original signal. To construct the input signal and
assuming there is no processing, the signals {ym (n)} must be upsampled
and filtered through the synthesis filters Fm (z ) . The reconstructed signal
can be written as
^

M 1 ^

X (e j ) = Ym (e jRm ) Fm (e j )
m =0

M 1

1
=
m=0 Rm

Rm 1

X (e
k =0

j (

2k
Rm

) H m (e

j (

2k
)
Rm

1.11

)Fm (e )

The drawback of this system is that the information about the

aliased signals in one channel is available in the other channel signals.
However, it is possible to design exactly reconstructing analysis and
synthesis systems despite existence of aliasing in every individual
channel [9]. A special case can be derived letting Ri = M ,0 i M 1 and
is called a maximally decimated filter bank where the number of samples
in the set of {ym (n)} and x(n) is equal. This type of filter bank will be
discussed in the next section.

12

1.4.

Maximally Decimated Filter Banks

As stated before, a simplification by setting Ri = M ,0 i M 1 in

the general filter bank system of Figure 7 leads to maximally decimated
case. In this system, the number of samples for full band and subband
signals is equal. To analyze this system, the input-output relationship can
be written as [10]
Y ( z) =

1 M 1
1
{ H k ( z ) Fk ( z )} X ( z ) +
M k =0
M

M 1 M 1

{ H
l =1

k =0

( zW l ) Fk ( z )}X ( zW l ) .

1.12

The output signal has two parts as follows:

The first term represents the amplitude and phase distortion and
M 1
its distortion function is as { H k ( z ) Fk ( z )} . For PR, the distortion
k =0

function should be a pure delay.

The second term represents the aliased signal and its transfer
M 1
function is as H k ( z.W l ) Fk ( z ) which in the ideal case, must be
k =0

zero.
The system can be analyzed by the use of the polyphase
representation. To do this, the architecture is redrawn according to the
polyphase matrices as shown in Figure 8.
1

x(n)

z 1

z ( M 1)

M
M .
M

.
.

E(z M )

R( z M )

z ( M 1)

M .

z ( M 2)

.
.

y(n M + 1)

Figure 8: Realization of the Analysis and Synthesis Banks Based on Polyphase Matrices.

In this architecture, the matrices E ( z M ) and R ( z M ) represent the

polyphase components of the analysis and synthesis filters in the sense
that, the ith row of E ( z M ) and the ith column of R( z M ) have the
polyphase components of the H i ( z ), Fi ( z ) respectively. In the
mathematical form, this can be shown as [10]

13

N
N
H 0 ( z ) E0,0 ( z ) ... E0, N 1 ( z ) 1

: =
:
:
. :

N
N
( N 1)

H N 1 ( z ) E N 1,0 ( z ) ... E N 1, N 1 ( z ) z

z ( N 1)
F0 ( z )
: = :

1
FN 1 ( z )

R0, 0 ( z N ) ... R0, N 1 ( z N )

.
:
:

RN 1, 0 ( z N ) ... RN 1,N 1 ( z N )

1.13

Using the noble identities, this system can further be simplified to

ease the extraction of perfect reconstruction conditions as shown in
Figure 9.
1

z 1

M .

x(n)

z ( M 1)

.
.

E (z )

R(z )

.
.
.

z ( M 1)

z ( M 2)

y(n M + 1)

+
M

Figure 9: Simplified Realization of Filter Banks Using Noble Identities.

In this system, the only part affecting the PR is the product

E ( z ) R ( z ) since the rest can be proved to be a PR system. It can be
shown that the system is a PR system if this product is a pseudo circulant
matrix. A pseudo circulant matrix is a circulant matrix i.e., a matrix
whose rows are cyclically shifted versions of a sequence, but the
elements below the main diagonal are multiplied by z 1 . So, the matrix is
of the form [10]
p1 ( z )
p0 ( z )
z 1 p ( z )
p0 ( z )
M 1

1
1
z pM 2 ( z ) z pM 1 ( z )
1
z 1 p2 ( z )
z p1 ( z )

...
...
p0 ( z )
...

pM 1 ( z )
pM 2 ( z ) .

p0 ( z )

1.14

In this case, the first row is comprised of the polyphase components

of distortion function p0 ( z M ) + z 1 p1 ( z M ) + ... + z ( M 1) pM 1 ( z M ) which must be a
pure delay for a PR system. As a conclusion, the condition for PR can be
derived as [10]

14

z I N r
, 0 r N 1
0

0
R ( z ) E ( z ) = 1
z Ir

1.15

where I N is the identity matrix with r being a constant. The PR

condition can be stated in another way. If we have the set of power
complementary analysis filters, by a proper choice of the synthesis filters
[3], the subband signals can be combined in a way to produce the original
input signal at the output. In general, a set of filters H k ( z ) is said to be
complementary of order p if we have [11]
p

M 1

H k ( e j ) = 1 .

1.16

k =0

Here p is a positive integer. In special cases, the magnitude and

power complementary filters are the set which satisfy the general
equation for values of p = 1,2 as
M 1

H
k =0

(e j ) = 1,

M 1

H
k =0

(e j ) = 1.

1.17

It can be shown [11] that the higher order complementary filters

can generate ordinary magnitude and power complementary filters while
maintaining superior cut-off characteristics.The procedure to design a
maximally decimated filter bank has the following steps [10]:
An appropriate method should be chosen to design all the analysis
filters.
Having designed the analysis filters, polyphase matrix E ( z ) can
be determined.
The polyphase matrix of the synthesis filters R( z ) can be
determined by inverting E ( z ) .
In general, we prefer the Finite Impulse Response (FIR) solutions
which are guaranteed to be stable despite having larger delays compared
to their Infinite Impulse Response (IIR) counterparts. However, the
inverse of a FIR matrix generally leads to IIR solutions which necessitate
stability checks. For a special case of FIR matrices, called unimodular
matrices, FIR inverse solutions exist. A polynomial matrix is called
unimodular, if [12] its determinant is a nonzero constant. It must be
mentioned that for ParaUnitary (PU) matrices, there exist FIR inverses
15

also. Usage of paraunitary matrices, leads to paraunitary PR filter banks

which will be discussed in the next section.

1.5.

Paraunitary Filter Banks

As stated before, paraunitary filter banks constitute a special class

of the maximally decimated filter banks where the polyphase matrices
are paraunitary. The definition of paraunitariness needs the concept of
paraconjugation to be defined. This property can be defined for two types
of transfer matrices as follows [10].
1. In the case of a scalar transfer function H (z ) , the paraconjugate is
~

defined as H ( z ) = H ( z 1 ) . Thus, to obtain the paracojugate, one

has to replace z by z 1 and also replace each coefficient by its
complex conjugate. On the unit circle, paraconjugation is
equivalent to complex conjugation since we have
~

H ( z)
z =e

= H * ( z 1 )

z =e j

= {H ( z ) z =e j }* .

1.18

2. In the case of a matrix transfer function H ( z ) , paraconjugate is

~

defined as H ( z ) = H T * ( z 1 ) . To obtain the paracojugate, one has to

transpose the matrix, replace z by z 1 , and replace each
coefficient by its complex conjugate. On the unit circle,
paraconjugation is equivalent to transpose conjugation since we
have
~

H ( z)
z =e

= H *T ( z 1 )

z =e j

= {H ( z ) z =e j }T

1.19

Having these definitions, a matrix transfer function H (z ) , is

~

defined to be paraunitary if H ( z ) H ( z ) = I . In the case of a square matrix

function, and using the concept of the inverse matrix, we have
~

1
H ( z ) = {H ( z )} . So the paraconjugate can be derived from the inverse

matrix.

16

Another property of the paraunitary matrices is that if two matrices

P1 ( z ), P2 ( z ) are paraunitary, then the product P1 ( z ) P2 ( z ) will also be
paraunitary. This fact can be used to make a conclusion about the system
in Figure 9 . If the polyphase matrices satisfy the relationship
~

R ( z ) E ( z ) = I N , and assuming E (z ) to be paraunitary i.e. E ( z ) E ( z ) = I , the

~

PR system can be obtained choosing R( z ) = E ( z ) . In this case, If E (z ) is

FIR, then R(z ) will also be FIR and there is no concern about the
stability. In the next section, some properties of the paraunitary filter
banks will be introduced.

1.5.1. Properties of Paraunitary PR Filter banks

The choice of matrices being paraunitary brings some useful properties
as follows:
1. If the polyphase matrix E (z ) is paraunitary, then E ( z N ) is
paraunitary also. Hence, assuming
N
N
H 0 ( z ) E0, 0 ( z ) ... E0, N 1 ( z ) 1

,
: =
:
:
:

H N 1 ( z ) E N 1,0 ( z N ) ... E N 1, N 1 ( z N ) z ( N 1)

1.20

it can be shown that the vector transfer function H (z ) composed of all

the analysis filters is paraunitary.
2. If the vector transfer function H (z ) is paraunitary and assuming
that its components are power complementary as
N 1

H
k =0

(e j ) = const.

1.21

then, we have a lossless system with one input and N outputs.

3. If we have the set of the analysis filters H k (z ) of order L , the
synthesis filter coefficients can be computed by conjugating the
analysis filter coefficients and reversing their order as
f k [ n] = hk*[ L n], 0 k N 1 . Using the Fourier series properties, it
can be shown that the magnitude response of the synthesis filter
Fk (z ) is the same as the magnitude response of the analysis filter
17

H k (z ) while its phase is negative. In this sense, they satisfy the

definition of the matched filters [13].
4. As a straightforward result, if the analysis filters are power
complementary, then the synthesis filters are power
complementary also.
A uniform DFT filter bank is a system where a cascade of DFT and
IDFT matrices replaces the polyphase matrices and will be discussed in
the next section.

1.6.

DFT Modulated Filter Banks

Before moving to the discussion of DFT modulated filter banks, we

will define the concept of uniform and non-uniform filter banks.

1.6.1. Uniform and Non-uniform Filter Bank

Based on the characteristics of the data signals, one can choose to
shift the analysis and synthesis filters uniformly or non-uniformly along
the frequency axis. This leads to new classes of filter banks whose
sample filter characteristics are shown in Figure 10.

Uniform

Non-uniform

Figure 10: Filter Characteristics for Uniform and Non-Uniform Filter Banks.

In the uniform case, the channel filters are derived from a real linearphase LowPass (LP) prototype filter g (n) of length L by modulation as
[14]
hi (n) = ai g (n)e

( i + 0.5 )( n

( L 1)
)
2

+ ai* g (n)e

18

( i + 0.5 )( n

( L 1)
)
2

, i = 0,..., M 1,

1.22

where subscript * denotes the complex conjugation. In this system, since

the LP prototype has real coefficients, the channel filters are obtained by
cosine modulation and will be discussed in Section 1.7. The complex
multiplying factors ai define the modulation phase. The synthesis filters
are similar to the analysis filters but with a different modulation phase
usually chosen so that the resulting filters are the time-reverse of the
analysis filters. In this case, the overall filter bank response will have
linear phase. By appropriate design of the prototype filter, the overall
frequency response can be made flat also.
On the other hand, for the case of non-uniform filter banks, the
analysis filter for channel i is generated by modulation of a possibly
complex lowpass prototype g i (n) of length Li , as [14]
hi (n) = ai g i (n)e

Mi

( k i + 0.5 )( n

( Li 1)
)
2

+ ai* gi* (n)e

Mi

( k i + 0.5 )( n

( Li 1)
)
2

, i = 0,..., M 1. 1.23

Hence, the synthesis filters are given as

fi (n) = bi gi (n)e

Here, the term

( L 1)

( k i + 0.5 )( n i
)
Mi
2

Mi

( k i + 0 .5 )

+ bi* gi* (n)e

( L 1)

( k i + 0.5)( n i
)
Mi
2

, i = 0,..., M 1.

1.24

defines the ith channel center frequency, ki is

an integer, and M i is the decimation factor. The coefficients ai , bi are

complex and define the modulation phase. The choice of different
decimation factors M i gives the possibility of having narrow channels at
low frequencies and wider channels at high frequencies or vice versa.

1.6.2. Uniform DFT Modulated Filter Banks

DFT filter banks can realize linear-phase analysis and synthesis
filters using a proper complex modulation of a real-valued lowpass
prototype filter. In an N-channel uniform filter bank, the prototype filter
P(z ) is uniformly shifted on the unit circle. To analyze this system, we
use the z transform properties [15] and define the set of analysis filters
H 0 ( z ), H1 ( z ),..., H N 1 ( z ) in the time and z domain as [16]

19

hk [ n ] = p[ n ]e

j 2 kn
N

, H k ( z ) = P ( ze

j 2 k
N

).

1.25

Assuming a non-causal prototype filter and in order to obtain causal

analysis and synthesis filters, the impulse responses are delayed by N 1
2

samples. Therefore, the time-domain representation of the analysis filters

will be [16]
hk [ n] = p[ n

N 1
]e
2

j 2 k
N 1
(n
)
2
N

, n = 0,1,..., N 1, k = 0,..., M 1 .

1.26

The synthesis filters are identical to the analysis filters. It can be

shown [16] that if the prototype filter has the zero phase property, then
all the analysis and synthesis filters will be linear-phase. In the
implementation phase, the polyphase decomposition can be used to
reduce the implementation complexity. The polyphase components of the
prototype filter can be written as
N 1

P ( z ) = z l El ( z N ) .

1.27

l =0

So, the analysis filters can be written as

N 1

N 1

l =0

l =0

H k ( z ) = P( ze j 2k / N ) = z l e j 2kl / N El ( z N e j 2kN / N ) = z lW kl El ( z N )

1.28

IDFT
64444444
744444448
0
0
W
H 0 ( z)
W
W0
...
W 0 E0 ( z N )

H ( z)
0
W 1
W 2
... W ( N 1) z 1.E1 ( z N )
W
1

H 2 ( z ) X ( z ) = W 0
W 2
W 4
... W 2 ( N 1) z 2 .E2 ( z N ) X ( z ),

:
:
:
:

:
:
W 0 W ( N 1) W 2 ( N 1) ... W ( N 1)2 z N +1.E ( z N )
H N 1 ( z )
N 1

1.29

and can be arranged in a matrix formulation as

where W = e j 2 / N .
As a conclusion, the whole analysis bank can be implemented at the
cost of one filter plus an IDFT as shown in Figure 11. At the design
20

phase, we only need to design the prototype filter since the other filters
are shifted versions of the prototype filter.
x(n)
z

E0 ( z )

E1 ( z )

E N 1 ( z )

z 1

x0 (n)

y0 ( n )

x1 (n)
.
.
.

IDFT
x N 1 (n)

.
.
.

y1 (n)

y N 1 ( n)

Figure 11: Analysis Bank Polyphase Realization of DFT Modulated Filter Banks.

To analyze the system in another way, we can use the definition of

a transfer function. The relationship between signals xi (n) and yk (n) can
be written as [3]
yk ( n) =

1
M

N 1

x (n)W
i =0

ik

1.30

, it can be verified that

By defining the transfer function H k ( z ) = Yk ( z )
X ( z)
is a shifted version of a prototype response P(z ) through Equation
1.26 namely H k ( z ) = P( zW k ) [3].
To analyze the synthesis side, we assume the synthesis filters to be
F0 ( z), F1 ( z),..., FN 1 ( z) that satisfy the time reverse property as
H k (z )

Fk ( z ) = e j 2k / N P( ze j 2k / N ) .

1.31

Again, using the polyphase representation, we have

N 1

P( z ) = z l Rl ( z N )

1.32

l =0

In general, for all of the synthesis filters, we have

N 1

Fk ( z ) = z lW k ( N 1l ) Rl ( z N )

1.33

l =0

Thus, the reconstructed output signal y(n) in the matrix domain can be
written as
21

 Y0 ( z )
Y ( z)
1
.
Y ( z ) = [F0 ( z ) F1 ( z ) F2 ( z ) ... FN 1 ( z )] Y2 ( z )

:
YN 1 ( z )

1.34

which using Equation 1.34 can be rewritten as

DFT
6444444
4
74444444
8
0
0
W
W
W0
W 0 Y0 ( z )
...
0

W1
W2
... W ( N 1) Y1 ( z )
W
z N +1RN 1 ( z N ) ... z 2 R2 ( z N ) z 1R1 ( z N ) R0 ( z N ) W 0
W2
W4
... W 2 ( N 1) Y2 ( z ) .

:
:
: :
:
2
W 0 W ( N 1) W 2 ( N 1) ... W ( N 1) YN 1 ( z )

1.35

Finally, the architecture of the synthesis bank can be drawn as Figure 12.
y0 (n)
y1 (n)

DFT
y N 1 (n)

R N 1 ( z )

R N 2 ( z )

.
.
.

R0 ( z )

z 1
+

z 1

y (n)

Figure 12: Analysis Bank Polyphase Realization of DFT Modulated Filter Banks.

where Rl ( z ) = E N11l ( z ) and is the second type of the polyphase

representation [3]. In this scheme, the first filter should be centered at
= 0 but using a Generalized Discrete Fourier Transform (GDFT) [7],
this constraint can be removed. It must be mentioned that by appropriate
choices of GDFT matrix, one can obtain a filter bank with M filters
2

having real coefficients for even M .

In the GDFT case, the analysis filters hk (n) are derived from a
real-valued LP prototype FIR filter p(n) that has even length L as [17]
hk (n) = p (n)e

2
( k + k0 )( n + n0 )
M

k, n N .

1.36

22

Here, the offsets k0 and n0 are introduced leading to the name GDFT.
Choosing a linear-phase prototype filter and setting n0 in a way to have a
transform symmetric to L 1 2 , the modulated filters will have the linearphase property also. If we choose k0 = 0.5 , the frequency range (0,2 ) will
be covered by M 2 subbands for even M . In this case, the remaining
subbands are complex conjugate versions and can be ignored in the
processing reducing the complexity. So, we have a filter bank with M 2
filters. The synthesis filters can be obtained by time reversion of the
analysis filter as f k (n) = hk* ( L n + 1) . Thus, all filters can be derived from
one single prototype. The procedure to design these filter banks can be
summarized in the following steps [10]:
The prototype filter must be designed according to the system
requirements.
Having the prototype filter, the polyphase components Ek (z ) can
be achieved.
Assuming that Ek (z ) can be inverted, the synthesis filters can be
chosen as Rk ( z ) = EN11 k ( z ) .
It can be shown that the maximally decimated DFT filter banks at the
same time satisfy perfect reconstruction, have FIR analysis and synthesis
filters, and are paraunitary.
As a simple case, assume a prototype filter of the form
P( z ) = 1 + z 1 + z 2 + ... + z N +1 ,

1.37

which leads in polyphase components as delays reducing the filter bank

structure to Figure 13.

x(n)

z 1

z ( M 1)

.
.
.

DFT

IDFT

.
.
.

z ( M 1)

z ( M 2 )

y (n M + 1)

Figure 13: Simplest Case of the DFT Modulated Filter Banks.

Keeping in mind the fact that the cascade of the IDFT and DFT matrices
is equal to a unity matrix i.e. IDFT .DFT 1 = I N , one can assume the overall
23

system response to be still a delay but doing so, the filtering order must
be changed. The analysis to derive equations for the analysis and
synthesis filters is similar to the general case. It can be shown that the
analysis filters can be modeled as
6444444DFT
74444448
H 0 ( z ) W 0 W 0
W0
W 0 1
...

1
H ( z)
0
W1
W2
... W N 1 z
1
W
2
H 2 ( z ) = W 0 W 2
W4
... W 2 ( N 1) z .

:
:
: :
: :
H N 1 ( z ) W 0 W N 1 W 2 ( N 1) ... W ( N 1) 2 z N +1

1.38

In other words, we can assume the analysis filters as

H k (e j ) = P(e j ( 2k / N ) ) ,

1.39

which are obviously the uniformly shifted versions of the prototype filter.

1.7.

Cosine Modulated Filter Banks

In uniform DFT modulated filter banks, assuming P(z ) to be the

prototype lowpass filter with a cutoff frequency at / N , the analysis

j 2 k

filters can be derived as H k ( z ) = P( ze N ) . Cosine modulated filter banks

are defined by the use of Discrete Cosine Transform (DCT) which has
four types as [18]

24

2
N

ki

ci ck cos( N ),

i, k = 0,..., N

DCT

Type I :

DCT

Type II :

DCT

Type III :

2
N

(i + 0.5)k

) ,
ck cos(
N

DCT

Type IV :

2
N

(k + 0.5)

(i + 0.5)),
cos(
N

(k + 0.5)i

) ,
ci cos(
N

2
N

i, k = 0,..., N 1

1.40

i, k = 0,..., N 1
i, k = 0,..., N 1

if i 0 or N
1,

ci =
1

2 , if i = 0 or N .

It can be shown [18] that only types II , IV can be used for cosine
modulated filter banks and there is a relationship between the filter banks
defined using these two types of modulation. To be specific, suppose we
have a type IV cosine modulated filter bank with P ( z ), H k ( z ), and Fk (z )
being its prototype, analysis, and synthesis filters, respectively as
L

(2k + 1)(n ) + (1) k . }

2N
2
4 ,

L
k
f k (n) = 2. p(n) cos{
(2k + 1)(n ) (1) . }
2N
2
4

hk (n) = 2. p(n) cos{

1.41

where N and L are respectively the number of the channels and order of
the prototype filter. Having this, the type II cosine modulated filter bank
can be derived as [18]

L +1
) + (1) k . }
2N
2
4 .
^
^

L +1

(2k + 1)(n
) (1) k . }
h k (n) = 2 p(n) cos{
2N
2
4
^

h k (n) = 2 p(n) cos{

(2k + 1)(n

Hence, if the prototype real-coefficient lowpass filter is

at / 2 N , the analysis filters are
H k ( z ) = k P ( ze

j ( k + 0.5 )

) + P ( ze
*
k

j ( k + 0.5 )

25

).

1.42

P(z )

with a cutoff

1.43

In the time domain, we have

hk [n] = 2 p[n] cos{

L
(k + 0.5)(n ) + (1) k } .
N
2
4

1.44

This is obviously a cosine modulation instead of exponential modulation.

So, if the prototype filter is a lowpass filter, the analysis filters are
bandpass filters with the characteristics of the prototype filter as shown in
Figure 14 [10].
P
H0

2N

2N
H1

2
Figure 14: Analysis Filters for the Cosine Modulated Filter Banks.

Exploiting the advantages of the polyphase decomposition, and

assuming that P(z ) has 2 N -fold polyphase, we have
L

2 N 1

k =0

l =0

P ( z ) = z k p[ k ] =

E l ( z 2 N ),

El ( z ) =

p[2 Nk + l ] .

1.45

k =

If P(z ) has a length of 2.m.N , then the analysis filters realization can be
shown in a matrix form as in Figure 15.
H 0 ( z)

E0 ( z 2 N )

x(k )
z

z 1

E1 ( z 2 N )

TN

H1 ( z )
2 N

H N 1 ( z )

E2 N 1 ( z 2 N )

Figure 15: Polyphase Realization Analysis Bank for the Cosine Modulated Filter Banks.

Here, the matrix TN 2 N is [10]

26

TN 2 N = N C [( I J ) ( I + J )]N 2 N

1 0 ... 0
0 ...
0 1 ... 0
0 ...

I=
J =
,
: :
:
:

0 0 ... 1
1 ...
0
cos{ (0.5) m}

cos{ (1 + 0.5) m}
0
=

:
:

0
0

0 1
1 0
: :

0 0
...
...

It can be verified that

Cij = 2 N cos(

1.46

... cos{ ( N 0.5) m}

0
0
:

(i 0.5)( j 0.5))

are the elements of an

N N type four DCT matrix as discussed before. In this case, the

implementation cost of the analysis block is the prototype filter plus the
DCT modulation.
If the prototype filter has L + 1 real coefficients where L + 1 = 2mN
for some integer m and it is linear-phase, then it can be shown that the
FIR cosine modulated analysis bank is paraunitary if and only if the
polyphase components Ek ( z ) , Ek + N ( z ) of the prototype filter are power
complementary. This means that we have a lossless one input-two output
system. Having this, the FIR synthesis bank can be obtained by
paraconjugation to satisfy the PR condition.
In general, the length of the prototype can be arbitrary [18]. To
illustrate this, assume E ( z), R( z ) be the polyphase component matrices of
the analysis and synthesis filter banks and the linear-phase prototype
filter has a length of L = 2mN + m1 with a transfer function
H ( z) =

2 N 1

z
l =0

Gl ( z 2 N ) ,

1.47

with Gl ( z 2 N ) being the polyphase components. It can be shown [18] that

Gk (z ) satisfies

27


Gm1 1k ( z ),
G k ( z) = z m
1
z G2 N +m1 1k ( z ),

k m1 1

1.48

k > m1.

Thus, the polyphase matrices

E ( z ), R( z )

^ g ( z 2 )

E ( z ) = C 1
z g ( z 2 )
1

can be written as

1.49

R( z ) = z 1 g 0 ( z 2 ) g 1 ( z 2 ) C

~T

where
g 0 ( z ) = diag (G0 ( z ) G1 ( z ) ... GM 1 ( z ))
g1 ( z ) = diag (GM ( z ) GM +1 ( z ) ... G2 M 1 ( z ))

1.50

^
k
C = 2 cos (2k + 1) 2 N (l 2 ) + (1) 4 .
k ,l

To derive the PR conditions, we define P ( z ) = R ( z ) E ( z ) . It can be shown

[18] that for PR, we must have
0
P ( z ) = ( v +1)
I2
z

z v I1
,
0

1.51

where v is a positive integer and the dimensions of unity matrices I1, I 2

add to N . Further simplification can be made leading to the necessary
and sufficient conditions for PR as [18]
~

G k ( z )Gk ( z ) + G N + k ( z )GN + k ( z ) =

1
,
2N

0 k N 1 .

1.52

As a conclusion, the property for PR in cosine modulated filter bank with

arbitrary prototype filter length is similar to the case when the length is
an even multiple of N .The design procedure is as follows [10]:
28

1. First, the lossless systems Ek ( z ) , Ek + N ( z ) must be parameterized

according to the structures.
2. The next step is to optimize parameterization to achieve the
linear-phase prototype filter based on these polyphase
components satisfying the given specifications.
As a comparison between cosine modulated and DFT modulated
filter banks, it must be mentioned that, in a maximally decimated cosine
modulated filter bank, two polyphase components of the prototype filter
replace only one polyphase component in the DFT modulated case. In
such a system, in order to satisfy paraunitariness, or equivalently having
FIR system to be PR, each such pair of polyphase filters should form a
power complementary pair. In other words, they must be a lossless
system. On the other hand, for a DFT modulated system to be
paraunitary, each polyphase component must separately be lossless
which means that each polyphase component should be an allpass
transfer function. There is a problem here since allpass functions are
usually IIR and concerns about stability arise. To further simplify the PR
conditions, the oversampled filter banks are used which will be discussed
in the next section.

1.8.

Oversampled PR Filter Banks

In the previous sections, we have discussed maximal decimation,

which causes aliasing and makes the PR property hard to achieve. A
solution to this is the choice of oversampled FBs that easily suppress
aliasing and allow the combination of smaller subbands into wider
subbands without introducing large aliasing distortion. To analyze the
oversampled FBs, the concept of frame expansion needs to be defined.
Signal decomposition in l 2 (Z ) , is the expansion of the signal through a
sliding window using a selected set of elementary waveforms as [12]
K 1

x( n) =

i =0 j =

i , j (n) ,

1.53

i, j

where the vectors i , j (n) are the translated versions of K waveforms

29

i , j (n) = i (n jN ),

N K.

1.54

Any signal in l 2 (Z ) can be represented using such an expansion if and

only if the family

= i , j : i , j (n) = i (n jN ),

i = 0,1,..., K 1, j Z ,

1.55

forms a frame in l 2 (Z ) . A family of vectors given in Equation 1.56 is

said to be a frame if for any x l 2 (Z ) ,
K 1

A x x, i , j
2

Bx ,

A > 0, B < .

1.56

i =0 j =

The constants A, B are called the frame bounds and the frame is tight if
A = B . If the family is a frame, there is another frame as

= i , j : i , j (n) = i (n jN ),

i = 0,1,..., K 1, j Z .

1.57

In this case, the coefficients of the new signal decomposition can be

written as
K 1

x(n) = x, i , j i , j (n) .

1.58

i =0 j =

In addition, x, i , j stands for the inner product. Furthermore, frames

, can be interchanged in a way that any signal in l 2 ( Z ) can also be
written as
K 1

x ( n ) = x, i , j i , j ( n ) .

1.59

i =0 j =

Using the concepts of duality and tightness [19], the expansion formula
can be written similar to orthogonal expansions as
30

x ( n) =

1 K 1
x, i , j i , j ( n ) .
A i =0 j =

1.60

The starting point to relate this expansion to filter banks is the fact that
the inner product of a signal x with the vectors of the family can be
obtained as the outputs of an analysis bank H 0 ( z), H1 ( z ),..., H K 1 ( z) followed
by downsampling ratio of N K . As discussed before, the analysis filters
are complex conjugates of the time-reversed versions of a prototype
filter. In this case, the prototype filter becomes the elementary
waveforms i . So, the analysis filters are constructed as
hi (n) = i* (n), i = 0,1,..., K 1 .

1.61

It must be added that the case N = K leads to critically sampled case.

Extending this idea to filter banks, if is a frame, any signal expanded
by the analysis bank can be reconstructed from the subband components.
This reconstruction is performed by the use of a synthesis filter bank
G0 ( z ), G1 ( z ),..., GK 1 ( z ) whose impulse responses are derived as gi (n) = i (n) .
Hence, the synthesis bank can be expressed as
K 1

x(n) = yi (n k ) i , j (k )

1.62

i =0 j =

where yi (n) is the input of the ith channel of upsampler and synthesis
filter. As with almost all the filter banks, the polyphase idea can be used
to reduce the implementation complexity. The polyphase matrix of the
analysis and synthesis banks can be written as

31

 H 00 ( z )
H ( z)
10
H ( z) =

...

H ( K 1) 0 ( z )
G00 ( z )
G ( z)
10
G( z) =

...

G( K 1) 0 ( z )

H 0 ( N 1) ( z )
H 1( N 1) ( z )
,

...
...

... H ( K 1)( N 1) ( z )

...
...

G0 ( N 1) ( z )
G1( N 1) ( z )
,

...

... G( K 1)( N 1) ( z )

N 1

H i ( z ) = z j H ij ( z N )
j =0

1.63

...

...
...

N 1

Gi ( z ) = z j Gij ( z N )
j =0

Some important theorems about the properties of the oversampled filter

banks based on these matrices can be summarized as follows [12]:
1. A filter bank implements a frame expansion if and only if its
polyphase analysis matrix is full rank on the unit circle. If A is an
M N matrix, its rank is the largest number of columns of A
forming a linearly independent set. This set of columns is not
unique, but the number of elements of this set is unique. A matrix
is Full Rank if rank( A) = min(M , N ) .
2. A filter bank implements a tight frame expansion if and only if its
polyphase analysis matrix is paraunitary.
3. For a frame associated with an FIR filter bank with the polyphase
analysis matrix H ( z) , its dual frame consists of finite length
~

vectors if and only if H ( z ) H ( z ) is unimodular.

A special case is oversampled DFT modulated filter banks which
are FIR, PR, DFT modulated, and paraunitary as we shall see. To analyze
this system, we assume Figure 9 with the property R( z ) E ( z ) = I N . As a
special case, one can choose different dimensions for R( z ), E ( z ) leading to
the system in Figure 16 [10].
x(n)

z 1

z ( M 1)

.
.
.

E ( z ) N M

R( z ) M N

z ( M 1)

z ( M 2)

.
.
.

y(n M + 1)

Figure 16: Architecture of Oversampled DFT Modulated Filter Bank.

32

In this system, the condition R( z ) E ( z ) = I N still guarantees the system to

be PR. It can be shown that the PR condition becomes easier to achieve if
M > N . As an example, if M = 2 N , we have [10]
E ( z)
R ( z ) E ( z ) = I N = [R1 ( z ) R2 ( z )] 1 .
E2 ( z )

1.64

It is important to note that this equation does not necessarily imply

R1 ( z ) = E11 ( z ) , so inverses may be avoided and we can expect FIR PR
filter banks. In general, an analysis filter bank with M channels and the
decimation ratio of N can be realized based on an M -point IDFT
cascaded with an M N polyphase matrix containing the N -fold
polyphase components of the prototype filter P(z ) . It can easily be
verified that if M = N , then B in Figure 17 is a diagonal matrix.
x(n)

H 0 ( z)

H1 ( z )

z 1

B( z N ) M N

IDFT

z N +1

H M 1 ( z )

Figure 17: Polyphase Realization of the Oversampled DFT Modulated Filter Bank.

So, the polyphase components of the prototype filter can be derived as

[10]
El ( z ) =

with

k =

N '=

h0 [ N ' k + l ] ,

MN
gcd( M , N )

Divisor of numbers

1.65

and

gcd(M , N )

representing the Greatest Common

M,N .

In this case, the polyphase matrix will be

33


E0 ( z M )
0
...
0

M
0
...
.
E1 ( z )

0
.

.
.

E N 1 ( z M )
.

B( z N ) =

z N EN ( z M )
0

8
N
z
E
z
0
(
)
.
N +1

.
0
.

.
4
M

0
...
z E 2 N 1 ( z )

1.66

It is shown that if E ( z ) = IDFT B ( z ) is paraunitary, then with the choice

~

of R( z ) = B( z ) DFT , the PR property can be satisfied. This means that

the analysis bank will be paraunitary if B(z ) is paraunitary. The
paraunitariness of B(z ) can be proved if and only if
are power complementary. Hence,
Ek ( z ), Ek + N ( z ), k = 0,1,..., N 1
oversampled PR FBs can at the same time be DFT modulated, FIR, and
paraunitary.
All the filter banks discussed up to now are Linear Time Invariant
(LTI) systems. However, for some applications, we may require time
varying systems to improve the efficiency. In the next section, we will
discuss the time varying filter banks and their properties.

1.9.

Time Varying Filter Banks

In Time Varying Filter Banks (TVFB), the analysis/synthesis

filters, the number of bands, the decimation ratios, and the frequency
coverage of the bands change in time. This is in contrary to the FBs
discussed in the previous sections, where the system structure does not
change with time. The typical structure of a TVFB is shown in Figure 18.

34

H 0 (n, z )

x(n)

H1 (n, z )

R0 (n)

Processing

R1 (n)

Processing

R0 (n)
R1 (n)

RM ( n ) 1 (n)

G1 (n, z )

.
.
.

.
.
.
H M ( n )1 ( n, z )

G0 (n, z )

G M ( n )1 ( n, z )

RM ( n ) 1 (n)

Processing

x ( n)

Figure 18: General Architecture of Time Varying Filter Banks.

The advantage of such a system is that, we can modify the analysis

section according to the input signal properties and hence improve the
system performance. In this case, the PR property is the same as the
^

regular FBs i.e. x(n) = x(n ) where is an integer. The design

problem is to choose the system parameters in a way that the PR property
holds for all times. TVFBs can be analyzed using the time-domain
formulation [20] where the time varying impulse response of the entire
filter bank is derived in terms of the analysis and synthesis filter
coefficients. To do this, the filter bank is divided into three stages namely
the analysis filters, the down/up samplers, and the synthesis filters as
shown in Figure 19.
w0 (n)

v0 (n)
v1 (n)

x(n)

w1 (n)

P (n)

(n)

Analysis
Filters

Down/Up
Samplers

Q (n)
Synthesis
Filters

.
.
.

x ( n)

wM ( n )1 (n)

v M ( n )1 ( n)

Figure 19: Different Stages of a Time Varying Filter Bank.

The analysis filters output is v (n) = [v0 ( n), v1 ( n),..., vM ( n )1 ( n) ]T , where

vi (n) is the output of the ith analysis filter at time n . Furthermore, the
down/up samplers output at time n is w( n) = [w0 ( n), w1 ( n),..., wM ( n )1 ( n) ]T .
Assuming the length

N (n)

input signal at time

T
x N ( n) = [x ( n), x ( n 1), x ( n 2),..., x( n N ( n) + 1)] , we have

35

to be

v ( n) = P ( n) x N ( n) ,

1.67

where P(n) is an M (n) N (n) matrix whose mth row contains the
coefficients of the mth analysis filter at time n . Similarly, we have

w(n) = (n)v(n) ,

1.68

where (n) is a diagonal matrix of size M (n) M (n) with mth diagonal
element at time n being one if the input and output of the mth down/up
sampler are identical. The last stage is the contribution of the synthesis
filters, modeled by a matrix as
g 0 (n,1)
g 0 (n,0)
g 0 (n,0)
g (n,0)
(
,
1
)
g
n
g1 (n,1)
1
1
g 2 (n,0)
g 2 ( n , 2)
g 2 ( n , 2)

.
Q ( n) =

g
(
n
,
0
)
g
(
n
,
1
)
g
M ( n ) 1
M ( n ) 1 ( n,2)
M ( n ) 1

= q0 (n) q1 (n) q 2 (n) . . . q N ( n ) 1 (n)

.
.
.
.

g 0 (n, N (n) 1)
g1 (n, N (n) 1)
g 2 (n, N (n) 1)

.
,

g M ( n ) 1 (n, N (n) 1)

1.69

where qi ( n) = [g 0 ( n, i ), g1 ( n, i ), g 2 ( n, i ),..., g M ( n ).1 ( n, i )]T , and gi (n, j ) is the

jth coefficient of the ith synthesis filter at time n . Having all these, the
FB output at time n can be written as
^

x ( n) =

N ( n )1

q
i =0

T
i

(n) w(n i )

1.70

To derive a matrix equation for the output, we first define

T
s ( n) = [q0T ( n), q1T ( n), q2T ( n),..., q TN ( n )1 ( n)] . So, we have

36

 ( n) P ( n) x N ( n)

(n 1) P (n 1) xN (n 1)

(n 2) P (n 2) xN (n 2)
^

.
T
x ( n) = s ( n)
.

(n N (n) + 1) P(n N (n) + 1) x (n N (n) + 1)

N

1.71

Using the fact that the last N (n) 1 elements of vector xN (n i) are
identical to the first N (n) 1 elements of vector xN (n i 1) , this equation
can be decomposed as
x(n)
0...
0
[(n)P(n)]

n
P
n
x
n

0
(
1
)
(
1
)
0
...
0
(
1
)
[
]

x(n 2)
[(n 2)P(n 2)] 0...
0
0
0
^

x(n) = sT (n)
.
.
.

.
.
.
.

.
.
.
.

0...
0 [(n N(n) +1)P(n N(n) +1)]x(n 2N(n) +1)

1.72

where 0 is the zero column vector of length M (n) . In this case, the
input/output relationship of the TVFB becomes [20]
^

x(n) = zT (n) xI (n), xI (n) = [x(n), x(n 1), x(n 2),..., x(n I + 1)]

1.73

I = 2 N ( n) 1

The time varying impulse response vector of the FB is defined as

z (n) = A(n)s(n) . In this case, the [2 N (n) 1] [N (n) M (n)] matrix A(n) is
defined as

37

P ( n) T ( n )

0T

0T
A(n) =
.

T
0
...

0T

[P(n 1)

(n 1)

.
.
.
0T

.
.

0T

T
P (n N (n) + 1) (n N (n) + 1)
0T

1.74

In order for the system to act as a delay of integer , it is necessary

and sufficient [20] that all elements except the ( + 1)th in z (n) be zero at
all times. Having a desired impulse response b(n) , PR property holds
[20] if and only if A(n) s(n) = b(n) for all n .

1.10.
Differences between Time Varying and LTI
Filter Banks
Since the nature of LTI systems is different from that of the time
varying systems, there are some differences as outlined below [6]:
1. For an LTI system, if a FB is PR, then the FB with analysis and
synthesis filters interchanged is also PR. This does not apply in
the LTV case.
2. For an LTI PU system, the analysis and synthesis banks are
lossless whereas in the LTV case, the losslessness of analysis
bank does not imply losslessness of the synthesis bank. A system
with a transfer function H (z ) is lossless if it preserves signal
energy for all inputs. Mathematically, if the system input and
output are x(n), y(n) respectively, we have

x ( n) =

n =

y ( n)

1.75

n =

1. Replacing the delay z 1 in the implementation of an LTI PU

system with z L for some integer L does not change the PU
property. However, this is usually not true for a LTV lossless
system.

38

1.11.

Filter Bank Design Issues

Regardless of the different types of filter banks, the general system

of Figure 7 can be viewed as a hierarchical system [9] having specific
requirements and issues at each level. In the next subsections, we will
discuss these hierarchies and outline some of their important constraints
on the overall system design.

1.11.1.

Filter Issues

This is the lowest level where we face the filter design problems
and tradeoffs. In the analysis and synthesis side, the stop band, pass band,
and transition band characteristics are important. Furthermore, for the
synthesis part, we must consider the reconstruction issues as well. There
are some common issues such as the implementation complexity and
numerical sensitivity also.

1.11.2.

Filter Bank Issues

In this context, the quality of the frequency coverage of the analysis

filter bank must be considered. In order to save in the realization
complexity, it must be noted that one can implement the whole filter
bank requiring less operations than the sum of the operations for the
individual filters and must be checked thoroughly. As the last issue, the
capability of the system to reconstruct the data in the presence of
distortion, forces some constraints on the individual filter characteristics
in a top down approach.

1.11.3.

Analysis/Synthesis Issues

If we do not consider the processing, the system goal is to

reconstruct the input at the output. In this case, the analysis/synthesis
level distortions can be modeled as LTI distortions in the form of
magnitude and phase as well as distortions caused by aliasing. Here,
there is a problem whose goal is to minimize these distortions. This
minimization problem imposes further constraints on the design.
39

1.11.4.

Total System Issues

The desired goal in the system design is to maximize system

performance. To increase the performance, one can do improvements in
the analysis/synthesis filters to reject out of band and out of time
processing distortion.

40

Chapter Two: Overview of Transmultiplexers

and Satellite Payload Systems

41

42

2. Introduction
Digital filter banks find applications in subband coders for speech
signals, frequency domain speech scramblers, image coding, and
frequency multiplexing/demultiplexing. In this chapter, we will describe
the mathematical theory of the transmultiplexers as duals of filter banks
followed by issues such as channel equalization and interference
cancellation. As a special case of transmultiplexers, TDMA, CDMA, and
FDMA systems will be studied. Next, we will discuss and compare
different payload architectures for satellite applications. The chapter ends
with introduction to applications of filter banks in payload systems. This
topic will be studied in detail in the third chapter.

2.1.

Transmultiplexers

By definition, Transmultiplexer (TM) converts the time multiplexed

components of a signal into a frequency multiplexed version and back
[21]. A TM can also be used for applications such as channel
equalization, channel identification, etc. In [22], it was shown that a FB
and a TM are duals and the transposition of the analysis/synthesis banks
gives the dual TM. Using the duality, at the transmitter side, M different
source signals are multiplexed into one transmit signal by upsamplers
and synthesis filters. On the receiver side, the received signal is
decomposed into M source signals by analysis filters and downsamplers.
As it can be predicted, non-ideal synthesis/analysis filters result in
crosstalk between channels. Since analysis/synthesis filters are reversed,
analysis bank removes crosstalk introduced by synthesis bank. However,
the perfect reconstruction theory still applies as we shall see.

2.1.1. Mathematical Representation of Transmultiplexers

Suppose we have a series of symbol streams, either generated by
different users or parts of a signal generated by one user, and we want to
transmit these signals through a channel. As shown in Figure 20, we can
pass the signals through a series of transmitter (pulse shaping) filters
Fk (z ) to produce the signals [21]
43

xk (n) = sk (i ) f k (n iP) .

2.1

The term pulse shaping comes from the fact that the filters take each
sample of sk (n) and put a pulse f k (n) around it [21].
s0 ( n )

s1 (n)

sM 1 (n)

F0 ( z )

x0 (n)

F1 ( z )

x1 (n)

FM 1 ( z )

s0 ( n )

H1 ( z )

s1 (n)

H M 1 ( z )

s M 1 ( n)

C (z )

H 0 ( z)

+
Noise

xM 1 (n)

Figure 20: General Architecture of a Transmultiplexer.

Here we have M users transmitting through a channel described by

a linear time invariant filter C (z ) followed by additive noise. The
constraint of being time invariant may not be valid in the case of mobile
communications, but as we will see, equalization of the channel is
possible even in these cases. Finally, at the receiver side, the filters H k ( z )
separate the signals and only a downsampling by P is needed to get the
original symbol streams. In this system, M signals are multiplexed into
one channel which necessitates the constraint P > M giving the name of
redundant transmultiplexer [21] as opposed to minimal transmultiplexers
where P = M . Ignoring the effects of the channel, the input-output
relationship can be written as [23]
^

S i (z N ) =

1
M

M 1

M 1

S ( z ) F ( zW
N

k =0

l =0

)H i ( zW 1 ),0 i N 1, W = e

2
M

2.2

M 1

The transfer function Tki ( z N ) = Fk ( zW 1 )H i ( zW 1 ) relates the

l =0

output signal S i ( z N ) to the input signal S k ( z N ) . In general, due to the

existence of Multi User Interference (MUI), Inter-Symbol Interference
(ISI) caused by the channel linear distortion, and the additive noise, there
is always a difference between the transmitted and the received signals.
In order to decrease the Bit Error Rate (BER) of the system, channel
equalization is needed. For the case P > M , channel equalization can be
done by the use of transmit and receive filters. The idea is shown in
44

Figure 21, where M different channels Ck ( z ) and a common additive

noise replace the channel in the previous model.
Transmitter Filters

s0 (n)

F0 ( z )

C0 ( z)

s1 (n)

F1 ( z )

C1 ( z )

FM 1 ( z )

sM 1 (n)

H0 ( z)

H1 ( z )

H M 1 ( z )

s0 ( n )
^

+
Additive Noise

CM 1 ( z )
Channels

s1 (n)

s M 1 ( n)

Receiver Filters

Figure 21: Architecture of Transmultiplexer with Transmit and receive Filters.

If the channel transfer functions are the same, the system is

equivalent to the system in Figure 20. Interestingly, in the case of
multiuser communications over wireless channels, the new representation
becomes useful. In the next section, we will discuss the perfect
reconstruction property of the system from a mathematical point of view.

2.1.2. Perfect Reconstruction in Transmultiplexers

Assuming the system in Figure 21 and using a mathematical result,
one can derive the perfect reconstruction constraint. To further simplify
the analysis, we deploy the fact that, if an LTI filter g (n) is placed
between an upsampler and a downsampler of ratio P , the overall system
is equivalent to the decimated version of the filter impulse response
which becomes g (nP) [21]. In this case, designing the transmit/receive
filters in each branch, so that the decimated version of H k ( z )Cm ( z ) Fm ( z )
becomes a pure delay, the MUI can be cancelled and the system is a PR
system. From the duality property mentioned in Section 2.1, and
according to the PR condition discussed in Section 1.4, if H P ( z ), GP ( z ) are
the polyphase matrices of the analysis and synthesis filters, then the FB is
PR if and only if GT P ( z ) H P ( z ) = I . On the other hand, a TM is PR if and
only if H P ( z)GT P ( z ) = I [24]. If the decimation ratio and the number of
channels are the same, the PR conditions are identical. It must be
mentioned that, the PR properties are independent of filter lengths,
causality of filters etc., and can be satisfied for both minimal and
redundant transmultiplexers. However, for the minimal case, there may
not always exist FIR or stable IIR solutions. So, allowing some
redundancy will make the solutions feasible.
45

2.1.3. Canceling InterBlock Interference in

Transmultiplexers
Using the polyphase realization of the transmit/receive filters, we
can derive the channel between the mth transmitter and kth receiver in
matrix form. The new system diagram is shown in Figure 22.
sm (n)

R0,m ( z )

cm (z )

z 1

R1,m ( z )

R P1,m ( z )

Channel between the

E k ,1 ( z )

mth transmitter and kth

+z
+

sk (n)
1

z 1

z 1

Channel

z
z

Ek ,0 ( z )

E k , p 1 ( z )

z 1

receiver

Figure 22: Modeling the Channel to Cancel InterBlock Interference.

In this figure, Ri ,m ( z ), Ek ,i ( z ) are the polyphase components of the

transmitter and receiver filters, respectively. It is shown in [21] that the
InterBlock Interference defined as the interference between input vectors
L

occurring at different times at the input of Cm ( z ) = cm (n) z n can be

n =0

cancelled through two methods as follows:

Zero Padding: In this scheme, a block of L zeros is inserted at the
end of each block of P L symbols. In other words, we have
RP L ,m ( z ) = ... = RP 1,m ( z ) = 0 .

Zero Jamming: In this scheme, a block of L samples at the

beginning of each block of P successive received symbols are set
to zero. Mathematically, we have Ek , 0 ( z ) = ... = Ek , L 1 ( z ) = 0 .

2.1.4. Canceling Multi User Interference in

Transmultiplexers

In reality, the signal s k (n) is affected by both sk (n) and sm (n), m k ,

the latter being called MUI. To cancel this, the mth channel matrix is
derived to be [21]
46

0
c m (0)
c (1) c (0)
m
m
.
.

.
.

.
.

Am = c m ( L)
0
c m ( L)

.
.

.
.

.
.

.
.

.
.
.

.
.
.

.
.

0
0
.

.
.
.

.
.

c m ( L)

2.3

This matrix is a banded Toeplitz matrix. An N N matrix

TN = [t k , j ; k , j = 0,1,...N 1] is Toeplitz if t k , j = t k j . In other words, the matrix
has constant values along negative slope diagonals. The matrix is banded
if there is a finite m for which tk = 0, k > m . A banded Toeplitz matrix has
a finite number of diagonals with nonzero entries and zeros everywhere
else [25]. Hence, for any nonzero number k we have

[1

k1 . . . k ( P 1) ]Am = Cm ( k )[1 k1 . . . k ( P L1) ].

2.4

In this case, if we choose transmit/receive filters as

H k ( z ) = ak (1 + k1 z + k2 z 2 + ... + k ( P1) z ( P1) )
Fm ( z ) = r0,m + r1,m z 1 + r2,m z 2 + ... + rP L1,m z ( P L1)
The

2.5

transfer

function
from
to
becomes
s k (n)
sm (n)
Tkm ( z ) = ak Cm ( k ) Fm ( k ) which in the ideal case, must equal (k m) . If

the multipliers ak are chosen to be

1 , the constraint to satisfy the

Ck ( k )

MUI cancellation becomes

Fm ( k ) = (k m),0 k , m M 1 .

2.6

Some conclusions can be made as follows:

The values k can be arbitrary numbers but they must be distinct.

47

The receive filter Fm (z) must at least have an order of M leading

to P M + L .
The MUI can be cancelled even if the channels are unknown
since only their order is important.
The idea of transmultiplexers can be used in several well known multiple
access schemes such as TDMA, FDMA, and CDMA. To describe these
architectures, we will define the time-frequency interpretation and link
the definition to the transmultiplexer theory. Doing so, we can
understand the relationship between the transmultiplexer theory and these
schemes.

2.1.5. Time Frequency interpretation

The uncertainty principle states that no function can at the same
time be centered in both the time and frequency domains. To get around,
two types of spread for a discrete time function are defined as [26]:
Time Spread:
The time spread of a function {h0 (n)} is defined as
n2 =

_
1
1
2
2
( n n) h 0 ( n) , E = h 0 ( n) =

E n
2
n

H (e
0

jw

) dw, n =

1
2
n h0 (n) , 2.7
E n

where E , n are the energy and time centers respectively.

Frequency Spread
The frequency domain spread of a function {h0 (n)} is defined as
w2 =

_
_
2
1
1
jw 2
2
(
w

w
)
H
(
e
)
dw
,
w
=
w H 0(e jw ) dw ,
0

2E
2E

2.8

with w being the frequency center. The general frequency-time tilde of

the signal is shown in Figure 23. The shape and the location of the tilde
can be modified according to the time and frequency centers.

48

2 n

2 w

n
Figure 23: Time Frequency Tilde of a General Discrete Time Function.

In the next sections, we will discuss CDMA, TDMA, and FDMA

schemes as results of expanding the time frequency interpretation to
design orthogonal structures.

2.1.6. CDMA System Based on Transmultiplexers

In Code Division Multiple Access (CDMA) systems, each user is
assigned a pseudo-random code sequence c1i , c2i , c3i ,..., c Ni chosen from a
set of orthogonal codes called spreading sequences. A set of codes are
said to be orthogonal, if for any cmj , cmi , i j we have [13]
N

c
m=1

c = i, j .

i j
m m

2.9

These codes are simultaneously spread in time and frequency which

means they are both allpass like and spread in time domain [26]. These
codes act as keys in transmission since they spread the undesired signal
spectrum improving the system immunity towards noise and jamming.
On the other hand, these codes if used properly despread the signal coded
by the same spreading sequence. In the transmission, each user symbol
u i [k ] is replaced with the sequence c1i u i [ k ], c 2i u i [ k ], c3i u i [ k ],..., c Ni u i [ k ] . This
is equal to time domain multiplication of the user data with the spreading
sequence. By setting the filter coefficients of a TM equal to orthogonal
user codes, a CDMA transmitter and receiver can be achieved. In this
sense, the transmitter code multiplication may be viewed as filtering
operation, with FIR transmit filter as C i ( z ) = c1i + z 1c2i + ... + z N +1c Ni .On
the receiver side, the receiver code multiplication and summation can
also be treated as a filtering operation, whose receive filter is
49

1
C i ( ) = z N 1 (c Ni + ... + z N + 2c2i + z N +1c1i ) .The architecture of the system with
z

a simple example is depicted in Figure 24 [10].

u1 [k ], u1 [k + 1]
u 2 [k ], u 2 [k + 1]

u N [k ], u N [k + 1]

N . C 2 ( z)

C1 ( z)

.
.

C 2 ( z 1 )

Channel

.
.
.

u2 [k + 1], u2 [k ]

C N (z )

Figure 24: CDMA System Based on Transmultiplexer.

2.1.7. TDMA System Based on Transmultiplexers

In TDMA, each user occupies the whole channel frequency and
transmits in a dedicated time slot. In the extreme case, the allocation of
the time slot can be done at sample level with a transfer function as
Fk (e j ) = e jk . In other words, we have spectrally spread synthesis filters
Fk (e j ) = 1, 0 , i = 0,1,..., M 1 [26]. So, we replace the synthesis

and analysis filters by delay operators as shown in Figure 25 [10].

u1 [k ], u1 [k + 1]
u 2 [k ], u 2 [k + 1]

u N [k ], u N [k + 1]

N .

z 1

.
.

zN
z N +1

Channel

z 1

z N +1

.
.
.

Figure 25: Simple TDMA System Based on Transmultiplexer.

In

this

case,

the

channel

will have series of the signals

which
can be retrieved after passing the
u1 [k ], u2 [k ],..., u N [k ], u1 [k + 1], u2 [k + 1],...
synthesis filters. In general, the time slot can be done at a frame level
consisting of several samples.

50

2.1.8. FDMA System Based on Transmultiplexers

In FDMA systems, each user is assigned a portion of the available
channel frequency. To do this, we can choose the synthesis and analysis
filters as frequency selective filters that add up to cover the full channel
frequency band. The filter characteristics for the ideal case are shown in
Figure 26 [26].
Fi (e j )

Figure 26: Transmultiplexer Synthesis/Analysis Filter Characteristics for FDMA System.

2.2.

Satellite Payload Architectures

In order to provide transponded satellite connectivity among

terminals with a wide range of data rates, wideband payloads will play an
important role in the next generation communications satellites. The
fundamental topics in this area are channelization and satellite routing. In
order to combat the effects of low radiated power and receiver sensitivity
at the mobiles, there is a need to increase the size and complexity of the
satellite antennas. The satellites must also generate a large number of
spot beams in order to cover the full field-of-view from the satellite [27].
It is necessary to have onboard switching or equivalently FDM
multiplexing and demultiplexing to direct the received carriers to the
desired spotbeams. In literature, four types of payload architectures have
been proposed namely Bentpipe, Full Processing, Partial Processing, and
Hybrid payloads [28]. The following sections will study and compare
these architectures.

51

2.2.1. Bentpipe Payload

This type of payload is the simplest architecture and converts the
uplink carrier frequency to another carrier frequency for downlink signal
without any processing. In this system, there is no information about the
bit level data. Hence, the routing is not intelligent. In order to increase
the efficiency and cover all possible beam-to-beam connections, a
transponder-hopping technique [28] must be used. However, the
drawback of transponder hopping solution is that, the number of
transponders is the square of the number of beams, which makes it
impractical for a large number of beams. Another solution is the onboard
beam switching technique based on a so-called Satellite-Switched Time
Division Multiple Access (SS/TDMA) [28]. However, the bentpipe
system has some drawbacks irrespective of the routing algorithms used.
In this system, the traffic from one beam to another beam may not always
be at the full capacity of the transponder. This means that a portion of the
capacity is always wasted, reducing the payload efficiency.
The main drawback of the bentpipe payload is the transfer of the
uplink noise to downlink, since there is no error correction algorithm in
the transponder. It is shown in [28] that the overall Signal to Noise Ratio
(SNR) of the system is
Eb

N 0 overall

E
E

= b
+ b

N 0 uplink
N 0 downlink

2.10

This obviously shows degradation in the end-to-end performance.

However, adding some processing in the payload can improve the
performance as will be discussed in the next sections.

2.2.2. Full Processing Payload

As opposed to the bentpipe technique, in a full processing payload,
the uplink signal is demodulated and decoded, so the routing can be done
at the packet level according to the destination information provided in
the transmitted user data packets. To transmit the routed data in the
downlink, there is a need for encoding and modulation, which can be
done considering the propagation characteristics of the downlink. This
method is usually called Satellite Based Asynchronous Transfer Mode
52

(Satellite-ATM). The advantage of this type of payload is that there is an

intelligent routing at the packet level, which increases the efficiency. The
routing subsystem for a full processing payload can be implemented
using a combination of analog and digital components reducing the
weight and power consumption. An important feature of the full
processing system is the decoupling of the uplink and downlink noise
since channel coding is applied twice. This results in an improvement on
the overall end-to-end BER performance. On the other hand, a full
processing system is complicated and there is a trade-off between
reduction of weight and power consumption, complexity, and ability of
using digital components.

2.2.3. Partial Processing Payload

As discussed before, the main features of bentpipe payload are
simplicity and degradation of overall SNR. On the other hand, a full
processing payload is complicated but it has better performance from the
SNR point of view. According to the applications, a compromise on
memory size, onboard decoder, speed, and power consumption can be
made using a partial processing system. A partial processing system
includes demodulator and modulator, but not decoder and encoder.
Therefore, channel corruption in uplink and downlink are decoupled but
there is no coding gain. Furthermore, a hard decision has to be applied in
the uplink signal demodulation, which will further destroy the soft
information of the received uplink signal since a detection error made
during the processing of the uplink signal cannot be corrected. From the
end-to-end BER performance, a partial processing system is generally
superior to a bentpipe system, but inferior to a full processing system.
However, the routing is not still in the bit level. To solve this, we can add
an uncoded header that contains the routing information. However, to
make the system less vulnerable to the uplink noise, we need a long
header to compensate for the lack of coding gain. Another approach can
be to include a simpler full processing subsystem to process headers thus
allowing for coded headers.

53

2.2.4. Hybrid Payload

In a hybrid [28] structure, a combination of bentpipe and full
processing architecture or bentpipe and partial processing is used. This
system has some of the disadvantages and advantages of the individual
architectures as discussed before. The next sections will discuss filter
bank solutions for frequency multiplexing/demultiplexing.

2.3.
Frequency Multiplexing/Demultiplexing
Using Filter Banks
Filter banks provide solutions to frequency multiplexing and
demultiplexing problems in the satellite communications. The main
solutions reported in the literature are as [29]:
Channel-individual digital filtering with single- or multi-step
decimation: In this algorithm, center frequency and bandwidth of
each channel is independent of adjacent channels, which brings
the highest flexibility. The drawback is that, this approach has the
maximum computational load.
Tree-structured filter bank: This architecture has cascaded
directional filter cells, where the complex FDM input is first
downsampled by two followed by a separation into two complex
subsignals with half bandwidth. Therefore, each directional filter
cell is a four channel oversampled complex modulated uniform
filter bank. However, only two channels are used for subsequent
processing. Before the whole filter bank, a Hilbert transform
converts the real FDM signal to its associated analytic
representation.
Complex modulated uniform DFT filter bank: As discussed in
Section 1.6, a single-step decimation and a polyphase filtering is
used. This approach has the highest efficiency in arithmetic
operations and storage. The previous versions have no flexibility
to channel allocation and bandwidth, but it will be shown in the
next chapter that a new solution utilizing a channel switch can
bring full flexibility to the system making it a suitable choice for
frequency multiplexing and demultiplexing.

54

Chapter Three: Proposed Bentpipe System

and Simulation Results

55

56

3. Introduction
In order to provide a solution to the increasing demands on
multimedia services supporting high bit rates and mobility, the European
Space Agency (ESA) has proposed three major network structures for
broadband satellite-based systems as [30]:
Distributed bentpipe satellite internet access network: In this
scenario, user terminals combine one or few user traffics
producing unbalanced forward/reverse link traffic. The bentpipe
architecture has simplicity and easy system evolution support.
Meshed type of regenerative satellite network for professional
users: This type of network will support different classes of
professional users. A set of earth stations will support different
classes and the system will be able to build Virtual Private
Networks (VPN).
Meshed type of regenerative satellite network for backbone
connectivity: The main advantage of this network over terrestrial
networks is the capability to interconnect several Internet Service
Provider (ISP) access points. This calls for a regenerative onboard
processor to flexibly direct spot beams creating an add-on to
terrestrial networks.
In all, the aim is to have a globally interconnected digital society, with
multimedia applications, information on demand, and low cost delivery
of advanced data services which is the users expectation and the
operators promise [31]. In these systems, satellites communicate with
users through multiple spot beams, which necessitate efficient use of the
limited available frequency spectrum. This calls for satellite onboard
signal processing to support frequency band reusage among the beams
and bring flexibility in bandwidth and transmission power allocated to
each user. To support services at different data rates and bandwidths, a
dynamic frequency reusage system is required. Consequently, there is a
need for digital Flexible Frequency Band Reallocation (FFBR) networks
(also referred to as frequency multiplexing and demultiplexing networks
[29]). These networks should bring Perfect Frequency Band Reallocation
(PFBR), flexibility, low complexity, parallelism, and implementation
simplicity.
In this chapter, a new class of FFBR networks based on variable
oversampled complex modulated filter banks (FBs) is studied. This
system uses some of the properties of the alternatives discussed in
57

Section 2.3 and can outperform the existing structures from flexibility,
low complexity, parallelism, PFBR property, and simplicity points of
view. The proposed system can be deployed in any communications
environment that requires transparent (bentpipe) reallocation of
information. In the next sections, we will start with the formulation of the
problem followed by the proposed SISO network. Next, we will study the
system from the implementation point of view with an extension to the
MIMO case where we will discuss different scenarios of the MIMO case.
We will illustrate the simulation results of the system to evaluate the
architecture for different input scenarios.

3.1.

Problem Formulation

We assume the input signal is divided into Q fixed granularity

bands as shown in Figure 27. Any user can occupy one or several (at
most Q ) of these granularity bands. Consequently, the input signal
contains a variable number of user subbands q where 1 q Q [32]. In
the extreme cases, q = Q and q = 1 , which means the user can occupy up
to whole the available frequency band and the system can support highest
possible data rates on demand.

Granularity Bands
2
Q

2
Q

Granularity Band

2 2
+
Q
Q

Guard Band

2 2
+
Q
Q

Figure 27: Illustration of Guard and Granularity Bands in the FFBR System.

The value of q can change during operation corresponding to a

specific reallocation scheme at any time. Furthermore, frequency guard
bands (or equivalently filter transition bands) are only present between
different user subbands, and ensure the realizability. In brief, the SISO
FFBR network has three major tasks:
Separate the input signal into the desired user subbands: This is
similar to an analysis bank as discussed in Section 1.2.1.
Shift the user subbands in frequency to the desired positions: The
use of a switch can accomplish this task as will be discussed later.
58

Combine the frequency-shifted user subbands into the output

signal: As discussed in Section 1.2.5, a synthesis bank can
perform this task.
As a conclusion, to complete the FFBR system, a filter bank needs to be
chosen. In the next section, we will discuss a class of filter banks used for
the proposed FFBR system.

3.2.
Class of Online Variable Oversampled
Complex Modulated Filter Banks
This section introduces the proposed class of variable oversampled
complex modulated FBs used in the proposed FFBR network. We will
start with constraints of the system followed by the structure of the
proposed filter bank. Finally, we will discuss the implementation issues
of the system.

3.2.1. System Constraints

As discussed in Section 3.1, the input signal consists of variable q
neighbouring users with Q being the fixed number of granularity bands.
Furthermore, the input/output sampling rates are the same and the
input/output subbands have unique positions. Consequently, the problem
becomes reallocating the subbands in the input spectrum to the desired
positions in the output spectrum and can be solved by using the filter
bank shown in Figure 28 [32].
Fixed Analysis FB

Adjustable Synthesis FB

H 0 ( z)

G0 ( z )

H1 ( z )

G1 ( z )

.
.
.

x ( n)

H N 1 ( z )

y0 (n)
y1 (n)

.
.
.

G N 1 ( z )

yq 1 (n)
Channel Combiner

Figure 28: FFBR system with Fixed Analysis and Adjustable Synthesis Bank.

59

y ( n)

In this system, the analysis filter bank splits the input signal into
subbands. Furthermore, the combination of downsamplers, upsamplers,
and synthesis filter bank with adjustable synthesis filters generates the
required frequency shifts and recombination of FB subbands into the q
shifted user subbands yi (n), i = 0,1,..., q 1 . To satisfy the system
requirements, specific constraints on N , M must be posed, which will be
discussed in the next section.

3.2.2. Constraints on Sampling Rate Converters and

Number of Channels
As discussed in Section 3.2.1, the choices of M and N play an
important role for the system to satisfy its properties. For instance, if
M = N = Q , the system becomes a maximally decimated FB and hence the
variable subband widths and zero aliasing cannot be achieved
simultaneously. As discussed in Section 1.8, letting a slight oversampling
by choosing M < N , makes the PR conditions milder. To generate all
integer frequency shifts of the granularity frequency, decimation and
interpolation by M can be used. Thus,
M = BQ, B 1,

B int . .

3.1

Since M < N , the number of uniform-band channels cannot equal the

number of granularity bands [33]. Instead, N must be a multiple of Q as
N = AQ =

AM
, A > B,
B

A int . .

3.2

It is shown in [33] that for a fixed N , the complexity is minimized by

selecting M as large as possible without introducing aliasing. Hence,
from Equations 3.1 and 3.2, B is selected as [33]
B = A K , 1 K A 1,

int . ,

3.3

60

where
M = N KQ .

3.4

In addition, K is the smallest integer allowed without introducing

aliasing. In practice, it is possible to make aliasing components arbitrarily
small by the stopband attenuation of the analysis filters [33]. Precisely, if
the filter bandwidth and transition band are 2 and 2 respectively, the
N

filters attenuate the aliasing and the minimum value for

K

N
N .
(
)
Q N +

is [33]
3.5

The next sections will study the building blocks of the system including
analysis filters, synthesis filters, channel switch, and the channel
combiner.

3.2.3. Analysis Filters

As discussed in the filter bank theory, the analysis and synthesis
filters are obtained from the prototype filter
D

P ( z ) = p ( n) z n

3.6

n =0

where D is the filter order with a constraint that it must be linear-phase

and symmetric so that, p (n) = p( D n) . Hence, its frequency response
can be written in the form of real zero-phase frequency response as [15]
P(e jT ) = e

jDT

PR (T ) .

3.7

To obtain the set of analysis filters, the complex modulation can be used
as
H k ( z ) = k P ( zWNk + ), k = 0,1,..., N 1 ,

61

3.8

where
WN = e

j 2

( k + ) D

, k = WN

3.9

In addition, is a real valued constant to shift the filters to the desired

center frequencies and the constants k compensate the phase rotations
caused by replacing the D th-order linear-phase filter H (z ) with
H ( zWNk + ) . In this way, all analysis filters become linear-phase FIR filters
with the same delay as the prototype filter.

3.2.4. Synthesis Filters

To obtain the synthesis filters, in this specific case, we are
interested both in PR and ability of shifting the signals to the desired
location in the frequency spectrum. It is shown in [33] that the choice of
the synthesis filters as
m r ND
2M
,
N

Gk ( z ) = kr H c kr ( z ), ckr = k + sr A, kr = W

3.10

with

sr 0
Bsr ,

,
mr =

M + Bsr , sr < 0

3.11

satisfies these constraints. Here, sr is the number of granularity band

shifts for each signal. In general, the synthesis filters are obtained by
replacing the D th-order linear-phase filters H k (z ) with H k ( zWMm ) . The
r

constants kr compensate the resulting phase rotations. For simplicity,

the constants can be made kr = 1 if [33]

62

mr D
= int . .
2M

3.12

As a result of this discussion, we are able to transform the adjustable

synthesis filter bank to the combination of a set of fixed filters, an
adjustable switch, and a series of multipliers. In the next section, the
adjustable switch operation that shifts the signals will be discussed.

3.2.5. Application of Switch in the FFBR Network

To decrease the complexity of the system and considering the fact
that variable filter banks are expensive to implement, with an appropriate
choice of filters and parameters in the FFBR network [33], it is possible
to implement the same function using variable channel switch and fixed
FBs according to the scheme in Figure 29.
Fixed Synthesis FB

Fixed Analysis FB

H 0 ( z)

H1 ( z )

G0 ( z )
G1 ( z )

kr

y1 (n)

.
.
.

x ( n)

H N 1 ( z )

y0 ( n )

M
Channel Switch

G N 1 ( z )

y(n)

yq1 (n)

Channel Combiner

Figure 29: FFBR system with Fixed Analysis/Synthesis Banks and Channel Switch.

In this system, the outputs from the analysis banks are connected to the
inputs of the synthesis bank. In this way, the complexity can be reduced
since fixed filters are less complex to implement. Furthermore, the fixed
analysis/synthesis FB, can be implemented using only one filter and an
IDFT/DFT block [3]. In the next section, the efficient implementation of
the system making use of the polyphase decomposition will be discussed.

63

3.2.6. Efficient Implementation

The polyphase decomposition reduces the implementation
complexity of filters and filter banks. The starting point in the analysis is
the construction of the polyphase components of the prototype filter as
N 1

P( z ) = z i Pi ( z N ) ,

3.13

i =0

where Pi (z ) are the polyphase components. Using this, the analysis filters
H k (z ) can be written as [32]
N 1

H k ( z ) = k z i i Pi ( z N W NN ) W N ki ,

i = W Ni .

i =0

3.14

As a result of the discussions in Section 1.6, it can easily be verified that

the system can be implemented using an N -point IDFT and an N -point
DFT as shown in Figure 30.
Analysis Bank
x ( n)
z

P0 ( z LW NN )

P1 ( z W N )
L

kr

Synthesis Bank
0

N 1

N 2

IDFT

.
.
.

z 1

N 1

PN 1 ( z W N )
L

PN 1 ( z LW NN )
N

PN 2 ( z W N )
L

M
M

DFT

N 1

N 1 0

z 1
+

.
.
.

z 1
N

P0 ( z W N )
L

y ( n)

Channel Switch

Figure 30: Polyphase Implementation of the FFBR Network.

Here k = kWNk compensate for the phase rotations and L = A is chosen

B
to be an integer. For the cases when L is not integer, a more general
polyphase implementation of the polyphase components followed by
downsampling has to be used [34]. The system in Figure 30 is a SISO
network. In general, we may have several inputs and outputs leading to a
MIMO network. In the next sections, extension of the SISO case to
MIMO will be studied.

64

3.3.

MIMO FFBR Network

In general, it is desired to have a system with several inputs/outputs

since the proposed system will be used in the satellite payloads for the
next generation communications networks. To deal with these
requirements, two scenarios and their corresponding system architectures
are derived. For the first case, the number of inputs and outputs are equal,
while for the latter case, the number of outputs will be larger.

3.3.1. K-Input K-Output FFBR Networks

Generalizing the SISO system considered so far to a MIMO system
with equal number of inputs and outputs, the system in Figure 31 can be
used [32].
MIMO FFBR

In 1

AFB

SFB

Out 1

In 2

AFB

SFB

Out 2

In K

AFB

SFB

Out K

Channel Swtich

Figure 31: K-Input K-Output MIMO FFBR with Fixed Analysis and Synthesis FBs.

In this system, the analysis FBs (AFBs) and synthesis FBs (SFBs)
are instances of the fixed FBs used so far but the channel switch can
redirect the outputs from one input beam to another output beam. If the
SISO FFBR network is designed to satisfy the required BER, the overall
performance for each output subband in the MIMO network will be the
same as in the SISO network. In general, the satellite payload may have
different number of inputs and outputs which is the topic of the next
section.

65

3.3.2. S-Input K-Output FFBR Networks

To handle different number of inputs and outputs, the system
shown in Figure 32 can be used [32].
MIMO FFBR
In 1

AFB

SFB

Ch Co

Out 1 to K1

In 2

AFB

SFB

Ch Co

Out K 1 + 1 to K 2

In S

AFB

SFB

Ch Co

Out

K r + 1 to K

Channel Combiners
Channel Switch

Figure 32: S-Input K-Output MIMO FFBR with Fixed Analysis and Synthesis FBs.

In this system, K = RS and the output beams bandwidth is R

times narrower than that of the input beam [32]. However, this case can
be generalized to allow outputs with different data rates requiring
different downsampling factors at the outputs. In the implementation,
different instances of polyphase synthesis FBs must be used with some of
the DFT inputs set to zero. This keeps the required signal branches only
and the task of the channel combiners is to add the necessary SFB
outputs to form the desired signals. The channel switch can direct the
signals to baseband giving more flexibility in the FFBR network. The
next section will illustrate the simulation results on system functionality
and performance.

3.4.

Simulation Results

To test the system functionality and quality, some issues were

considered as follows:
Selection of system parameters
Construction of a transmitter/receiver pair with low BER
Implementation of the SISO system
Implementation of the MIMO system
The system performance was measured in the Mean Square Error (MSE)
sense. In other words, the variance of the error between the transmitted
66

and the received signals was calculated, which can be converted into
BER. The next sections will discuss these issues and illustrate the
simulation results.

3.4.1. System Parameters Selection

As discussed in Section 3.2.2, and in order to control the aliasing by
the stopband attenuation of the filters, the system parameters must satisfy
Equations 3.1 to 3.5. In the simulation of the system, the following
parameters were considered.
No.
1

Parameter
Number of Granularity Bands ( Q )

Value
4

2
3
4

Number of FB Channels ( N )
Downsampling Factor ( M )
Transition Band Width ( )

8
4

Number of Subbands ( q )

6
7
8

Prototype Filter Order ( D )

Phase Rotation Factor ( L )
Frequency Offset ( )

134
2
0.5

0.125

It must be added that according to Equation 3.12, the choice of the

prototype filter order to be a multiple of 2 M will make kr equal to unity
since mr will in any case be an integer.

3.4.2. Transmitter/Receiver Filter Design

The purpose of the transmitter/receiver pair is to evaluate the
quality of the FFBR system for different types of input data i.e. M-QAM
or Gaussian signals. Transmit and receive filters design is a traditional
communication problem. In brief, the transmit filter constrains the
transmitted signals spectrum to a limited bandwidth while the receive
filter rejects out of band noise thus maximizing SNR. Furthermore, the
cascade of the transmit and receive filter must minimize ISI, which
means that the convolution of the transmit and receive filters impulse
responses must satisfy [35]

67

1
,
M
h( n) =

0,

n = L 1

2
,

3.15

L +1
n = L 1 kM , k = 1,2,...,
1
2
2M

where n, L, and M are the filter impulse index, filter length, and
oversampling factor respectively. The oversampling factor relates the
sampling rate with the baud rate through
M =

Fs

Furthermore,

3.16

L +1

2M

must be an integer. This type of filter is called a

Nyquist filter. Having designed the equiripple linear-phase Nyquist filter

with a nonnegative frequency response, one can use the standard spectral
factorization methods [36] to extract transmit and receive filters.
However, the main focus of the thesis was to evaluate the FFBR system.
To do so, the designed receive filter passband covered passband and the
transition band of the M-band transmit filter with a sharp transition band
and large attenuation in the stopband as shown in Figure 33. This leads to
high order filters resulting in expensive receivers and a future research
topic will be to use the spectral factorization methods to reduce the order.
H (e j )
Dashed Line: Receive Filter
Solid Line: Transmit Filter

Figure 33: Transmit and Receive Filter Characteristics to Evaluate the FFBR Network.

In this thesis and for simulation purpose, third and sixth band filters
whose characteristics satisfied the constraints of the system were
designed. The MATLAB program to design these filters can be found in
Appendix A. These filters are used to form the user signals. To do so,
three sets of Gaussian or M-QAM signals are generated. The signals are
then upsampled and filtered with the third or sixth band filters,
respectively. Finally, different users are modulated to appropriate
68

positions in the frequency spectrum and summed to form a beam of

signals. The MATLAB program constructing two test beams for the
general MIMO case can be found in Appendix B.
Having constructed input beams, the transmit and receive filters with
characteristics shown in Figure 33 were designed. The designed
transmit/receive filters had a MSE in the order of 10 11 , so it could detect
larger errors caused by the FFBR system. The test setup to verify the
performance was as Figure 34.
Transmit/Receive Pair

Input
Data

Transmit
Filter

Receive
Filter

FFBR Network

MSE

Figure 34: Test Setup for FFBR Network Evaluation.

3.4.3. Implementation of the SISO System

As discussed before, the FFBR has two types of implementation as
shown in Figure 29 and Figure 30. One aim in this thesis was to verify
the equivalence of these systems in the presence of the channel switch.
To implement the system in Figure 29, the MATLAB program in
appendix C is used. It must be mentioned that this program implements a
general MIMO case. However, setting the number of inputs to one will
result in a SISO case. As mentioned before, the polyphase decomposition
can reduce the implementation complexity. To verify the system in
Figure 30, the MATLAB program in Appendix D is used. This program
implements the general MIMO case shown in Figure 31 and can easily be
converted to a SISO system by setting the number of inputs to one.
Regarding the polyphase implementation of the FFBR network, some
issues must be mentioned as follows:
Prototype Filter Polyphase Decomposition
Since the designed prototype filter in this thesis was linear-phase and it
had symmetry, the polyphase decomposition resulted in polyphase
components with different lengths. Otherwise, the prototype filter would
lack symmetry resulting in nonlinear-phase characteristics.
Consequently, the vectors at the outputs of the filter blocks in Figure 30
were of different lengths. Since the DFT/IDFT operation was
69

implemented in a matrix multiplication form, the outputs of the filter

blocks were zero padded to ease the multiplication process. However, in
a sample-based implementation, zero padding can be avoided thus
allowing vectors of different lengths.
DFT/IDFT Implementation
The DFT and IDFT operations can be modelled as the multiplication of a
column vector whose elements are the values of each branch in Figure 30
at time n , with square matrices given by Equations (1.30) and (1.36).
The result would also be a column vector whose elements are the values
of branches in Figure 30 at time n .
An important issue in the design of filter banks is the procedure to
design the prototype filter. Generally, two techniques namely minimax
and least-squares are used to design the prototype filter for filter banks.
In this thesis, the minimax algorithm was used and the MATLAB
program to design the prototype filter can be found in Appendix E.
To test the SISO system, a data pattern composed of three user
subbands was generated at the input of the FFBR Network. A channel
switch directs outputs of the analysis filters to different synthesis filters
according to the reallocation scheme. For the SISO case, redirection
occurs between branches of one filter bank. An example switch scheme
is shown in Figure 35.

Analysis Bank
Outputs

Synthesis Bank
Inputs

Figure 35: Example Channel Switch for SISO Case.

The input, output, and the analysis/synthesis filters of the FFBR

network with this switch is shown in Figure 36.

70

X3

X2

X1

X3

X2

X1

Figure 36: Input, Output, and Analysis Filters for SISO Polyphase FFBR Network.

As it can be seen, three user signals have been shifted to different

locations in the spectrum.

3.4.4. Implementation of the MIMO System

Extension of the SISO case to MIMO is done by increasing the
instances of the filter bank and adding a channel switch capable of
directing signals from one filter bank to another. An example of the
switch structure for the case with two inputs and two outputs is shown in
Figure 37 .
From AFB 2

From AFB 1

To SFB 2

To SFB 1

Figure 37: Example Channel Switch for Two-Input Two-Output MIMO FFBR Network.

71

In this scheme, two input beams each containing three different

users are multiplexed into two output beams where four users share one
beam and the remaining two users are present in the other beam. The
inputs and outputs of the FFBR network in Figure 31 with the switch in
Figure 37 are depicted in Figure 38.

X3

X2

X1

X4

X2

X6

X6

X5

X4

X5

X3

X1

Figure 38: Inputs and Outputs for MIMO FFBR Network with two Inputs and two Outputs.

As discussed in Section 3.3.2, by increasing the number of SFBs, the

FFBR Network can handle S Inputs and K = RS , R > 1 outputs. To have
more flexibility, it is desired to direct all the signals to baseband. This
needs modifications in the channel switch as well as setting some of the
DFT inputs of the SFBs to zero. The latter removes the unnecessary
branches.
As an example, duplicating the AFB output, setting branches five to
eight in the first SFB, and setting branches one to four in the second SFB
to zero keeps the required user signals without shifting them to baseband.
The input and outputs of the FFBR network in Figure 32, without any
channel switch are shown in Figure 39.

72

X2

X1

X1

X2

Figure 39: Input and Outputs of the FFBR Network without Channel Switch.

The use of channel switch can direct the signals to baseband. An example
channel switch to shift the user signal X 2 to baseband is shown in Figure
40.
From AFB 2

From AFB 1

To SFB 2

To SFB 1

Figure 40: Example One-Input/Two-Output Channel Switch for MIMO FFBR Network.

The outputs of the system with this channel switch are shown in Figure
41.

73

X2

X1

X1

X2

Figure 41: Input and Outputs of the FFBR Network with Channel Switch of Figure 40.

As it can be seen, the outputs are at baseband increasing the multiplexing

flexibility of the network.

3.5.
Finite Word Length Effects on the FFBR
Network
Any system designer deals with the tradeoffs of the system cost and
performance. The more bits we specify for the system, the better
performance we get. However, for some applications, a specific
performance (usually measured in BER) is required, which will help the
designer decide on the system resolution. To evaluate the FFBR network
performance, the quantization effects were introduced in the filter
coefficients, filter outputs, and DFT/IDFT outputs as shown in Figure 42.

74

 kr

x ( n)
z 1

P0 ( z LW NN )

P1 ( z W
L

N
N

Q
Q

.
.
.

z 1
M

N 1

N 1 P ( z LW N )
N 1
N

N 2 P

N 2

(z WN )
L

N 1

N 1

0 P ( z LW N )
N
0

z 1
M

DFT

IDFT

PN 1 ( z LW NN )

.
.
.
Q

z 1
M

y ( n)

Channel Switch

Figure 42: Quantization in the Polyphase FFBR Network.

Having the resolution of the data bits at the output of the filter
block will help us define the required number of bits according to the
filter implementation structure. In the quantization scheme, we assume
different resolutions at different branches of the filter bank. This needs an
investigation on the propagation of error in different branches of the
system and is a future research topic. To illustrate the effects of finite
word length on the FFBR network performance, Figure 43 shows the
constellation for a 64-QAM data multiplexed according to the channel
switch in Figure 37 for three different filter coefficient lengths.

75

Figure 43: Multiplexed 64-QAM Data Constellation for Three Filter Coefficient Lengths.

This figure further proves the fact that the stopband attenuation of
the prototype filter suppresses aliasing and the designer can make
tradeoffs according to the required BER. To compare the system MSE
for different stopband attenuations, we can utilize the fact that since the
transmit/receive pair has a MSE in the order of 1011 , it can detect larger
errors. To illustrate the MSE trend for different stopband characteristics,
the variance of the input/output difference for six different channels of
Figure 38 is shown in Figure 44.

76

X1
X2
X3
X4
X5

X6

Figure 44: FFBR Network Noise Variance for Channels in Figure 38.

The results show the same trend for all the channels and are in
accordance with the fact that the attenuation of the stopband suppresses
aliasing. It must be mentioned that to achieve lower BER, the prototype
filter must have larger attenuation and for higher attenuation, the
difference between noise variance in different channels is negligible.
Thus, the noise behaviour of system is stable for high attenuation values.
It is important to note that the noise is white and Gaussian.

3.6.

Concluding Remarks and Future Topics

The proposed FFBR network is based on a new class of variable

oversampled complex modulated N -channel FBs, which has fixed
decimation and interpolation ratios M in the final implementation. The
network handles a variable q input and output user subbands where
1 q Q . The proposed system architecture uses the following:
Oversampled FB: The oversampled FBs have the advantage of
easy suppression of aliasing allowing the combination of smaller

77

subbands into wider subbands without introducing large aliasing

distortion. This property brings full flexibility to the system.
More FB channels than granularity bands: This helps generate all
possible frequency shifts.
Complex modulated FBs: This results in very low complexity and
simplicity in terms of analysis, design, and implementation.
By properly selecting N , M , and analysis/synthesis filters with given a
maximum value of Q , this new class of FBs can:
Handle all possible frequency shifts and all possible user subband
widths.
Achieve as low complexity as in regular complex modulated FBs.
Achieve as much parallelism as in any of the previously existing
FFBR methods.
Approximate PR as close as desired via a proper design.
Easily be analyzed, designed, and implemented compared to
previously existing FFBR networks.
In comparison with other FFBR systems, the proposed system can:
Outperform the regular modulated FB based networks in terms of
flexibility.
Outperform the tree-structured FB based networks in terms of
flexibility and complexity.
Outperform the overlap/save DFT/IDFT based networks in terms
of PR.
Furthermore, both tree-structured FB and overlap/save DFT/IDFT based
networks are more complicated to analyze and design.
The future research topics can be as follows:
Analysis of the error propagation in the FB branches: This will
help define different word lengths for different branches of the
system leading to tradeoffs on complexity and performance.
Application of the general polyphase decomposition: As
discussed in Section 3.2.6, the variable L is an integer. However,
if L is not an integer, the general polyphase decomposition must
be used.
The study on transmit/receive filters for the test setup. In this
thesis, the purpose was to evaluate the FFBR network, therefore
the chosen transmit/receive filters had high orders making the
receiver expensive. Reduction of the orders using spectral
factorization methods is a future research topic.

78

References
[1]
[2]

[3]
[4]
[5]
[6]
[7]
[8]
[9]
[10]
[11]

[12]
[13]

Wittig M., Satellite onboard processing for multimedia

applications, IEEE Comm. Mag., vol. 38, no. 6, pp. 134140, June 2000.
Craig A.D., Leong C.K., Wishart A., Digital signal
processing in communications satellite payloads,
Electronics & Comm. Eng. J., vol. 4, no 3, pp. 107-114,
June 1992.
P.P. Vaidyanathan, A tutorial on multirate digital filter
banks, Proc. IEEE Int. Symp. Circuits Syst., pp. 22412248, June 1988.
Mitra S.K., Structural subband decomposition: a new
concept in digital signal processing, Proc. IEEE Int. Conf.
ASSP, vol. 1, pp. 31-34, 21-24 Apr. 1997.
P.P. Vaidyanathan, Multirate digital filters, filter banks,
polyphase networks, and applications: a tutorial, Proc.
IEEE, vol. 78, no. 1, pp. 56-93, Jan. 1990.
See May Phoong and P.P. Vaidyanathan, Time varying
filters and filter banks: some basic principles, IEEE Trans.
SP, vol. 44, no. 12, pp. 2971-2987, Dec. 1996.
R.E. Crochiere and L.R. Rabiner, Multirate Digital Signal
Processing, Englewood Cliffs, NJ: Prentice Hall, 1983.
George L. Turin, An introduction to matched filters, IRE
Trans. Information Theory, 1960.
T.P. Barnwell III and M.J.T. Smith, Filter banks for
analysis-reconstruction systems: a tutorial, Proc. IEEE Int.
Symp. Circuits Syst., pp. 1993-2003, May 1990.
Marc Moonen, DSP II Course, Lecture Notes 5-8:
http://homes.esat.kuleuven.be/~rombouts/dspII/
S. Radhakrishnan Pillai and Gregory H. Allen,
Generalized magnitude and power complementary filters,
Proc. IEEE Int. Conf. ASSP, vol. 3, pp. 585-588, 19-22
Apr. 1994.
Cvetkovic Z., Vetterli M., Oversampled filter banks,
IEEE Trans. SP, vol. 46, no. 5, pp. 1245-1255, May 1998.
Proakis, John G., Digital Communications, McGraw-Hill,
1995.

79

[14]
[15]
[16]

[17]

[18]

[19]

[20]
[21]

[22]
[23]
[24]
[25]
[26]

Princen, J., The design of nonuniform modulated

filterbanks, IEEE Trans. SP, vol. 43, no. 11, pp. 25502560, Nov. 1995.
Alan V. Oppenheim, Ronald W. Schafer, and John R. Buck,
Discrete-Time Signal Processing, Prentice Hall, 2nd
edition, Feb. 1999.
Karp, T. and Fliege, N.J., Modified DFT filter banks with
perfect reconstruction, IEEE Trans. Circuits Syst. II:
Analog and Digital Signal Processing, vol. 46, no. 11, pp.
1404-1414, Nov. 1999.
Weiss, S., Harteneck, M., and Stewart, R.W., On
implementation and design of filter banks for subband
adaptive systems, Proc. IEEE Workshop SP Syst., SIPS
98, pp. 172-181, 8-10 Oct. 1998.
Nguyen, T.Q. and Koilpillai, R.D., The theory and design
of arbitrary-length cosine-modulated filter banks and
wavelets, satisfying perfect reconstruction, IEEE Trans.
SP, vol. 44, no. 3, pp. 473-483, Mar. 1996.
Kumar R., Nguyen T.M., Wang C.C., Goo G.W., Signal
processing techniques for wideband communications
systems, Proc. IEEE Military Comm. Conf., MILCOM
1999, vol. 1, pp. 452-457, 31 Oct.-3 Nov. 1999.
Douglas B. Williams and Vijay Madisetti, The Digital
Signal Processing Handbook, CRC Press, 1999.
P.P. Vaidyanathan and B. Vrcelj, Transmultiplexers as
precoders in modern digital communication: a tutorial
review, Proc. IEEE Int. Symp. Circuits Syst., pp. 405-412,
May 2004.
M. J. Vetterli, A theory of multirate filter banks, IEEE
Trans. ASSP, vol. 35, no. 3, pp. 356-372, Mar. 1987.
Ramachandran R.P. and Kabal P., Configuration and
performance of modulated filter banks, Proc. IEEE Int.
Symp. Circuits Syst., pp. 1809-1812, 1-3 May 1990.
R.A. Gopinath and C.S. Burrus, A tutorial overview of
filter banks, wavelets, and interrelations, Proc. IEEE Int.
Symp. Circuits Syst., pp. 104-107, May 1993.
Robert M. Gray, Toeplitz and Circulant Matrices: A
Review, Department of Electrical Engineering, Stanford
University, http://www-ee.stanford.edu/~gray/toeplitz.pdf.
Akansu A.N., Duhamel P.M, Xueming Lin, de Courville
M., Orthogonal transmultiplexers in communication: a
80

[27]

[28]

[29]

[30]

[31]
[32]

[33]

[34]
[35]

[36]

review, IEEE Tran. SP, vol. 46, no. 4, pp. 979-995, Apr.
1995.
Andersen B.R., Digital filter bank designs for satellite
transponder payloads: implementation on VLSI circuits,
5th IEEE Int. Conf. Universal Personal Comm., vol. 2, pp.
750-754, 29 Sept.-2 Oct. 1996.
Tien Nguyen, Hant J., Taggart D., Chit-Sang Tsang,
Johnson D.M., Jo-Chieh Chuang, Design concept and
methodology for the future advanced wideband satellite
system, Proc. IEEE Military Comm. Conf., MILCOM
2002, vol. 1, pp. 189-194, 7-10 Oct. 2002.
H. G. Gckler and B. Felbecker, Digital onboard FDMdemultiplexing without restrictions on channel allocation
and bandwidth, 7th Int. Workshop on Digital Sign.
Processing Techn. for Space Comm., 1-3 Oct. 2001,
Sesimbra, Portugal.
B. Arbesser-Rastburg, R. Bellini, F. Coromina, R. De
Gaudenzi, O. del Rio, M. Hollreiser, R. Rinaldo, P. Rinous,
and A Roederer, R&D directions for next generation
broadband multimedia systems: an ESA perspective, Int.
Comm. Satellite Syst. Conf., Montreal, May 2002.
Del Re E., Pierucci L., Next-generation mobile satellite
networks, IEEE Comm. Mag., vol. 40, no. 9, pp. 150-159,
Sept. 2002.
H. Johansson and P. Lwenborg, Flexible frequency band
reallocation network based on variable oversampled
complex modulated filter banks, to appear in European J.
Applied SP, 2006.
H. Johansson and P. Lwenborg, Flexible frequency band
reallocation network based on variable oversampled
complex modulated filter banks, Proc. IEEE Int. Conf.
Acoust. Speech, Signal Processing, Philadelphia, USA,
Mar. 2005.
P.P. Vaidyanathan, Mulitrate Systems and Filter Banks,
Englewood Cliffs, NJ: Prentice-Hall, 1993.
Sullivan J.L., Adams J.W., Reisner R.A., Armstrong R.L.,
New optimization algorithm for digital communication
filters, Proc. 36th Asilomar Conf. Signals, Syst., and
Computers, vol. 1, pp. 323-327, 3-6 Nov. 2002.
Samueli H., On the design of optimal equiripple FIR
digital filters for data transmission applications, IEEE
81

Trans. Circuits Syst., vol. 35, no. 12, pp. 1542-1546, Dec.
1988.

82

Appendixes

83

84

Appendix A: MATLAB Program to Design

Third and Sixth Band Filters
warning off;close all;clc;clear all;format long
wsT=0.21875*pi;Mth_Band=6;N=89; % 6th band
%wsT=0.46875*pi;Mth_Band=3;N=25;% third band
t=1;M=N/2+1;m=1:M;Ks=1000;wT=linspace(wsT,pi,Ks) ;
D=zeros(1,Ks);W=ones(1,Ks) ;A=[trigMat(t,m,wT) -1./W'] ;
A=[A' [-trigMat(t,m,wT) -1./W']']';
b=[D -D]';c=[zeros(1,M) 1]';
vlb=1/Mth_Band ; vub=1/Mth_Band ;
vlb(2:M)=-1 ; vub(2:M)=1 ;
for k=Mth_Band+1:Mth_Band:M
vlb(k)=0 ;
vub(k)=0 ;
end
x=linprog(c,A,b,[],[],vlb,vub) ;
h=[0.5*fliplr(x(2:M)') x(1) 0.5*x(2:M)'] ;
wT=linspace(0,pi,4000) ;
H=freqz(h,1,wT) ;Mag=20*log10(abs(H)) ;plot(wT/pi,Mag);grid on
figure(2);plot(Mag(1:790));grid on
find(h == 1/Mth_Band)
find(h == 0)
figure(3);plot(h);grid on

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86

Appendix B: MATLAB Program to Generate

User Signals
function[y,x60,x30,x6,x3]=MIMO_Transmitter(QAM_Type,Sample_Number,h6,h3,Ch
_Shift,Type,Data_Bits,Desired_Mean,Desired_Variance)
if (Type == 1)
x60(1,:)=Desired_Mean+sqrt(Desired_Variance)*randn(1,Sample_Number);
x60(2,:)=Desired_Mean+sqrt(Desired_Variance)*randn(1,Sample_Number);
x60(3,:)=Desired_Mean+sqrt(Desired_Variance)*randn(1,Sample_Number);
x60(4,:)=Desired_Mean+sqrt(Desired_Variance)*randn(1,Sample_Number);
Sample_Number = Sample_Number /2 ;
x30(1,:)=Desired_Mean+sqrt(Desired_Variance)*randn(1,Sample_Number);
x30(2,:)=Desired_Mean+sqrt(Desired_Variance)*randn(1,Sample_Number);
else
x60(1,:)=qam(Sample_Number,QAM_Type);
x60(2,:)=qam(Sample_Number,QAM_Type);
x60(3,:)=qam(Sample_Number,QAM_Type);
x60(4,:)=qam(Sample_Number,QAM_Type);
Sample_Number = Sample_Number /2 ;
x30(1,:)=qam(Sample_Number,QAM_Type);
x30(2,:)=qam(Sample_Number,QAM_Type);
end
x6(1,:)=6*conv(upsample(x60(1,:),6),h6);n=0:length(x6(1,:))1;x6(1,:)=x6(1,:).*exp(j*Ch_Shift(1,1)*pi*n);
x6(2,:)=6*conv(upsample(x60(2,:),6),h6);n=0:length(x6(2,:))1;x6(2,:)=x6(2,:).*exp(j*Ch_Shift(1,3)*pi*n);
x6(3,:)=6*conv(upsample(x60(3,:),6),h6);n=0:length(x6(3,:))1;x6(3,:)=x6(3,:).*exp(j*Ch_Shift(2,2)*pi*n);
x6(4,:)=6*conv(upsample(x60(4,:),6),h6);n=0:length(x6(4,:))1;x6(4,:)=x6(4,:).*exp(j*Ch_Shift(2,3)*pi*n);
x3(1,:)=3*conv(upsample(x30(1,:),3),h3);n=0:length(x3(1,:))1;x3(1,:)=x3(1,:).*exp(j*Ch_Shift(1,2)*pi*n);
x3(2,:)=3*conv(upsample(x30(2,:),3),h3);n=0:length(x3(2,:))1;x3(2,:)=x3(2,:).*exp(j*Ch_Shift(2,1)*pi*n);
y(1,:)= x6(1,:)+[x3(1,:) zeros(1,( length(x6(1,:))-length(x3(1,:)) ))]+x6(2,:);
y(2,:)= x6(3,:)+[x3(2,:) zeros(1,( length(x6(3,:))-length(x3(2,:)) ))]+x6(4,:);
y=quant(y,2^(-Data_Bits+1)) ;

87

88

Appendix C: MATLAB Program to Implement

the System in Figure 29
function FBR_Out =
HL_MIMO(x,Shift,N,M,L,h0,alpha,wT,High_Level_FBR_Out_Data_Bits,Num_Inputs)
%**********************************************************
h=[];g=[];n=0:length(h0)-1;K=length(n)-1;
for k=0:N-1 h(k+1,1:K+1)=h0.*exp(j*((k+0.5)*(nK/2)*2*pi/N));g(k+1,1:K+1)=h0.*exp(j*((k+0.5)*(n-K/2)*2*pi/N));end
%**********************************************************
for k=0:N-1
for l=0:Num_Inputs-1
vtemp( (l*N)+k+1 , : )=conv( x(l+1,:) , h(k+1,:) );
vtemp_down( (l*N)+k+1 , : )=downsample( vtemp((l*N)+k+1,:) , M);
end
end
%**********************************************************
Channel_Switch_Out=[];
for k=0:N-1
for l=0:Num_Inputs-1
Channel_Switch_Out((l*N)+k+1+Shift((l*N)+k+1),:) = vtemp_down((l*N)+k+1,:)
;
end
end
%**********************************************************
vmat=[];
for k=0:N-1
for l=0:Num_Inputs-1
vmat( (l*N)+k+1 , : )= upsample( Channel_Switch_Out( (l*N)+k+1,: ) ,M);
end
end
%**********************************************************
ymat=[];
for k=0:N-1
for l=0:Num_Inputs-1
ymat( (l*N)+k+1 , : )=conv( vmat((l*N)+k+1,:) , g(k+1,:) );
end
end
FBR_Out=zeros(Num_Inputs,length(ymat(1,:)));
for k=0:N-1
for l=0:Num_Inputs-1
FBR_Out( l+1 , : )=FBR_Out( l+1 , : )+ymat( (l*N)+k+1 , : );
end
end
for l=0:Num_Inputs-1

89

FBR_Out(l+1,:)=M* (quant(FBR_Out(l+1,:),2^(High_Level_FBR_Out_Data_Bits(l+1)+1))) ;
end

90

Appendix D: MATLAB Program to Implement

the System in Figure 31
function [FBR_Out] =
Poly_MIMO(x,Shift,N,M,L,h0,alpha,wT,Filter_Out_Bits,FFT_Out_Bits,Final_Out_Bit
s,Num_Inputs)
D = length(h0) -1 ;n=0:length(h0)- 1 ;K=length(n) - 1 ;h=[];g=[];
for k=0:N-1
h(k+1,1:K+1)=h0.*exp(j*((k+alpha)*(n-K/2)*2*pi/N));
g(k+1,1:K+1)=h0.*exp(j*((k+alpha)*(n-K/2)*2*pi/N));
end;
for k = 0:N-1 Poly(1,k+1) = {h0(k+1:N:end)};end
for k = 0:N-1
Temp_P = Poly{k+1} ;
for m = 1:length(Poly{k+1})
Temp_P(m) = ((-1)^(m+1)) .* Temp_P(m) ;
end
P(1,k+1) = {Temp_P};
end
for k = 0:N-1 P_Up(k+1,:) = {upsample(P{1,k+1},L)}
;end;P_Upsampled_Flipped=P_Up;
%****************delayed versions of the input signal
Delayed_x=[];
for k=0:N-1
for l=0:Num_Inputs-1
Delayed_x((l*N)+k+1,:)= [zeros(1,k) x(l+1,1:end-k)];
end
end
Decimated_Delayed_x=[];
for k=0:N-1
for l=0:Num_Inputs-1
Decimated_Delayed_x((l*N)+k+1,:)=downsample(Delayed_x((l*N)+k+1,:),M);
end
end
%****************filtering
vtmep=[];
for k=0:N-1
for l=0:Num_Inputs-1
vtemp{(l*N)+k+1}=quant(conv(Decimated_Delayed_x((l*N)+k+1,:),P_Upsampled_Fli
pped{k+1}),2^(-Filter_Out_Bits((l*N)+k+1)+1));
end
end
% all the inputs should have the same length .zero padd if needed
new_vtemp = [] ;

91

for k=0:Num_Inputs*N-1
if (length(vtemp{k+1}) ~= length(vtemp{1}) )
new_vtemp(k+1,:) = [ vtemp{k+1} zeros(1,length(vtemp{1})length(vtemp{k+1}))];
else
new_vtemp(k+1,:) = vtemp{k+1} ;
end
end
vtemp = [];vtemp = new_vtemp ;
% multiplication by alpha
for k=0:N-1
for l=0:Num_Inputs-1
vtemp((l*N)+k+1,:)=vtemp((l*N)+k+1,:) * exp(j*(2*pi/N)*alpha*k);
end
end
%*********IDFT operation in matrix form
DFT_Matrix=fft(eye(N)) ;IDFT_Matrix=DFT_Matrix';vtemp_IDFT=[];
for m=0:length(vtemp(1,:))-1
for l=0:Num_Inputs-1
vtemp_IDFT((l*N)+1:(l*N)+N,m+1)=quant((IDFT_Matrix *
vtemp((l*N)+1:(l*N)+N,m+1)),2^(-FFT_Out_Bits((l*N)+k+1)+1));
end
end
%*********multiplication by beta
for k=0:N-1
for l=0:Num_Inputs-1
vtemp_IDFT((l*N)+k+1,:)=vtemp_IDFT((l*N)+k+1,:) * exp(j*(2*pi/N)*(k+alpha)*(D/2));
end
end
%*********channel switch
Channel_Switch_Out=[];
for k=0:N-1
for l=0:Num_Inputs-1
Channel_Switch_Out((l*N)+k+1+Shift((l*N)+k+1),:) = vtemp_IDFT((l*N)+k+1,:)
;
end
end
%********************
for k=0:N-1
for l=0:Num_Inputs-1
vtemp_IDFT_Mu_Gama((l*N)+k+1,:)=Channel_Switch_Out((l*N)+k+1,:) *
exp((+j*(2*pi/N))*(((k+alpha)*D/2)-k));
end
end
%*******************
for m=0:length(vtemp_IDFT_Mu_Gama(1,:))-1
for l=0:Num_Inputs-1

92

vtemp_DFT((l*N)+1:(l*N)+N,m+1)=quant((DFT_Matrix *
vtemp_IDFT_Mu_Gama((l*N)+1:(l*N)+N,m+1)),2^(-Filter_Out_Bits((l*N)+k+1)+1));
end
end
%*******************reverse alpha multiplication
for k=0:N-1
for l=0:Num_Inputs-1
vtemp_DFT((l*N)+k+1,:)=vtemp_DFT((l*N)+k+1,:) * exp(j*(2*pi/N)*alpha*(N-1k));
end
end
%*******************reverse filtering
for k=0:N-1
for l=0:Num_Inputs-1
vtmep_DFT_fil{(l*N)+k+1}=quant(conv(vtemp_DFT((l*N)+k+1,:),P_Upsampled_Flip
ped{N-k}),2^(-FFT_Out_Bits((l*N)+k+1)+1));
end
end
new_vtemp=[];
%*******************
for k=0:Num_Inputs*N-1
if (length(vtmep_DFT_fil{k+1}) ~= length(vtmep_DFT_fil{N}) )
new_vtemp(k+1,:)=[vtmep_DFT_fil{k+1} zeros(1,length(vtmep_DFT_fil{N})length(vtmep_DFT_fil{k+1}))];
else
new_vtemp(k+1,:)=vtmep_DFT_fil{k+1} ;
end
end
%*******************
for k=0:Num_Inputs*N-1
vtmep_DFT_fil_up(k+1,:)=upsample(new_vtemp(k+1,:),M);
end
new_vtemp=[];new_vtemp=vtmep_DFT_fil_up;
%*******************
for k=0:N-1
for l=0:Num_Inputs-1
Delayed_new_vtemp((l*N)+k+1,:)=[zeros(1,N-1-k) new_vtemp((l*N)+k+1,1:end(N-1-k))];
end
end;
%*******************
y = zeros(Num_Inputs,length(Delayed_new_vtemp(1,:)));
for k=0:N-1
for l=0:Num_Inputs-1
y(l+1,:)=y(l+1,:)+Delayed_new_vtemp((l*N)+k+1,:);
end
end
for l=0:Num_Inputs-1

93

FBR_Out(l+1,:)=M* (quant(y(l+1,:),2^(-Final_Out_Bits(l+1)+1))) ;
end

94

Appendix E: MATLAB Program to Design

Prototype Filters Using Minimax Algorithm
clc;close all;clear all
for iteration=0:70
M=4;N=8;Q=4;delta=0.125*pi/Q;
PB_dB=0.0000025
SB_dB=2+iteration
A=10^((PB_dB)/20);dc=(A-1)/(A+1)
B=10^((SB_dB)/20);ds=(1+dc)/B
[n0,f,m0,w]=remezord([pi/N-delta pi/N+delta],[1 0],[dc ds],2*pi);
n0=8*(fix(n0/8)+1)
h0=remez(2*round(n0/2),f,m0,w);n=0:length(h0)-1;K=length(n)-1;
options=foptions;options(1)=1;
wT=linspace(0,pi/N+delta,250);wsT=linspace(pi/N+delta,pi,200);
x0=[0.1 h0(1:K/2+1)];
x=minimax('FIR_fun_new',x0,options,[],[],[],wT,wsT,N,K,n);
h0=x(2:length(x));h0=[h0 fliplr(h0(1:length(h0)-1))];
h(iteration+1)={h0}
end;save Prototypes_25eMinus7_0to70.mat h
function [f,g]=FIR_fun_new(x,wT,wsT,N,K,n)
f=x(1);h0=x(2:length(x));h0=[h0 fliplr(h0(1:length(h0)-1))];
H=freqz(h0,1,wT);Hc=freqz(h0,1,wT-2*pi/N);
g1=abs(abs(H).*abs(H)+abs(Hc).*abs(Hc)-1)-10*x(1);
H=freqz(h0,1,wsT);g2=abs(H)-x(1); g=[g1 g2];

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