S-88.

3104 Digital Signal Processing Systems (6 cr)

Fall semester, I period, 2013

Risto Wichman
Course data
From 2009 onward: lectures in English only
Lectures start on September 9
th
, 2+2 hours during 6
weeks
Exercises start on September 10
th
Teaching asistant: Ejaz Aqib
Course book: S.K. Mitra: Digital Signal Processing. A
Computer-based Approach, McGraw-Hill, 3rd Edition,
2006/ 4
th
Edition, 2011
Chapters 13-14 + some things from Chapters 7 and 10
Course material is in Noppa.
6 credits based on lectures (exam) and mandatory homeworks
! Basics of digital signal processing (eg., Aalto course
T-61.3010/ELEC-C5230) or equal knowledge (Signals
and systems etc.).
! Basic Matlab knowledge will help in solving the Matlab-
homework problems. Simulink is also acceptable
" Freeware Octave, www.octave.org, can be used as
well

Course prerequisities
To pass the course you must pass the exam and mandatory
homework assignments.
Alternative way to pass: Make Simulink + Pro Audio
development kit C6727-PADK work together
Lectures: background and theory, Matlab examples
Concentrate on few concepts and learn them well (instead
of superficially touching dozens of topics)!
Exercises: Application of concepts presented in the lectures
Exam: No additional material (calculators, tables of formulas
etc.) is allowed
Questions do not stay the same, old exams in web
Course arrangement

! Final grade of the course comes from the following formula:
Final grade = min[1 ; E ; H] * round{max[E ; ((0.4 + BIAS)*H + 0.6*E)]}
where
" E is the grade of the exam (0, 1, 2, 3, 4 or 5),
" H is the grade of the homework (0, 1, 2, 3, 4 or 5) and
" BIAS = 0.02.
" About 50% of points are needed to pass the course
! This formula will give the final grades as illustrated in the table below.


E = 0 E = 1 E = 2 E = 3 E = 4 E = 5
H = 0 0 0 0 0 0 0
H = 1 0 1 2 3 4 5
H = 2 0 1 2 3 4 5
H = 3 0 2 2 3 4 5
H = 4 0 2 3 3 4 5
H = 5 0 3 3 4 5 5
Course grading
Exercise system
Exercise paper + hint session on Tuesday.
Q&A session on Friday, new and old exercises
Deadline to return solutions to the box in G4 on Monday
Course contents
Assumed background knowledge
Discrete-Time Signals and Systems
Discrete-Time Signals
Typical Sequences and Sequence Representation
The Sampling Process
Discrete-Time Systems
Time-Domain Characterization of LTI Discrete-Time Systems
Simple Interconnection Schemes
Finite-Dimensional LTI Discrete-Time Systems
Classification of LTI Discrete-Time Systems
Correlation of Signals
Discrete-Time Fourier Transform
The Continuous-Time Fourier Transform
The Discrete-Time Fourier Transform
Discrete-Time Fourier Transform Theorems
The Frequency Response of an LTI Discrete-Time System
Phase and Group Delays
From ELEC-C5230
Assumed background knowledge
Digital Processing of Continous-Time Signals
Sampling of Continuous-Time Signals
Analog Lowpass Filter Design
Anti-Aliasing Filter Design
Reconstruction Filter Design
Effect of Sample-and-Hold Operation
Finite-Length Discrete Transforms
Orthogonal Transforms
The Discrete Fourier Transform
Relation Between the Fourier Transform and the DFT, and Their Inverses
Operations on Finite-Length Sequences
Classification of Finite-Length Sequences
Linear Convolution Using DFT
Assumed background knowledge
z-Transform
Definition and Properties
Rational z-Transforms
Region of Convergence of a Rational z-Transform
The Inverse z-Transform
z-Transform Properties
Computation of the Convolution Sum of Finite-Length Sequences
The z-Transform Function
Discrete-Time Systems in the Transform Domain
Transfer Function Classification Based on Magnitude Characteristics
Transfer Function Classification Based on Phase Characteristics
Types of Linear-Phase Transfer Functions
Simple Digital Filters
Assumed background knowledge
Digital Filter Structures
Block Diagram Reprensentation
Basic FIR Digital Filter Structures
Basic IIR Filter Structures
IIR Digital Filter Design
Bilinear Transform Method of IIR Filter Design
Design of Lowpass IIR Digital Filters
Design of Highpass, Bandpass, and Bandstop IIR Digital Filters
Spectral Transformations of IIR Filters
IIR Filter Design Using Matlab
Computer-Aided Design of IIR Digital Filters
FIR Digital Filter Design
FIR Filter Design Based on Windowed Fourier Series
Computer-Aided Design of Equiripple Linear-Phase FIR Filters
FIR Digital Filter Design Using Matlab
! A key characteristic of multirate algorithms is their high
computational efficiency facilitating an efficient
implementation
! Altering the sampling rate at some point of the process
creates several opportunities and enhancements for the
filter design.
! The techniques are mostly related to aliasing due to the
sampling rate change.
! Elementary theory and simple building blocks lead to
sophisticated filter and system designs.
" Combinations abound. No use to learn solutions by heart but
learn the principles

Course contents
Course contents
Classical treatment of digital signal processing
Filter (frequency response) design but no filtering
Filters with fixed coefficients i.e. no adaptive filters
Noise attenuation is not considered
Delay from filtering is not considered
Real-coefficient filters, not complex ones (except in
uniform filter banks)
Basic building blocks
Up-sampling and down-sampling
Discrete Fourier Transform (DFT)
Polyphase decomposition
Everything in the course can be derived and understood
based on these three basic things

! Up-sampling and down-sampling
" Aliasing
" Linear but time-variant operations
" Noble identities
" Fractional (rational) sampling rate change
" Sampling of band-pass signals
! Properties of linear-phase FIR filters
" Magnitude response, amplitude/zero-phase response
! Window-based design of FIR filters
! Sparse impulse response
! Multistage filters
! No IIR filters in this course
Basic concepts
Basic concepts
! Polyphase decomposition, DFT
! Interpolation
" Interpolation of samples vs. filter coefficients
! Nyquist filters
! Modulated filter banks
! QMF banks
" Perfect reconstruction
" Power complementary filters
" Power symmetric filters
! L-channel QMF banks
! Multilevel filter banks
! Filter banks and wavelets
! Multistage Filters
" Cascaded integrator comb filters (CIC)
" Interpolated FIR filters (IFIR)
" Complementary filters
! Interpolation, sampling rate change, fractional delay
" Direct implementation of H(z)
" Polyphase implementation of H(z)
" Lagrange interpolation (no splines)
" Farrow filters
" Polynomial approximation of polyphase structure
Filter structures
! Nyquist filters
" Half-band filters
" Lth-band filters
" Raised cosine filters (type of Lth band filters)
" Ideal band-limited filter
! Modulated filter banks
" Discrete Fourier transfrom (DFT) banks
" Non-critically sampled DFT banks
" OFDM and filter banks


Filter structures
! QMF banks
" Half-band product filter
" Linear-phase FIR QMF banks
" Biorthogonal filter banks
" Orthogonal filter banks
" Paraunitary filter banks
" Lattice realization

Filter structures
Filter structures
! L-channel filter banks
! Modulated filter banks
! Direct inverse of modulation matrix H
(m)
(z)
! Polyphase representation of L-channel QMF bank
! Lattice L-channel QMF bank
! Multilevel, octave filter banks
" Mother wavelet, scaling function
Some Applications
Analog low-pass filter
must compensate the
in-band distortion of
S&H, provide -80 dB
attenuation in the
stopband and have a
(24.1-20) KHz
transition band
Compact-disk 4-to-1
Oversampling
Due to oversampling, the
analog low-pass filter may
have a transition band of
(176.4-40) KHz, and the
system becomes more cost-
efficient
Direct conversion transmitter for OFDM
Orthogonal frequency division multiplexing (OFDM)
modulation
OFDM is a simple transmultiplexer - filterbank
The same technique as in the previous slide to relax the
specifications of the analog Tx filter
Dirty RF
Subband systems and transmultiplexers
Filterbank Transmultiplexer
Material
Course material
Lecture slides 13.1 -13.6 and 14.1-14.5 from Mitra 4
th

edition
All things in the slides are not covered: spline interpolation and
IIR filters skipped
Supplementary slides
Bandpass sampling
Farrow filters
Commutator representation of interpolation and decimation
(Root) raise cosine Nyquist filters, OFDM
Non-critically sampled filter banks, interpolation
Lattice QMF structure
Tiling patterns of tree-structured filter banks
Course material
Matlab examples
Further illustrate the concepts in lecture slides
Naming system
rw_program_X.m: Modified from the programs that came with
Mitras book
example_X.m: Examples in the book but not among the programs
accompanying the book
Some_name.m: Other things that have come up the lectures
! S.K. Mitra: Digital Signal Processing. A Computer-based
Approach, McGraw-Hill, 3rd Ed., 2006, 4
th
Ed. 2011
" Course book 2006-
" Matlab examples, didactic treatment
" No communications applications
! f.j. harris: Multirate Signal Processing for Communication
Systems, Prentice-Hall, 2005
" Practical, real world applications
" Theory casually explained
! N. Fliege: Multirate Signal Processing
" Course book -2005
" Lots of stuff
Books on the subject
! P. Diniz, E. da Silva, S. Netto, Digital Signal Processing -
System Analysis and Design, 2nd Edition, Cambridge
University Press, 2010
! Contains more advanced concept that Mitras book
! P.P. Vaidyanathan: Multirate Systems and Filter Banks,
Prentice-Hall, 1993
" The book on multirate systems
" Theoretical, advanced
! A.N. Akansu, R.A. Haddad: Multiresolution Signal
Decomposition, Academic Press, 2nd Edition, 2001
" Emphasizes source coding
! R.E. Crochiere, L.R. Rabiner: Multirate Digital Signal
Processing, Prentice-Hall 1983
! M. Vetterli, J. Kovacevic: Wavelets and Subband Coding,
Prentice Hall, 1995
! H.S. Malvar: Signal Processing with Lapped Transforms,
Artech House, 1992
! O. Jahromi: Multirate Statistical Signal Processing,
Springer 2007
! Chapters on multirate filtering in dozens of DSP text
books.