Title: Digital Signal Processing II Advanced Topics
1Digital Signal Processing IIAdvanced Topics
- Marc Moonen
- Dept. E.E./ESAT, K.U.Leuven
- marc.moonen_at_esat.kuleuven.be
- www.esat.kuleuven.be/scd/
2Lecture-1 Introduction
- General Intro
- Aims/Scope
- Why study DSP ?
- DSP in applications GSM, ADSL
- Overview
- Activities
- Lectures - Course Notes/Literature
- Homeworks/Exercise sessions
- Project
- Exam
- Review of Discrete-time SignalsSystems
(10-slides)
3Why study DSP ?
- Analog Systems vs. Digital Systems
- - translate analog (e.g. filter) design into
digital - - going digital allows to expand
functionality/flexibility/ - (e.g. how would you do analog speech
recognition ? analog audio compression ? ? )
4DSP in applications GSM
- Cellular mobile telephony (e.g. GSM)
- Basic network architecture
- -country covered by a grid of cells
- -each cell has a base station
- -base station connected to land telephone
network and - communicates with mobiles via a radio
interface - -digital communication format
5DSP in applications GSM
- DSP for digital communications (physical layer
) -
- a common misunderstanding is that digital
communications - is simple.
-
- While in practice
6DSP in applications GSM
- DSP for digital communications (physical layer
) - In practice
- This calls for channel modeling compensation
(equalization)
!!
.59,.41,.76,.05,.37,
Transmitter 1,0,1,1,0,
Multipath Channel
Receiver
??
1,0,1,1,0,
noise
7DSP in applications GSM
- GSM specs/features
- Multi-path channel is modeled with short (35
taps) FIR filter - H(z) ab.z-1c.z-2d.z-3e.z-4
(interpretation?) - Channel is highly time-varying (e.g. terminal
speed 120 km/hr !) - Channel coefficients (cfr. a,b,c,d,e) are
identified in receiver based on - transmission of pre-defined training
sequences, in between data bits - (problem to be solved is given channel
input and channel output, - compute channel coefficients)
- This leads to a least-squares parameter
estimation procedure - (cfr. Linear Algebra course!)
- Channel model is then used to design suitable
equalizer (channel inversion), or (better) for
reconstructing transmitted data bits based on
Maximum-likelihood sequence estimation (Viterbi
decoding) - All this is done at burst-rate (gt100/sec)
SPECTACULAR !!
8DSP in applications GSM
- GSM specs/features (continued)
- - Multiplexing
- Capacity increase by time frequency
multiplexing - FDMA e.g. 125 frequency channels for
GSM/900MHz - TDMA 8 time slots(users) per channel,
burst mode communication - (PS in practice, capacity per cell ltlt
8125 ! ) - - Speech coding
- Original PCM-signal has 64kbits/sec
8 ksamples/sec 8bits/sample - Reduce this to lt11kbits/sec, while
preserving quality - Coding based on speech generation model
(vocal tract,), where - model coefficient are identified for
each new speech segment (e.g. - 20 msec) (i.e. least-squares parameter
estimation, again). - Then transmit model coefficients instead
of signal samples... - - Etc..
BOX FULL OF DSP/MATHEMATICS !!
9DSP in applications ADSL
- Telephone Line Modems
-
- voice-band modems up to 56kbits/sec in 0..4kHz
band -
- ADSL modems up to 8Mbits/sec in 30kHz1MHz band
- (3,55km)
- VDSL modems up to 52Mbits/sec in 12MHz
band - (0.31.5km)
- How has this been made possible?
-
X 1000
10DSP in applications ADSL
- Communication Impairments
- Channel attenuation
- Received signal may be attenuated by more than
60dB - (attenuation increases with line length
larger at high (MHz) frequencies) - PS this is why for a long time, only the
voiceband (up to 4kHz) was used - Frequency-dependent attenuation introduces
inter-symbol interference (ISI). ISI channel
can (again) be modeled with an FIR filter. Number
of taps will be much larger here (gt500!)
11DSP in applications ADSL
- Communication Impairments
- Coupling between wires in same or adjacent
binders introduces crosstalk - Near-end Xtalk (NEXT) (upstream in
downstream, downstream in upstream) - Far-end Xtalk (FEXT) (upstream in
upstream, downstream in downstream) - Meaning that a useful signal may be drowned in
(much larger) signals from other users.. - leading to signal separation and spectrum
management problems - Other
- Radio Frequency Interference
- (AM broadcast, amateur radio)
- Echo due to impedance mismatch
- Etc..
Conclusion Need advanced modulation, DSP,etc. !
12DSP in applications ADSL
- ADSL spectrum divide available transmission
band in 256 narrow bands (tones), transmit
different sub-streams over different sub-channels
(tones) (DMT, Discrete
Multi-tone Modulation)
13DSP in applications ADSL
- ADSL-DMT Transmission block scheme
- DFT/IDFT (FFT/IFFT) based modulation/demodulation
scheme - pointer www.adslforum.com PS do
not try to understand details here...
14DSP in applications ADSL
- ADSL specs
- 512-point (I)FFTs (or similar) for
DMT-modulation - FFT-rate 4.3215 kHz
- (i.e. gt4000 512-point FFTs per second
!!!!) - basic sampling rate is 2.21 MHz
(5124.3215k) - 8.84 MHz A/D or D/A (multi-rate
structure) - fixed HP/LP/BP front-end filtering for frequency
duplex - adjustable time-domain equalization filter (TEQ)
- e.g. 32 taps _at_ 2.21 MHz
- filter initialization via
least-squares/eigenvalue procedure - adaptive frequency-domain equalization filters
(FEQ) - VDSL specs
- e.g. 4096-point (I)FFTs, etc.
-
BOX FULL OF DSP/MATHEMATICS !!
15DSP in applications Other
- Speech
- Speech coding (GSM, DECT, ..), Speech
synthesis (text-to-speech), - Speech recognition
- Audio Signal Processing
- Audio Coding (MP3, AAC, ..), Audio
synthesis - Editing, Automatic transcription,
Dolby/Surround, 3D-audio,. - Image/Video
- Digital Communications
- Wireline (xDSL,Powerline), Wireless (GSM,
3G, Wi-Fi, WiMax - CDMA, MIMO-transmission,..)
-
16DSP in applications
- Enabling Technology is
- Signal Processing
- 1G-SP analog filters
- 2G-SP digital filters, FFTs, etc.
- 3G-SP full of mathematics, linear algebra,
- statistics, etc...
- VLSI
- etc...
SignalsSystems course (JVDW)
DSP-I (PW)
DSP-II
17DSP-II Aims/Scope
- Basic signal processing theory/principles
- filter design, filter banks, optimal
filters adaptive filters - Recent/advanced topics
- robust filter realization, perfect
reconstruction filter banks, - fast adaptive algorithms, ...
- Often birds-eye view
- skip many mathematical details (if
possible ? ) - selection of topics (non-exhaustive)
-
18Overview (I)
- INTRO
- Lecture-1
- Part I Filter Design Implementation
- Lecture-2 IIR FIR Filter Design
- Lecture-3 Filter Realization
- Lecture-4 Filter Implementation
19Overview (II)
- Part II Filter Banks Subband Systems
- Lecture-5 Filter Banks Intro/Applications
(audio coding/CDMA/) - Lecture-6/7 Filter Banks Theory
- Lecture-8 Special Topics
- (Frequency-domain
processing, Wavelets,) - .
20Overview (III)
- Part III Optimal Adaptive Filtering
- Lecture-9 Optimal/Wiener Filters
- Lecture-10 Adaptive Filters/Recursive Least
Squares - Lecture-11 Adaptive Filters/LMS
- Lecture-12 Fast Adaptive Filters
- Lecture-13 Kalman Filters
- .
21Overview (IV)
- OUTRO Lecture 14 - Case study ADSL/VDSL
-
-
22Prerequisites
- Systeemtheorie en Regeltechniek (JVDW)
- Digitale Signaalverwerking I (PW)
- signaaltransformaties, bemonstering,
multi-rate, DFT, - Toegepaste Algebra en
- Analytische Meetkunde (JVDW)
23Literature / Campus Library Arenberg
- A. Oppenheim R. Schafer
- Digital Signal Processing (Prentice Hall
1977) - L. Jackson
- Digital Filters and Signal Processing
(Kluwer 1986) - P.P. Vaidyanathan
- Multirate Systems and Filter Banks
(Prentice Hall 1993) - Simon Haykin
- Adaptive Filter Theory (Prentice Hall
1996) - M. Bellanger
- Digital Processing of Signals (Kluwer
1986) - etc...
Part-I
Part-II
Part-III
24Literature / DSP-II Library
- Collection of books is available to support
course material - List/info/reservation via DSP-II webpage
- contact Vincent.LeNir_at_esat (E)
25Activities Lectures
- Lectures 14 2 hrs
- Course Material
- Part I-II-III Slides (use version 2008-2009
!!) - ...download from DSP-II webpage
- Part III Introduction to Adaptive Signal
Processing, - Marc Moonen Ian.K. Proudler
- support material, not mandatory !
- (if needed) download from DSP-II webpage
26Activities Homeworks/Ex. Sessions
- Homeworks
- to support course material
- 6 Matlab/Simulink Sessions
- to support homeworks
- come prepared !
- contact Sylwester.Szczepaniak_at_esat
(EnglishPolish) - Vincent.LeNir_at_esat
(EnglishFrench) - Prabin.Kumarpandey_at_esat
(EnglishNepali) - Pepe.Gilcacho_at_esat
(EnglishSpanish)
27Activities Project
- Discover DSP technology in present-day systems
- examples 3D-audio, music synthesis,
automatic - transcription, speech
codec, MP3, GSM, ADSL, - Select topic/paper from list on DSP II webpage
(submit 1st/2nd choice by Oct.5 to
sylwester.szczepaniak_at_esat) - Study www surfing
- Build demonstration model experiment in
Matlab/Simulink - Deliverable
- Intermediate presentation (.ppt or similar)
Oct.. - Final presentation, incl. Matlab/Simulink
demonstration Dec.. - (20 mins per group)
- Software
- Groups of 2
28Activities Project
- Topics/Papers
- List available under DSP-II web page
- Other topics subject to approval !
- (email 1/2-page description to
sylwester.szczepaniak_at_esat - before Oct. 5)
- Tutoring
- 14 research assistants/postdocs
- All PPT presentations will be made available,
for ref.
29Activities Exam
- Oral exam, with preparation time
- Open book
- Grading
- 5 pts for question-1
- 5 pts for question-2
- 5 pts for question-3
- 5 pts for project (software/presentation)
- ___
- 20 pts
30homes.esat.kuleuven.be/sszczepa/dspII
- Contact sylwester.szczepaniak_at_esat
- Slides
- Homeworks
- Projects info/schedule
- Exams 2000-2001, ..
- DSP-II Library
- FAQs (send questions to
- sylwester.szczepaniak_at_esat
- or marc.moonen_at_esat )
31Review of discrete-time systems 1/10
- Discrete-time (DT) system is sampled data
system - Input signal uk is a sequence of samples
(numbers) -
..,u-2,u-1,u0,u1,u2, - System then produces a sequence of output
samples yk -
..,y-2,y-1,y0,y1,y2, - Will consider linear time-invariant (LTI) DT
systems - Linear
- input u1k -gt output y1k
- input u2k -gt output y2k
- hence a.u1kb.u2k-gt a.y1kb.y2k
- Time-invariant (shift-invariant)
- input uk -gt output yk, hence input
uk-T -gt output yk-T
32Review of discrete-time systems 2/10
- Causal systems
- iff for all input signals with uk0,klt0 -gt
output yk0,klt0 - Impulse response
- input ,0,0,1,0,0,0,...-gt output
,0,0,h0,h1,h2,h3,... - General input u0,u1,u2,u3 (cfr.
linearity shift-invariance!) -
-
33Review of discrete-time systems 3/10
convolution sum
34Review of discrete-time systems 4/10
H(z) is transfer function
35Review of discrete-time systems 5/10
- Z-Transform
- input-output relation
- may be viewed as shorthand notation
- (for convolution operation/Toeplitz-vector
product) - stability
- bounded input uk -gt bounded output yk
- --iff
-
- --iff poles of H(z) inside the unit circle
- (for causal,rational systems)
-
36Review of discrete-time systems 6/10
- Example-1 Delay operator
- Impulse response is ,0,0,0, 1,0,0,0,
- Transfer function is
- Example-2 Delay feedback
- Impulse response is ,0,0,0, 1,a,a2,a3
- Transfer function is
37Review of discrete-time systems 7/10
- Will consider only rational transfer functions
-
-
- In general, these represent infinitely long
impulse response (IIR) systems - N poles (zeros of A(z)) , N zeros (zeros of B(z))
- corresponds to difference equation
- Hence rational H(z) can be realized with finite
number of delay elements, multipliers and adders
38Review of discrete-time systems 8/10
- Special case is
- N poles at the origin z0 (hence guaranteed
stability) - N zeros (zeros of B(z)) all zero
filters - corresponds to difference equation
-
moving average (MA) filters - impulse response is
-
finite impulse response (FIR) filters
39Review of discrete-time systems 9/10
- H(z) frequency response
- given a system H(z)
- given an input signal complex sinusoid
- output signal
-
-
- frequency
response - H(z) evaluated on
the unit circle
40Review of discrete-time systems 10/10
- H(z) frequency response
- periodic period
- for a real impulse response hk
- Magnitude response is
even function - Phase response
is odd function - example (low pass filter)
Nyquist frequency