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Digital Signal Processing II
`Advanced Topics’
Marc MoonenDept. E.E./ESAT, K.U.Leuven
[email protected]/scd/
DSP-II Version 2009-2010 Lecture-1 Introduction p. 2
Lecture-1 : Introduction
• Aims/Scope Why study DSP ? DSP in applications : GSM, ADSL,…
• Overview
• ActivitiesLectures - Course Notes/LiteratureHomeworks/Exercise sessionsProjectExam
DSP-II Version 2009-2010 Lecture-1 Introduction p. 3
Why 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 ? …? )
IN OUT IN OUTA/D D/A
2+2
=4
DSP-II Version 2009-2010 Lecture-1 Introduction p. 4
DSP 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
DSP-II Version 2009-2010 Lecture-1 Introduction p. 5
DSP in applications : GSM
• DSP for digital communications (`physical layer’ ) :
– a common misunderstanding is that digital communications is `simple’….
– While in practice…
Transmitter1,0,1,1,0,…
Channel
x +
a noise1/a
x
Receiver
deci
sion
.99,.01,.96,.95,.07,…
1,0,1,1,0,…
DSP-II Version 2009-2010 Lecture-1 Introduction p. 6
DSP in applications : GSM
• DSP for digital communications (`physical layer’ ) :
– In practice…
– This calls for channel modeling + compensation (equalization)
Transmitter1,0,1,1,0,…
+
Receiver
1,0,1,1,0,…??noise
`Multipath’Channel
.59,.41,.76,.05,.37,… !!
DSP-II Version 2009-2010 Lecture-1 Introduction p. 7
DSP in applications : GSM
• GSM Channel Estimation/Compensation– Multi-path channel is modeled with short (3…5 taps) FIR filter
H(z)= a+b.z^-1+c.z^-2+d.z-3+e.z^-4 (interpretation?)
– 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 at receiver is: `given channel input and channel output, compute channel coefficients’).
This leads to a least-squares parameter estimation procedure
(see 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’).– Channel is highly time-varying (e.g. terminal speed 120 km/hr !)
=> All this is done at `burst-rate’ (+- 100 times per sec).
= SPECTACULAR !!
DSP-II Version 2009-2010 Lecture-1 Introduction p. 8
DSP in applications : GSM
• GSM Channel Estimation/Compensation
• GSM Speech Coding– Original `PCM’-signal has 64kbits/sec = 8 ksamples/sec * 8bits/sample.
– How to reduce this to <11kbits/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).
– This leads to a least-squares parameter estimation (again), executed +- 50times per second). Fast algorithm is used, e.g. `Levinson-Durbin’ algorithm
(see (Advanced) Linear Algebra course).
– Then transmit model coefficients instead of signal samples.– Synthesize speech segment at receiver
(that `sounds like’ original speech segment).
= SPECTACULAR !!
DSP-II Version 2009-2010 Lecture-1 Introduction p. 9
DSP in applications : GSM
• GSM Channel Estimation/Compensation
• GSM Speech Coding
• GSM Multiple Access Schemes– 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 << 8*125 ! )
• Etc..
= BOX FULL OF DSP/MATHEMATICS !!
DSP-II Version 2009-2010 Lecture-1 Introduction p. 10
DSP in applications : ADSL
Telephone Line Modems
– voice-band modems : up to 56kbits/sec in 0..4kHz b and
– ADSL modems : up to 8Mbits/sec in 30kHz…1MHz band
(3,5…5km)– VDSL modems : up to 52Mbits/sec in …12MHz ban d
(0.3…1.5km)
How has this been made possible?
X 1000
DSP-II Version 2009-2010 Lecture-1 Introduction p. 11
DSP 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 (>500!)
DSP-II Version 2009-2010 Lecture-1 Introduction p. 12
DSP 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. !
DSP-II Version 2009-2010 Lecture-1 Introduction p. 13
DSP 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’)
DSP-II Version 2009-2010 Lecture-1 Introduction p. 14
DSP 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...
DSP-II Version 2009-2010 Lecture-1 Introduction p. 15
DSP in applications : ADSL
ADSL specs• 512-point (I)FFT’s (or `similar’) for DMT-modulation
FFT-rate = 4.3215 kHz (i.e. >4000 512-point FFTs per second !!!!)
• basic sampling rate is 2.21 MHz (=512*4.3215k)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 @ 2.21 MHzfilter initialization via least-squares/eigenvalue procedure
• adaptive frequency-domain equalization filters (FEQ)
VDSL specs • e.g. 4096-point (I)FFT’s, etc….
= BOX FULL OF DSP/MATHEMATICS !!
DSP-II Version 2009-2010 Lecture-1 Introduction p. 16
DSP in applications : Other…
• Speech Speech coding (GSM, DECT, ..), Speech synthesis (text-to-speech), Speech recognition
• Audio Signal Processing Audio Coding (MP3, AAC, ..), Audio synthesisEditing, Automatic transcription, Dolby/Surround, 3D-audio,.
• Image/Video• Digital Communications
Wireline (xDSL,Powerline), Wireless (GSM, 3G, Wi-Fi, WiMaxCDMA, MIMO-transmission,..)
• …
DSP-II Version 2009-2010 Lecture-1 Introduction p. 17
DSP in applications
Enabling Technology is• Signal Processing
1G-SP: analog filters2G-SP: digital filters, FFT’s, etc.
3G-SP: full of mathematics, linear algebra, statistics, etc...
• VLSI• etc...
Signals&Systems course (JVDW)
DSP-I (PW)
DSP-II
DSP-II Version 2009-2010 Lecture-1 Introduction p. 18
DSP-II Aims/Scope
• Basic signal processing theory/principlesfilter design, filter banks, optimal filters & adaptive filters
• Recent/advanced topicsrobust filter realization, perfect reconstruction filter banks, fast adaptive algorithms, ...
• Often `bird’s-eye view’skip many mathematical details (if possible… ☺ )selection of topics (non-exhaustive)
DSP-II Version 2009-2010 Lecture-1 Introduction p. 19
0 0 .5 1 1 .5 2 2 .5 30
0 .2
0 .4
0 .6
0 .8
1
1 .2
P assb and R ip p le
S top band R ipp le
P assb and C uto ff -> < - S topb and C uto ff
Overview (I)
• INTRO : Lecture-1Lecture-2 : Signals and Systems Review
• Part I : Filter Design & ImplementationLecture-3 : IIR & FIR Filter DesignLecture-4 : Filter RealizationLecture-5 : Filter Implementation
DSP-II Version 2009-2010 Lecture-1 Introduction p. 20
Overview (II)
• Part II : Filter Banks & Subband SystemsLecture-6 : Filter Banks Intro/Applications (audio coding/CDMA/…)Lecture-7/8 : Filter Banks Theory Lecture-9 : Special Topics
(Frequency-domain processing, Wavelets,…)
.
3 subband processing 3H1(z) G1(z)
3 subband processing 3H2(z) G2(z)
3 subband processing 3H3(z) G3(z)
3 subband processing 3H4(z) G4(z)
+IN OUT
DSP-II Version 2009-2010 Lecture-1 Introduction p. 21
Overview (III)
• Part III : Optimal & Adaptive FilteringLecture-10 : Optimal/Wiener FiltersLecture-11: Adaptive Filters/Recursive Least SquaresLecture-12: Adaptive Filters/LMSLecture-13: `Fast’ Adaptive FiltersLecture-14: Kalman Filters
.
DSP-II Version 2009-2010 Lecture-1 Introduction p. 22
Prerequisites
`Systeemtheorie en Regeltechniek’ (JVDW)
`Digitale Signaalverwerking I’ (PW)signaaltransformaties, bemonstering, multi-rate, DFT, …
`Toegepaste Algebra en Analytische Meetkunde’ (JVDW)
DSP-II Version 2009-2010 Lecture-1 Introduction p. 23
Literature / 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-III
Part-II
Part-I
DSP-II Version 2009-2010 Lecture-1 Introduction p. 24
Literature / DSP-II Library
• Collection of books is available to support course material
• List/info/reservation via DSP-II webpage
• contact: beier.li@esat (E/C)
DSP-II Version 2009-2010 Lecture-1 Introduction p. 25
Activities : Lectures
Lectures : 14 * 2 hrs
Course Material :• Part I-II-III : Slides (use version 2009-2010 !!)
...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
DSP-II Version 2009-2010 Lecture-1 Introduction p. 26
Activities : Homeworks/Ex. Sessions
• `Homeworks’…to support course material
• 6 Matlab/Simulink Sessions…to support homeworks…come prepared !
• contact: amir.forouzan@esat (English+Persian)
beier.li@esat (English+Chinese)
prabin.kumarpandey@esat (English+Nepali)
pepe.gilcacho@esat (English+Spanish)
DSP-II Version 2009-2010 Lecture-1 Introduction p. 27
Activities : Project
• Discover DSP technology in present-day systemsexamples: 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 pepe.gilcacho@esat) • Study & www surfing• Build demonstration model & experiment in Matlab/Simulink
• Deliverable : – Project Plan : 1 page status report & time plan
(send to marc.moonen@esat by Nov 1st) – Presentation: .ppt or similar, incl. Matlab/Simulink demonstration
(December, 20 mins per group)– Software
• Groups of 2
DSP-II Version 2009-2010 Lecture-1 Introduction p. 28
Activities : Project
Topics/Papers• List available under DSP-II web page• Other topics : subject to approval !
(email 1/2-page description to pepe.gilcacho@esatbefore Oct. 5)
Tutoring10 research assistants/postdocs
All PPT presentations will be made available, for r ef.
DSP-II Version 2009-2010 Lecture-1 Introduction p. 29
Activities : Exam
• Oral exam, with preparation time• Open book• Grading :
5 pts for question-15 pts for question-25 pts for question-3
5 pts for project (software/presentation)___
= 20 pts
DSP-II Version 2009-2010 Lecture-1 Introduction p. 30
homes.esat.kuleuven.be/~pepe/dspII
• Contact: pepe.gilcacho@esat• Slides• Homeworks• Projects info/schedule• Exams• DSP-II Library• FAQs (send questions to
pepe.gilcacho@esator marc.moonen@esat )