1
INTRODUCTION
1.1SignalsandInformation
1.2SignalProcessingMethods
1.3ApplicationsofDigitalSignalProcessing
1.4SamplingandAnalog to DigitalConversion
ignal processing is concerned with the modelling, detection,
identificationandutilisationofpatternsandstructuresinasignal
[Link]-
fi,digitalTVandradio,cellularmobilephones,voicerecognition,vision,
radar,sonar,geophysicalexploration,medicalelectronics,andingener al
anysystemthatisconcernedwiththecommunicationorprocessingof
information. Signal processing theory plays a central role in the
developmentofdigitaltelecommunicationandautomationsystems,andin
efficientandoptimaltransmission,receptionanddecodingofinformati on.
Statisticalsignalprocessingtheoryprovidesthefoundationsformodel ling
thedistributionofrandomsignalsandtheenvironmentsinwhichthesignal s
propagate. Statistical models are applied in signal processing, and in
decision-makingsystems,forextractinginformationfromasigna lthatmay
benoisy,[Link] of
signals,andabriefintroductiontovarioussignalprocessingmethodologie s.
Weconsiderseveralkeyapplicationsofdigitalsignalprocessing inadaptive
noise reduction, channel equalisation, pattern classification/recognition,
audiosignalcoding,signaldetection,spatialprocessingfordirectiona l
receptionofsignals,Dolbynoisereductionandradar. The chapter conclude s
withanintroductiontosamplingandconversionofcontinuous-timesignals
to digital signals.
S
H HH H E EE E LL LL LL LL O OO O
2 Introduction
1.1SignalsandInformation
Asignalcanbedefinedasthevariationofaquantitybywhichinform ation
isconveyedregardingthestate,thecharacteristics,thecomposi tion,the
trajectory,thecourseofactionortheintentionofthesignalsource .Asignal
isameanstoconveyinformation. Theinformationconveyedin a signal may
beusedbyhumansormachinesforcommunication,forecasting,decision-
making,control,explorationetc.Figure1.1illustratesaninformations ource
followedbyasystemforsignallingtheinformation,acommunication
channelforpropagationofthesignalfromthetransmittertothere ceiver,
andasignalprocessingunitatthereceiverforextractionofthe information
[Link],thereisamappingoperationthatmaps the
information I(t)tothesignal x(t)thatcarriestheinformation,thismapping
function may be denoted as T[]and expressed as
)] ( [ ) ( t I T t x = (1.1)
For example, in human speech communication, the voi ce-generating
mechanismprovidesameansforthetalkertomapea chwordintoadistinct
acousticspeechsignalthatcanpropagatetotheli [Link]
word w,thetalkergeneratesanacousticsignalrealisati onoftheword;this
acousticsignal x(t) maybecontaminatedbyambientnoiseand/ordistort ed
byacommunicationchannel,orimpairedbythespea kingabnormalitiesof
thetalker,andreceivedasthenoisyanddistorted signal y(t).Inadditionto
conveyingthespokenword,theacousticspeechsign alhasthecapacityto
conveyinformationonthespeakingcharacteristic, accentandtheemotional
[Link] formationbyprocessingthe
signaly(t).
Inthepastfewdecades,thetheoryandapplication sofdigitalsignal
processinghaveevolvedtoplayacentralroleint hedevelopmentofmodern
telecommunication and information technology system s.
Signalprocessingmethods are central to efficient communication, and to
thedevelopmentofintelligentman/machineinterfac esinsuchareasas
Information
source
Informationto
signalmapping
Signal
DigitalSignal
Processor
Channel
Noise
Noisy
signal
Signal&
Information
Figure1.1 Illustrationofacommunicationandsignalprocessingsystem.
SignalProcessingMethods 3
speechandvisualpatternrecognitionformultimedi [Link],
digitalsignalprocessingisconcernedwithtwobro adareasofinformation
theory:
(a) efficientandreliablecoding,transmission,re ception,storageand
representation of signals in communication systems, and
(b) the extraction of information from noisy signal s for pattern
recognition, detection, forecasting, decision-makin g, signal
enhancement, control, automation etc.
Inthenextsectionweconsiderfourbroadapproach estosignalprocessing
problems.
1.2SignalProcessingMethods
Signalprocessingmethodshaveevolvedinalgorithm iccomplexityaiming
foroptimalutilisationoftheinformationinorder toachievethebest
[Link] ntofsignalprocessing
methodsincreases,oftenexponentially,withtheal gorithmiccomplexity.
However,theimplementationcostofadvancedsignal processingmethods
hasbeenoffsetandmadeaffordablebytheconsiste nttrendinrecentyears
ofacontinuingincreaseintheperformance,couple dwithasimultaneous
decrease in the cost, of signal processing hardware .
Dependingonthemethodused,digitalsignalproces singalgorithmscan
becategorisedintooneoracombinationoffourbr [Link]
non parametricsignalprocessing,model-basedsignalpr ocessing,Bayesian
statisticalsignalprocessingandneuralnetworks. Thesemethodsarebriefly
described in the following.
1.2.1Non parametricSignalProcessing
Non parametricmethods,asthenameimplies,do not utiliseaparametric
modelofthesignal generation or a model of the st atistical distribution of the
[Link] sequenceofdigits.
Non parametric methods are not specialised to any parti cular class of
signals,theyarebroadlyapplicablemethodsthatc anbeappliedtoany
signalregardlessofthecharacteristicsorthesou [Link]
drawback of these methods is that they do not utili se the distinct
characteristics of the signal process that may lead to substantial
4 Introduction
[Link] parametricmethods
includedigitalfilteringandtransform-basedsigna lprocessingmethodssuch
astheFourieranalysis/synthesisrelationsandthe discretecosinetransform.
Somenon parametricmethodsofpowerspectrumestimation,in terpolation
and signal restoration are described in Chapters 9, 10 and 11.
1.2.2Model-BasedSignalProcessing
Model-basedsignalprocessingmethodsutiliseapar ametricmodelofthe
[Link] mallydescribesthe
predictablestructuresandtheexpectedpatternsin thesignalprocess,and
canbeusedtoforecastthefuturevaluesofasign alfromitspasttrajectory.
Model-basedmethodsnormallyoutperformnon parametricmethods,since
theyutilisemoreinformationintheformofamode lofthesignalprocess.
However,theycanbesensitivetothedeviationsof a signal from the class of
[Link] usedparametricmodel
isthelinearpredictionmodel,describedinChapte [Link]
modelshavefacilitatedthedevelopmentofadvanced signalprocessing
methodsforawiderangeofapplicationssuchaslo w bit ratespeechcoding
incellularmobiletelephony,digitalvideocoding, high resolutionspectral
analysis, radar signal processing and speech recogn ition.
1.2.3BayesianStatisticalSignalProcessing
Thefluctuationsofapurelyrandomsignal,orthe distributionofaclassof
randomsignalsinthesignalspace,cannotbemodel ledbyapredictive
equation,butcanbedescribedintermsofthestat isticalaveragevalues,and
modelledbyaprobabilitydistributionfunctionin amultidimensionalsignal
[Link],asdescribedinChapter8,ali nearpredictionmodel
drivenbyarandomsignalcanmodeltheacousticre alisationofaspoken
[Link],therandominputsignalofthelinea rpredictionmodel,or
thevariationsinthecharacteristicsofdifferent acousticrealisationsofthe
same word across the speaking population, can only be described in
statisticaltermsandintermsofprobabilityfunct [Link]
theoryprovidesageneralisedframeworkfor statist ical processing of random
signals,andforformulatingandsolvingestimation anddecision-making
problems.Chapter4describestheBayesianinferenc emethodologyandthe
estimation of random processes observed in noise.
Applications of Digital Signal
Processing
5
1.2.4NeuralNetworks
Neuralnetworksarecombinationsofrelativelysimp lenon-linearadaptive
processing units, arranged to have a structural res emblance to the
transmissionandprocessingofsignalsinbiologica [Link]
networkseverallayersofparallelprocessingeleme ntsareinterconnected
withahierarchicallystructuredconnectionnetwork .Theconnectionweights
aretrainedtoperformasignalprocessingfunction suchaspredictionor
classification. Neural networks are particularly us eful in non-linear
partitioningofasignalspace,infeatureextracti onandpatternrecognition,
[Link] ernrecognitionsystems
neuralnetworksareusedtocomplementBayesianinf [Link]
themainobjectiveofthisbookistoprovideacoh erentpresentationofthe
theoryandapplicationsofstatisticalsignalproce ssing,neuralnetworksare
not discussed in this book.
1.3ApplicationsofDigitalSignalProcessing
Inrecentyears,thedevelopmentandcommercialava ilabilityofincreasingly
powerfulandaffordabledigitalcomputershasbeen accompaniedbythe
developmentofadvanceddigitalsignalprocessinga lgorithmsforawide
varietyofapplicationssuchasnoisereduction,te lecommunication,radar,
sonar,videoandaudiosignalprocessing,patternr ecognition, geophysics
explorations,dataforecasting,andtheprocessing oflargedatabasesforthe
identificationextractionandorganisationofunkno wnunderlyingstructures
and patterns. Figure 1.2 shows a broad categorisati on of some DSP
[Link] eralkeyapplicationsof
digital signal processing methods.
1.3.1AdaptiveNoiseCancellationandNoiseReducti on
Inspeechcommunicationfromanoisyacousticenvir onmentsuchasa
movingcarortrain,oroveranoisytelephonechan nel,thespeechsignalis
[Link] surementsystemsthe
information-bearing signal is often contaminated by noise from its
surrounding environment. The noisy observation y(m) can be modelled as
6 Introduction
y(m) = x(m) + n(m) (1.2)
where x(m) and n(m)arethesignalandthenoise,and misthediscrete-
[Link],forexamplewhenus ingamobiletelephone
inamovingcar,orwhenusingaradiocommunicatio ndeviceinanaircraft
cockpit,itmaybepossibletomeasureandestimate theinstantaneous
amplitudeoftheambientnoiseusingadirectional [Link]
x(m) maythenberecoveredbysubtractionofanestimat eofthenoisefrom
the noisy signal.
Figure 1.3 shows a two-input adaptive noise cancell ation system for
[Link] tionalmicrophonetakes
DSP Applications
Information Transmission/Storage/Retrieval
Informationextraction
Signal Classification
Speech recognition, image
and character recognition,
signaldetection
Parameter Estimation
Spectral analysis,radar
andsonarsignal processing,
signal enhancement,
geophysics exploration
ChannelEqualisation
Source/ChannelCoding
Speech coding, image coding,
datacompression,communication
overnoisychannels
Signal and data
communicationon
adversechannels
Figure1.2 Aclassificationoftheapplicationsofdigitalsignalprocessing.
y(m) = x (m)+n(m)
n(m+)
x(m)
^
n(m)
^
z z
. . .
Noise Estimation Filter
Noisy signal
Noise
Noise estimate
Signal
Adaptation
algorithm
z
1
w
2
w
1
w
0 w
P-1
1 1
Figure1.3 Configurationofatwo-microphoneadaptivenoisecanceller.
ApplicationsofDigitalSignalProcessing 7
asinputthenoisysignal x(m) + n(m) ,andaseconddirectionalmicrophone,
positioned some distance away, measures the noise n(m+ ). The
attenuationfactor andthetimedelay providearatherover-simplified
modeloftheeffectsofpropagationofthenoiseto differentpositionsinthe
space where the microphones are placed. The noise f rom the second
microphoneisprocessedbyanadaptivedigitalfilt ertomakeitequaltothe
noisecontaminatingthespeechsignal,andthensub tractedfromthenoisy
[Link] cancellerismoreeffective
incancellingoutthelow-frequencypartofthenoi se,butgenerallysuffers
from the non-stationary character of the signals, a nd from the over-
simplified assumption that a linear filter canmode lthediffusionand
propagation of the noise sound in the space.
In many applications, for example at the receiver o f a
telecommunicationsystem,thereis noaccesstotheinstantaneousvalueof
thecontaminatingnoise,andonlythenoisysignal [Link]
thenoisecannotbecancelledout,butitmaybere duced,inanaverage
sense,usingthestatisticsofthesignalandthen oiseprocess.Figure1.4
showsabankofWienerfiltersforreducingadditiv enoisewhenonlythe
.
.
.
y(0)
y(1)
y(2)
y(N-1)
Noisy signal
y(m)=x(m)+n(m)
x(0)
x(1)
x(2)
x(N-1)
^
^
^
^
I
n
v
e
r
s
e
D
i
s
c
r
e
t
e
F
o
u
r
i
e
r
T
r
a
n
s
f
o
r
m
.
.
.
Y(0)
Y(1)
Y(2)
Y(N-1)
D
i
s
c
r
e
t
e
F
o
u
r
i
e
r
T
r
a
n
s
f
o
r
m
X(0)
X(1)
X(2)
X(N-1)
^
^
^
^
WN -1
W
0
W
2
Signal and noise
powerspectra
Restored signal
Wienerfilter
estimator
W
1
.
.
.
.
.
.
Figure1.4 AfrequencydomainWienerfilterforreducingadditivenoise.
8 Introduction
[Link] entsattenuateeachnoisy
signalfrequencyininverseproportiontothesigna ltonoiseratioatthat
[Link],der ivedinChapter6,are
calculatedfromestimatesofthepowerspectraoft hesignalandthenoise
processes.
1.3.2BlindChannelEqualisation
Channelequalisationistherecoveryofasignaldi stortedintransmission
throughacommunicationchannelwithanon-flatmag nitudeoranon-linear
[Link] ntheprocessof
[Link] equalisationhasawide
range of applications, for example in digital telec ommunications for
removalofinter-symbolinterferenceduetonon-ide alchannelandmulti-
pathpropagation,inspeechrecognitionforremoval oftheeffectsofthe
microphonesandthecommunicationchannels,incorr ectionofdistorted
images,analysisofseismicdata,de-reverberation ofacousticgramophone
recordings etc.
Inpractice,blindequalisationisfeasibleonlyif someuseful statistics of
[Link] lindequalisationmethod
dependsonhowmuchisknownaboutthecharacterist icsoftheinputsignal
andhowusefulthisknowledgecanbeinthechannel identificationand
equalisationprocess.Figure1.5illustratestheco nfigurationofadecision-
[Link] composedoftwodistinct
sections:anadaptiveequaliserthatremovesalarg epartofthechannel
distortion, followed by a non-linear decision devic e for an improved
[Link] cisiondeviceisthefinal
Channel noise
n(m)
x(m)
Channel distortion
H(f)
f
y(m)
x(m)
^
Error signal
-
+
Adaptation
algorithm
+
f
Equaliser
Blind decision-directed equaliser
H
inv
(f)
Decision device
+
Figure1.5 Configurationofadecision-directedblindchannelequaliser.
ApplicationsofDigitalSignalProcessing 9
estimate of the channel input, and it is used as th e desired signal to direct the
[Link] coveredindetailin
Chapter 15.
1.3.3SignalClassificationandPatternRecognition
Signalclassificationisusedindetection,pattern recognitionanddecision-
[Link],asimplebinary-state classifiercanactasthe
detectorofthepresence,ortheabsence,ofaknow [Link]
signalclassification,theaimistodesignaminim um-errorsystemfor
labellinga signal with one of a number of likely classes of signal.
Todesignaclassifier;asetofmodelsaretrained fortheclassesof
[Link] esimplestformthatthe
models can assume is a bank, or code book, of wavef orms, each
representingtheprototypeforoneclassofsignals .Amorecompletemodel
foreachclassofsignalstakestheformofaproba bilitydistributionfunction.
Intheclassificationphase,asignalislabelledw iththenearestorthemost
[Link],incommunicationofabi narybitstreamovera
band-passchannel,thebinaryphaseshiftkeying(B PSK)schemesignals
thebit1usingthewaveform A
c
sin
c
t andthebit0using A
c
sin
c
t .
Atthereceiver,thedecoderhasthetaskofclassi fyingandlabellingthe
receivednoisysignalasa1ora0.Figure1.6 illustratesacorrelation
[Link] hastwocorrelators,
eachprogrammedwithoneofthetwo symbols represe nting the binary states
Receivednoisysymbol
Correlator for symbol "1"
Correlator forsymbol "0"
Corel(1)
Corel(0)
"
1
"
i
f
C
o
r
e
l
(
1
)
C
o
r
e
l
(
0
)
"
0
"
i
f
C
o
r
e
l
(
1
)
<
C
o
r
e
l
(
0
)
"1"
Decision
device
Figure1.6 Ablockdiagramillustrationoftheclassifierinabinaryphase-shiftkeying
demodulation.
10 Introduction
[Link] atestheunlabelledinput
signalwitheachofthetwocandidatesymbolsands electsthecandidatethat
has a higher correlation with the input.
Figure1.7illustratestheuseofaclassifierina limitedvocabulary,
[Link] reare Vwordsinthe
[Link],onma nydifferentexamples
ofthespokenword,tocapturetheaveragecharacte risticsandthestatistical
[Link] oabankof V+1models,
oneforeachwordinthevocabularyandanaddition almodelforthesilence
[Link],thetask istodecodeandlabelan
M
ML
.
.
.
Speech
signal
Feature
sequence
Y
f
Y|
M
(Y|M
1
)
Wordmodel M
2
likelihood
of M
2
M
o
s
t
l
i
k
e
l
y
w
o
r
d
s
e
l
e
c
t
o
r
Feature
extractor
Wordmodel M
V
Wordmodel M
1
f
Y|
M
(Y|M
2
)
f
Y|
M
(Y|M
V
)
likelihood
of M
1
likelihood
of M
v
Silencemodel
M
sil
f
Y|
M
(Y|M
sil
)
likelihood
of M
sil
Figure1.7 Configurationofspeechrecognitionsystem,f(Y|M
i
)isthelikelihoodof
themodelM
i
givenanobservationsequenceY.
ApplicationsofDigitalSignalProcessing 11
acousticspeechfeaturesequence,representinganu nlabelledspokenword,
asoneofthe [Link] he
classifiercalculatesaprobabilityscoreandselec tsthewordwiththehighest
score.
1.3.4LinearPredictionModellingofSpeech
Linearpredictivemodelsarewidelyusedinspeech processingapplications
such as lowbitrate speech coding in cellular tele phony, speech
[Link] atedbyinhalingair
intothelungs,andthenexhalingitthroughthevi bratingglottiscordsand
[Link],noise-like,airflowf romthelungsisspectrally
shapedandamplifiedbythevibrations of the glott al cords and the resonance
[Link] theglottalcordsandthe
vocaltractistointroduceameasureofcorrelatio nandpredictabilityonthe
[Link] 1.8illustratesamodel
[Link] ndemitsarandom
excitationsignalwhichisfiltered,firstbyapit chfiltermodeloftheglottal
cords and then by a model of the vocal tract.
Themainsourceofcorrelationinspeechisthevo caltractmodelledbya
[Link] amplitudeofthesignalat
time m, x(m) , using a linear combination of P previous samples
x(m1),L, x(m P) [ ]as
=
=
P
k
k
k m x a m x
1
) ( ) ( (1.3)
where
x(m) is the prediction of the signal x(m) , and the vector
] , , [
1
T
P
a a K = a isthecoefficientsvectorofapredictoroforder [Link]
Excitation
Speech
Random
source
Glottal (pitch)
model
P(z)
Vocal tract
model
H(z)
Pitch period
Figure1.8 Linearpredictivemodelofspeech.
12 Introduction
predictionerror e(m),[Link] x(m)
and its predicted value
x(m) , is defined as
e(m) = x(m) a
k
x(m k)
k=1
P
(1.4)
Thepredictionerror e(m)mayalsobeinterpretedastherandomexcitation
ortheso-calledinnovationcontentof x(m) .FromEquation(1.4)asignal
generated by a linear predictor can be synthesised as
x(m) = a
k
x(m k) + e(m)
k=1
P
(1.5)
Equation (1.5) describes a speech synthesis model i llustrated in Figure 1.9.
1.3.5DigitalCodingofAudioSignals
Indigitalaudio,thememoryrequiredtorecordas ignal,thebandwidth
requiredforsignaltransmissionandthesignalto quantisationnoiseratio
arealldirectlyproportionaltothenumberofbits [Link]
inthedesignofacoderistoachievehighfidelit ywithasfewbitsper
sampleaspossible,atanaffordableimplementation [Link]
codingschemesutilisethestatisticalstructureso fthesignal,andamodelof
thesignalgeneration,togetherwithinformationon thepsychoacousticsand
[Link],therea retwomaincategoriesof
audiocoders:model-basedcoders,usedforlowbit ratespeechcodingin
z
1
z
1
z
1 . . .
u(m)
x(m-1) x(m-2) x(mP)
a
a
2
a
1
x(m)
G
e(m)
P
Figure1.9 Illustrationofasignalgeneratedbyanall-pole,linearprediction
model.
ApplicationsofDigitalSignalProcessing 13
applicationssuchascellulartelephony;andtransf orm-basedcodersusedin
highquality coding of speech and digital hi-fi aud io.
Figure1.10showsasimplifiedblockdiagramconfig urationofaspeech
codersynthesiserofthetypeusedindigitalcellu [Link]
signalismodelledastheoutputofafilterexcite [Link]
randomexcitationmodelstheairexhaledthroughth elung,andthefilter
modelsthevibrationsoftheglottalcordsandthe [Link]
transmitter,speechissegmentedintoblocksofabo ut30mslongduring
whichspeechparameterscanbeassumedtobestatio [Link]
speechsamplesisanalysedtoextractandtransmit asetofexcitationand
filterparametersthatcanbeusedto synthesis the speech. At the receiver, the
model parameters and the excitation are used to rec onstruct the speech.
A transform-based coder is shown in Figure 1.11. T he aim of
transformationistoconvertthesignalintoaform whereitlendsitselftoa
moreconvenientandusefulinterpretationandmanip ulation.InFigure1.11
theinputsignalistransformedtothefrequencydo mainusingafilterbank,
oradiscreteFouriertransform,oradiscretecosi [Link]
advantages of coding a signal in the frequency doma in are:
(a)Thefrequencyspectrumofasignalhasarelati velywelldefined
structure, for example most of the signal power is usually
concentrated in the lower regions of the spectrum.
Synthesiser
coefficients
Excitation e(m)
Speech x(m)
Scalar
quantiser
Vector
quantiser
Model-based
speech analysis
(a) Source coder
(b) Source decoder
Pitch and vocal-tract
coefficients
Excitationaddress
Excitation
codebook
Pitchfilter
Vocal-tract filter
Reconstructed
speech
Pitchcoefficients
Vocal-tract coefficients
Excitation
address
Figure1.10 Blockdiagramconfigurationofamodel-basedspeechcoder.
14 Introduction
(b) Arelativelylowamplitudefrequencywouldbemaske dinthenear
vicinityofalargeamplitudefrequencyandcan the refore be coarsely
encoded without any audible degradation.
(c) The frequency samples are orthogonal and can be cod ed
independently with different precisions.
Thenumberofbitsassignedtoeachfrequencyofa signalisavariable
thatreflectsthecontributionofthatfrequencyto thereproductionofa
[Link] der,theallocationofbits
todifferentfrequencies is made to vary with the t ime variations of the power
spectrum of the signal.
1.3.6DetectionofSignalsinNoise
Inthedetectionofsignalsinnoise,theaimisto determine if the observation
consistsofnoisealone,orifitcontainsasignal .Thenoisyobservation
y(m) can be modelled as
y(m) = b(m)x(m) + n(m) (1.6)
wherex (m)isthesignaltobedetected, n(m)isthenoiseand b(m) isa
binary-valuedstateindicatorsequencesuchthat b(m) =1indicatesthe
presenceofthesignal x(m) and b(m) = 0 indicatesthatthesignalisabsent.
Ifthesignal x(m) hasaknownshape,thenacorrelatororamatched filter
.
.
.
x(0)
x(1)
x(2)
x(N-1)
.
.
.
X(0)
X(1)
X(2)
X(N-1)
.
.
.
.
.
.
X(0)
X(1)
X(2)
X(N-1)
Inputsignal BinarycodedsignalReconstructed
signal
x(0)
x(1)
x(2)
x(N-1)
^
^
^
^
^
^
^
^
n
0
bps
n
1
bps
n
2
bps
n
N-1
bps
T
r
a
n
s
f
o
r
m
T
E
n
c
o
d
e
r
D
e
c
o
d
e
r
.
.
.
I
n
v
e
r
s
e
T
r
a
n
s
f
o
r
m
T
-
1
Figure1.11 Illustrationofatransform-basedcoder.
ApplicationsofDigitalSignalProcessing 15
canbeusedtodetectthesignalasshowninFigure [Link]
response h(m)ofthematchedfilterfordetectionofasignal x(m) isthe
time-reversed version of x(m) given by
1 0 ) 1 ( ) ( = N m m N x m h (1.7)
whereN is the length of x(m) . The output of the matched filter is given by
=
=
1
0
) ( ) ( ) (
N
m
m y k m h m z (1.8)
The matched filter output iscomparedwithathresh oldandabinary
decision is made as
=
otherwise 0
threshold ) ( if 1
) (
m z
m b (1.9)
where
b(m)isanestimateofthebinarystateindicatorsequ ence b(m),and
itmaybeerroneousinparticularifthesignalto noiseratio is low. Table1.1
listsfourpossibleoutcomesthattogether b(m)anditsestimate
b(m) can
[Link] hesensitivityofthe
Matchedfilter
h(m) =x (N 1m)
y(m)=x(m)+n(m)
z(m)
Threshold
comparator
b(m)
^
Figure1.12 Configurationofamatchedfilterfollowedbyathresholdcomparatorfor
detectionofsignalsinnoise.
b(m)
b(m) Detector decision
0 0 Signal absent Correct
0 1 Signal absent ( Missed)
1 0 Signal present ( False alarm )
1 1 Signal present Correct
Table1.1 Fourpossibleoutcomesinasignaldetectionproblem.
16 Introduction
[Link],thelesstheli kelihoodthatnoisewould
beclassifiedassignal,sothefalsealarmratefa lls,buttheprobabilityof
misclassification of signal as noise increases. The risk in choosing a
threshold value can be expressed as
( ) ) ( ) ( Threshold
Miss Alarm False
P P + = = R (1.10)
Thechoiceofthethresholdreflectsatrade-offbe tweenthemisclassification
rateP
Miss
() and the false alarm rate P
False Alarm
().
1.3.7DirectionalReceptionofWaves:Beam-forming
Beam-formingisthespatialprocessingofplanewav esreceivedbyanarray
ofsensorssuchthatthewavesincidentatapartic ularspatialangleare
passedthrough,whereasthosearrivingfromotherd irectionsareattenuated.
Beam-formingisusedinradarandsonarsignalproc essing(Figure1.13)to
steerthereceptionofsignalstowardsadesireddi rection,andinspeech
processing for reducing the effects of ambient nois e.
Toexplaintheprocessofbeam-formingconsidera uniformlineararray
[Link] linear array impliesthat
thearrayofsensorsisspatiallyarrangedinastr aightlineandwithequal
spacing [Link] ldplanewave
withafrequency F
0
propagatingtowardsthesensorsatanincidencean gle
[Link] ssamplestheincoming
Figure1.13 Sonar:detectionofobjectsusingtheintensityandtimedelayof
reflectedsoundwaves.
ApplicationsofDigitalSignalProcessing 17
[Link] thewavetotravela
distance of d between two adjacent sensors is given by
=
d sin
c
(1.11)
where cisthespeedofpropagationofthewaveinthemed [Link]
difference corresponding to a delay of is given by
c
d
F
T
sin
2 2
0
0
= = (1.12)
whereT
0
[Link] priatecorrective
W
N1,P1
W
N1,1 W
N1,0
+
0
1
N-1
Array of sensors
I
n
c
i
d
e
n
t
p
l
a
n
e
w
a
v
e
Array offilters
Output
.
.
.
.
.
.
. ..
W
2,P1
W
2,1
W
2,0
+
.. .
z
1
W
1,P1
W
1,1 W
1,0
+
. ..
d
dsin
z
1
z
1
z
1
z
1
z
1
Figure1.14 Illustrationofabeam-former,fordirectionalreceptionofsignals.
18 Introduction
timedelaysinthepathofthesamplesateachsens or,andthenaveragingthe
outputsofthesensors,thesignalsarrivingfromt hedirection willbetime-
aligned and coherently combined, whereas those arri ving from other
directionswillsuffercancellationsandattenuatio ns.Figure1.14illustratesa
beam-formerasanarrayofdigitalfiltersarranged [Link]
actsasatwodimensionalspacetimesignalprocess [Link]
filteringallowsthebeam-formertobesteeredtowa rdsadesireddirection,
forexampletowardsthedirectionalongwhichthei ncomingsignalhasthe
[Link] sthetimedelay,and can
[Link] magnitudefrequency
response of each filter can be used to remove the o utofband noise.
1.3.8DolbyNoiseReduction
Dolbynoisereductionsystemsworkbyboostingthe energy and the signal to
noiseratioofthehighfrequencyspectrumofaudio [Link]
audio signals is mostly concentratedinthelowfre quencypartofthe
spectrum(below2kHz).Thehigherfrequenciesthat conveyqualityand
sensationhaverelativelylowenergy,andcanbede gradedevenbyalow
[Link] dedonamagnetictape,
thetapehissnoiseaffectsthequalityofthere [Link],
thehigherfrequencypartofanaudiosignalrecord edonatapehavesmaller
signaltonoiseratiothanthelowfrequencyparts. Thereforenoiseathigh
frequenciesismoreaudibleandlessmaskedbythe [Link]
noisereductionsystemsbroadlyworkontheprincip leofemphasisingand
boostingthelowenergyofthehighfrequencysigna lcomponentspriorto
[Link] isprocessedandencoded
using a combination of a pre-emphasis filter and dy namic range
[Link],thesignalisrecoveredu singadecoderbasedon
acombinationofade-emphasisfilterandadecompr [Link]
encoderanddecodermustbewellmatchedandcancel outeachotherin
order to avoid processing distortion.
Dolbyhasdevelopedanumberofnoisereductionsy stemsdesignated
DolbyA,[Link] nthenumberofbands
andthepre-emphasisstrategythatthattheyemploy .DolbyA,developedfor
professionaluse,dividesthesignalspectruminto fourfrequencybands:
band 1 is low-pass and covers 0 Hz to 80 Hz; band 2 is band-pass and covers
80Hzto3kHz;band3ishigh-passandcoversabov e3kHz;andband4is
[Link] encod er the gain of each band
ApplicationsofDigitalSignalProcessing 19
isadaptivelyadjustedtoboostlowenergysignalc [Link]
providesamaximumgainof10to15dBineachband ifthesignallevel
[Link] DolbyBandDolbyC
systemsaredesigned for consumer audio systems, an d use two bands instead
[Link] aboostofupto10dB
whenthesignallevelislow(lessthan45dBthan themaximumreference)
and Dolby C provides a boost of up to 20 dB as illu strated in Figure1.15.
1.3.9RadarSignalProcessing:DopplerFrequencySh ift
Figure1.16showsasimplediagramofaradarsyste mthatcanbeusedto
estimatetherangeandspeedofanobjectsuchasa movingcaroraflying
[Link] (transmitter/receiver) that
generatesandtransmitssinusoidalpulsesatmicrow [Link]
signaltravelswiththespeedoflightandisrefle ctedbackfromanyobjectin
[Link] ssuchinformationas
range, speed, and acceleration. The received signal has the form
0.1
1.0 10
-35
-45
-40
-30
-25
R
e
l
a
t
i
v
e
g
a
i
n
(
d
B
)
Frequency(kHz)
Figure1.15 Illustrationofthepre-emphasisresponseofDolby-C:upto20dB
boostisprovidedwhenthesignalfalls45dBbelowmaximumrecordinglevel.
20 Introduction
]} / ) ( 2 [ cos{ ) ( ) (
0
c t r t t A t x = (1.13)
whereA (t),thetime-varyingamplitudeof the reflected wave , depends on the
positionandthecharacteristicsofthetarget, r(t)isthetime-varyingdistance
oftheobjectfromtheradarand [Link]-varying
distance of the object can be expanded in a Taylor series as
L & & & & & & + + + + =
3 2
0
! 3
1
! 2
1
) ( t r t r t r r t r (1.14)
where r
0
isthedistance, r& isthevelocity, r& & istheaccelerationetc.
Approximating r(t)withthefirsttwotermsoftheTaylorseriesexp ansion
we have
t r r t r & +
0
) ( (1.15)
Substituting Equation (1.15) in Equation (1.13) yie lds
] / 2 ) / 2 cos[( ) ( ) (
0 0 0 0
c r t c r t A t x = & (1.16)
Note that the frequency of reflected wave is shifte d by an amount
c r
d
/ 2
0
& = (1.17)
ThisshiftinfrequencyisknownastheDopplerfre [Link]
movingtowardstheradarthenthedistance r(t)isdecreasingwithtime, r& is
negative,andanincreaseinthefrequencyisobser [Link]
r=
0
.5T
c
cos(
0
t)
Cos{
0
[t-2r(t)/c]}
Figure1.16 Illustrationofaradarsystem.
SamplingandAnalogtoDigitalConversion 21
objectismovingawayfromtheradarthenthedista ncer(t) is increasing, r& is
positive,andadecreaseinthefrequencyisobserv [Link]
analysisofthereflectedsignalcanrevealinforma tiononthedirectionand
speed of the object. The distance r
0
is given by
c T r = 5 . 0
0
(1.18)
whereTistheround-triptimeforthesignaltohittheo bjectandarriveback
at the radar and c is the velocity of light.
1.4SamplingandAnalogtoDigitalConversion
Adigitalsignalisasequenceofrealvaluedorco mplexvaluednumbers,
representingthefluctuationsofaninformationbea ringquantitywithtime,
[Link] basic elementarydiscrete-timesignalis
the unit-sample signal (m)defined as
(m) =
1 m = 0
0 m 0
(1.19)
[Link] x(m)canbeexpressedas
the sum of a number of amplitude-scaled and time-sh ifted unit samples as
x(m) = x(k)(m k)
k=
(1.20)
[Link] yrandomprocesses,such
asspeech,music,radarandsonargeneratesignals thatarecontinuousin
Discretetime
m
Figure1.17 Adiscrete-timesignalanditsenvelopeofvariationwithtime.
22 Introduction
[Link] saretermedanalog
becausetheirfluctuationswithtimeareanalogous tothevariationsofthe
[Link],analogsigna lsaresampled,andeach
sampleisconvertedintoan [Link]
performedsuchthattheoriginalsignalcanbereco veredfromitsdigital
versionwithnolossofinformation,andwithashi ghafidelityasisrequired
inanapplication.Figure1.18illustratesablock diagramconfigurationofa
[Link] low-passfilterremoves
outofbandsignalfrequenciesaboveapre-selected [Link]
andhold (S/H) unit periodically samples the signal to convert the
continuous-time signal into a discrete-time signal.
The analogtodigital converter (ADC) maps each co ntinuous
amplitudesampleintoan [Link],thedigitaloutputof
theprocessorcanbeconvertedback into an analog signal using a digitalto
analog converter (DAC) and a low-pass filter as ill ustrated in Figure 1.18.
1.4.1Time-DomainSamplingandReconstructionofAn alog
Signals
Theconversionofananalogsignaltoasequenceof n-bitdigitsconsistsof
[Link] amplingprocess,when
performedwithsufficientlyhighspeed,cancapture thefastestfluctuations
ofthesignal,andcan be a loss-less operation in that the analog signal can be
recoveredthroughinterpolationofthesampledsequ enceasdescribedin
[Link] n-bitdigit,involves
some irrevocable error and possible loss of informa tion. However, in
practice the quantisation error can be made negligi ble by using an
appropriatelyhighnumberofbitsasinadigitala [Link]
signalcanbemodelledastheproductofacontinuo us-timesignal x(t)anda
periodic impulse train p(t)as
Analog input
y(t)
LPF &
S/H
ADC
DAC
LPF
y(m) x(m) x(t)
Digital signal
processor
x
a
(m) y
a
(m)
Figure1.18 Configurationofadigitalsignalprocessingsystem.
Sampling and AnalogtoDigital
Conversion
23
=
=
=
m
s
mT t t x
t p t x t x
) ( ) (
) ( ) ( ) (
sampled
(1.21)
where T
s
isthesamplingintervalandthesamplingfunction p(t)isdefined
as
p(t) = (t mT
s
)
m=
(1.22)
Thespectrum P( f )ofthesamplingfunction p(t)isalsoaperiodicimpulse
train given by
=
=
k
s
kF f f P ) ( ) ( (1.23)
whereF
s
=1/T
s
[Link] twotime-
domainsignalsisequivalenttotheconvolutionof theirfrequencyspectra
we have
=
= = =
k
s
kF f f P f X t p t x FT f X ) ( ) ( * ) ( )] ( ). ( [ ) (
sampled
(1.24)
wheretheoperator FT[.] [Link](1.24)
theconvolutionofasignalspectrum X( f )witheachimpulse ) (
s
kF f ,
shifts X( f ) andcentresiton kF
s
. Hence,asexpressedinEquation(1.24),
thesamplingofasignalx (t)resultsinaperiodicrepetitionofitsspectrum
X( f ) centred on frequencies K , 2 , , 0
s s
F F . When the sampling
frequencyishigherthan twice the maximum frequenc y content of the signal,
thentherepetitionsofthesignalspectraaresepa ratedasshowninFigure
[Link],theanalogsignalcanberecove redbypassingthesampled
signalthroughananaloglow-passfilterwithacut -offfrequencyof F
s
.Ifthe
samplingfrequencyislessthan2 F
s
,thentheadjacentrepetitionsofthe
spectrum overlap and the original spectrum cannot b e recovered. The
distortion,duetoaninsufficientlyhighsampling rate,isirrevocableandis
24 Introduction
knownas [Link] Nyquistsampling
theoremwhichstates: aband-limitedcontinuous-timesignal,withahighe st
frequencycontent(bandwidth)of BHz,canberecoveredfromitssamples
provided that the sampling speed F
s
>2B samples per second.
Inpracticesamplingisachievedusinganelectroni cswitchthatallowsa
capacitortochargeupordowntothelevelofthe inputvoltageonceevery
T
s
[Link]- and-holdsignalcanbe
modelledastheoutputofafilterwitharectangul arimpulseresponse,and
Time domain
Frequency domain
Impulse-train-sampling
function
Sample-and-holdfunction
x(t)
t
X(f)
f
f
F
s
F
s
=1/Ts
0
Ts
x
p
(t)
Xp( f )
sh(t)
SH( f )
X
sh
(t) |X( f )|
f
f
f
t
t
S/H-sampled signal
Impulse-train-sampled
signal
B B
2B
...
...
0
*
= =
Ts 0
*
=
=
0
t
t
F
s
/2
...
. ..
.. .
. .. . ..
... ...
F
s
/2
F
s
F
s
=
=
k
s
kF f f P ) ( ) (
0
F
s
/2 F
s
/2
Figure1.19 Sample-and-Holdsignalmodelledasimpulse-trainsamplingfollowed
byconvolutionwitharectangularpulse.
Sampling and AnalogtoDigital
Conversion
25
with the impulsetrainsampled signal as the input as illustrated in
Figure1.19.
1.4.2Quantisation
For digital signal processing, continuous-amplitude samples from the
sample-and-holdarequantisedandmappedinto [Link]
quantisationto nbits,theamplituderangeofthesignalisdivided into2
n
discretelevels,andeachsampleisquantisedtoth enearestquantisation
level,andthenmappedtothebinarycodeassigned tothatlevel.Figure1.21
illustratesthequantisationofasignalinto4 dis crete levels. Quantisation is a
many-to-onemapping,inthatallthevaluesthatfa llwithinthe continuum of
aquantisationbandaremappedtothecentreofthe [Link]
betweenananalogsample x
a
(m)anditsquantisedvalue x (m) canbe
expressed as
[ ] ) ( ) ( m x Q m x
a
= (1.25)
whereQ[] is the quantising function.
Theperformanceofaquantiserismeasuredbysign altoquantisation
noise ratio SQNR per bit. The quantisation noise is defined as
) ( ) ( ) ( m x m x m e
a
= (1.26)
C
R
2
R
1
x(t)
x(mT
s
)
T
s
Figure1.20 Asimplifiedsample-and-holdcircuitdiagram.
26 Introduction
Nowconsideran n-bitquantiserwithanamplituderangeof [Link]
quantisationstepsizeis =2V/2
n
.Assumingthatthequantisationnoiseisa
zero-meanuniformprocesswithanamplituderangeo f/2wecanexpress
the noise power as
[ ] ( )
3
2
12
) ( ) (
1
) ( ) ( ) ( ) (
2 2 2
2 /
2 /
2
2 /
2 /
2 2
n
E
V
m de m e
m de m e m e f m e
= =
= =
E
(1.27)
where f
E
e(m)
( )
=1/ is the uniform probability density function of the
noise. Using Equation (1.27) he signaltoquantisat ion noise ratio is given
by
n
P
V
V
P
m
m x
n SQNR
n
n
e
6 77 . 4
2 log 10 log 10 3 log 10
3 / 2
log 10
) (
) (
log 10 ) (
2
10
Signal
2
10 10
2 2
Signal
10
2
2
10
] [
] [
+ =
+
|
|
\
|
=
|
|
\
|
=
|
|
\
|
=
E
E
(1.28)
where P
signal
isthemeansignalpower,and istheratioindecibelsofthe
peaksignalpower V
2
tothemeansignalpower P
signal
.Therefore,from
Equation(1.28)everyadditionalbitinananalogt odigitalconverterresults
in 6 dB improvement in signaltoquantisation noise ratio.
Sampling and AnalogtoDigital
Conversion
27
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Continuousamplitude samples
Discreteamplitude samples
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V
Figure1.21 Offset-binaryscalarquantisation
28 Introduction
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