DSP in Hearing Aids and Cochlear Implants

183 Pages·2007·11.27 MB·English

EURASIP Journal on Applied Signal Processing
DSP in Hearing Aids
and Cochlear Implants
Guest Editors: Simon Doclo, Søren Holdt Jensen, Philippe A. Pango,
Sören K. Riis, and Jan Wouters
EURASIP Journal on Applied Signal Processing
DSP in Hearing Aids and Cochlear Implants
EURASIP Journal on Applied Signal Processing
DSP in Hearing Aids and Cochlear Implants

Guest Editors: Simon Doclo, Søren Holdt Jensen,
Philippe A. Pango, Sören K. Riis, and Jan Wouters
Copyright © 2005 Hindawi Publishing Corporation. All rights reserved.
This is a special issue published in volume 2005 of “EURASIP Journal on Applied Signal Processing.” All articles are open access
articles distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction
in any medium, provided the original work is properly cited.
Editor-in-Chief
Marc Moonen, Belgium
Senior Advisory Editor
K. J. Ray Liu, College Park, USA
Associate Editors
Gonzalo Arce, USA Arden Huang, USA King N. Ngan, Hong Kong
Jaakko Astola, Finland Jiri Jan, Czech Douglas O’Shaughnessy, Canada
Kenneth Barner, USA Søren Holdt Jensen, Denmark Antonio Ortega, USA
Mauro Barni, Italy Mark Kahrs, USA Montse Pardas, Spain
Jacob Benesty, Canada Thomas Kaiser, Germany Wilfried Philips, Belgium
Kostas Berberidis, Greece Moon Gi Kang, Korea Vincent Poor, USA
Helmut Bölcskei, Switzerland Aggelos Katsaggelos, USA Phillip Regalia, France
Joe Chen, USA Walter Kellermann, Germany Markus Rupp, Austria
Chong-Yung Chi, Taiwan Lisimachos P. Kondi, USA Hideaki Sakai, Japan
Satya Dharanipragada, USA Alex Kot, Singapore Bill Sandham, UK
Petar M. Djurić, USA C.-C. Jay Kuo, USA Dirk Slock, France
Jean-Luc Dugelay, France Geert Leus, The Netherlands Piet Sommen, The Netherlands
Frank Ehlers, Germany Bernard C. Levy, USA Dimitrios Tzovaras, Greece
Moncef Gabbouj, Finland Mark Liao, Taiwan Hugo Van hamme, Belgium
Sharon Gannot, Israel Yuan-Pei Lin, Taiwan Jacques Verly, Belgium
Fulvio Gini, Italy Shoji Makino, Japan Xiaodong Wang. USA
A. Gorokhov, The Netherlands Stephen Marshall, UK Douglas Williams, USA
Peter Handel, Sweden C. Mecklenbräuker, Austria Roger Woods, UK
Ulrich Heute, Germany Gloria Menegaz, Italy Jar-Ferr Yang, Taiwan
John Homer, Australia Bernie Mulgrew, UK
Contents
Editorial, Simon Doclo, Søren Holdt Jensen, Philippe A. Pango, Søren K. Riis, and Jan Wouters
Volume 2005 (2005), Issue 18, Pages 2911-2914
Signal Processing in High-End Hearing Aids: State of the Art, Challenges, and Future Trends,
V. Hamacher, J. Chalupper, J. Eggers, E. Fischer, U. Kornagel, H. Puder, and U. Rass
Volume 2005 (2005), Issue 18, Pages 2915-2929
An Improved Array Steering Vector Estimation Method and Its Application in Speech Enhancement,
Zhu Liang Yu and Meng Hwa Er
Volume 2005 (2005), Issue 18, Pages 2930-2937
An Auditory-Masking-Threshold-Based Noise Suppression Algorithm GMMSE-AMT[ERB] for
Listeners with Sensorineural Hearing Loss, Ajay Natarajan, John H. L. Hansen,
Kathryn Hoberg Arehart, and Jessica Rossi-Katz
Volume 2005 (2005), Issue 18, Pages 2938-2953
Speech Enhancement with Natural Sounding Residual Noise Based on Connected Time-Frequency
Speech Presence Regions, Karsten Vandborg Sørensen and Søren Vang Andersen
Volume 2005 (2005), Issue 18, Pages 2954-2964
A Block-Based Linear MMSE Noise Reduction with a High Temporal Resolution Modeling of the
Speech Excitation, Chunjian Li and Søren Vang Andersen
Volume 2005 (2005), Issue 18, Pages 2965-2978
The Effects of Noise on Speech Recognition in Cochlear Implant Subjects: Predictions and Analysis
Using Acoustic Models, Jeremiah J. Remus and Leslie M. Collins
Volume 2005 (2005), Issue 18, Pages 2979-2990
Sound Classification in Hearing Aids Inspired by Auditory Scene Analysis, Michael Büchler,
Silvia Allegro, Stefan Launer, and Norbert Dillier
Volume 2005 (2005), Issue 18, Pages 2991-3002
Multichannel Dynamic-Range Compression Using Digital Frequency Warping,
James M. Kates and Kathryn Hoberg Arehart
Volume 2005 (2005), Issue 18, Pages 3003-3014
A Low-Power Two-Digit Multi-dimensional Logarithmic Number System Filterbank Architecture for
a Digital Hearing Aid, Roberto Muscedere, Vassil Dimitrov, Graham Jullien, and William Miller
Volume 2005 (2005), Issue 18, Pages 3015-3025
An Intrinsically Digital Amplification Scheme for Hearing Aids, Peter J. Blamey, David S. Macfarlane,
and Brenton R. Steele
Volume 2005 (2005), Issue 18, Pages 3026-3033
Effects of Instantaneous Multiband Dynamic Compression on Speech Intelligibility, Tobias Herzke
and Volker Hohmann
Volume 2005 (2005), Issue 18, Pages 3034-3043
A Psychoacoustic “NofM”-Type Speech Coding Strategy for Cochlear Implants, Waldo Nogueira,
Andreas Büchner, Thomas Lenarz, and Bernd Edler
Volume 2005 (2005), Issue 18, Pages 3044-3059
SPAIDE: A Real-time Research Platform for the Clarion CII/90K Cochlear Implant, L. Van Immerseel,
S. Peeters, P. Dykmans, F. Vanpoucke, and P. Bracke
Volume 2005 (2005), Issue 18, Pages 3060-3068
Ultra-Wideband Transceivers for Cochlear Implants, Thomas Buchegger, Gerald Oßberger,
Alexander Reisenzahn, Erwin Hochmair, Andreas Stelzer, and Andreas Springer
Volume 2005 (2005), Issue 18, Pages 3069-3075
Design of Low-Cost FPGA Hardware for Real-time ICA-Based Blind Source Separation Algorithm,
Charayaphan Charoensak and Farook Sattar
Volume 2005 (2005), Issue 18, Pages 3076-3086
EURASIPJournalonAppliedSignalProcessing2005:18,2911–2914
(cid:1)c 2005HindawiPublishingCorporation
Editorial
SimonDoclo
DepartmentofElectricalEngineering(ESAT-SCD),KatholiekeUniversiteitLeuven,KasteelparkArenberg10,3001Leuven,Belgium
Email:[email protected]
SørenHoldtJensen
DepartmentofCommunicationTechnology,InstituteofElectronicSystems,AalborgUniversity,FredrikBajersVej7A,
9220Aalborg,Denmark
Email:[email protected]
PhilippeA.Pango
GennumCorporation,P.O.Box489,StationA,Burlington,ON,CanadaL7R3Y3
Email:philip [email protected]
SørenK.Riis
OticonA/S,Kongebakken9,2765Smoerum,Denmark
Email:[email protected]
JanWouters
Exp.ORL,DepartmentofNeurosciences,KatholiekeUniversiteitLeuven,O&N,Herestraat49,bus721,3000Leuven,Belgium
Email:[email protected]
Digitalsignalprocessingforhearingaidswasinitiatedas suchsoundprocessingstrategiesposesadditionalsignalpro-
a topic of research in the mid-late 1980s. However, it was cessingchallenges,butatthesametimebuildsonknowledge
notuntil1995thatthetechnologymaturedtoalevelwhere acquiredthroughphysiologicalandpsychophysicalstudies.
small-size and low-power consumption allowed the market ThisspecialissueonDSPinHearingAidsandCochlear
introductionofhearingaidswithfulldigitalsignalprocess- Implants gathers 15 articles. It reflects aspects of the mul-
ingcapabilities. tiple disciplines necessary for the treatment of hearing im-
Today, 83% of hearing aids sold worldwide are digital. pairment. Indeed, the included papers address a variety of
Advancedpackagingtechnologiesenablehearingaidsthatfit methods and algorithms, all related to the research in sig-
completely in the ear canal, and the introduction of truly nal processing for hearing aids and cochlear implants. It is
programmable platforms has allowed the development of clearfromthesubmissionsthatthroughtheyears,theinclu-
advanced digital signal processing algorithms that provide sionofperceptioninsignalprocessingandthedevelopment
the hearing-impaired user a natural sound picture with in- ofpsychoacousticallymotivatedsignalprocessingalgorithms
creasedspeechintelligibilityandcomfort. arebecomingmoreandmorerelevantandimportant,asin
Moderncochlearimplantsystemsarecapableoffarmore otherdomainsofaudioprocessing.
advancedprocessingthanbefore.Whereascochlearimplants The papers in this issue are organized according to the
adopted digital technology prior to hearing aids, it is only topic of research, since some of these contributions are ap-
until very recently that they have integrated some special- plicable to both hearing aids and cochlear implants. The
izedalgorithmssuchasadaptivenoisereduction.Acochlear most frequent themes are speech intelligibility, speech en-
implantneeds,inaddition,aspeechprocessingstrategythat hancementandnoisesuppression(6papers),andnewsignal
convertstheacousticalsignalintoelectricalsignalstobeap- processingdevelopmentsinfilterbanksandcompressional-
plied to the electrodes placed in the cochlea. The design of gorithmsimplementation(4papers).Furthermoretheissue
2912 EURASIPJournalonAppliedSignalProcessing
presents5contributionsfromvariousresearchdomainssuch traction attenuation rule in time-frequency regions where
asauditorysceneanalysisforclassificationofinputsounds, speechispresentandaconstantattenuationruleinregions
anew cochlear implant processing strategy, a versatile re- wherespeechisabsent.Theproposedspeechdetectiontech-
search platform for cochlear implant research, a new wire- niqueprovidessmoothlyconnectedtime-frequencyregions
lesslinkbetweentheexternalandinternalcochlearimplant inaperceptuallyfunctionalwayandenablesanewbiascom-
parts,andblindsourceseparation. pensation method for minimum-statistics-based noise esti-
“Signalprocessinginhigh-endhearingaids:stateofthe mation.Listeningtestsshowthattheproposedmethodpro-
art,challenges,andfuturetrends”(V.Hamacheretal.)pro- duces a higher mean opinion score than minimum mean-
vides a discussion of signal processing in modern hearing squareerrorlog-spectralamplitude(MMSE-LSA)speechen-
aids. The authors distinguish between two types of algo- hancementmethods.
rithms:thosethataimatcompensatingthehearinglossand Thepaper“Ablock-basedlinearMMSEnoisereduction
improvinghearingabilityandthosethataimatcompensat- withahightemporalresolutionmodelingofthespeechex-
ing side effects of hearing aids. The former category com- citation” (C. Li and S. V. Andersen) proposes a method for
prises,forexample,amplificationstrategies,noisereduction, single-channel speech enhancement. The method is based
and directional (beamformer) systems, whereas the latter on an all-pole model of speech production and estimates
comprises, for example, acoustic feedback cancellation and the clean speech spectral envelope and LPC residual sep-
automaticcontrolofthesignalprocessinginthehearingaid. arately by a frequency-domain version of the linear mini-
Foreachsignalprocessingcomponentadiscussionoffuture mum mean-square error estimator. Objective performance
trendsisgiven. measuresshowthattheproposedmethodcomparesadvan-
In“Animprovedarraysteeringvectorestimationmethod tageouslytoknownmethodsforspeechsignalsinwhitenoise
anditsapplicationinspeechenhancement”(Z.L.YuandM. atanSNRof10dB.
H. Er), a multimicrophone speech enhancement method is In“Theeffectsofnoiseonspeechrecognitionincochlear
presented.Thismethodisanextensionofthetransferfunc- implant subjects: predictions and analysis using acoustic
tion generalized sidelobe canceller (TF-GSC), developed by models” (J. J. Remus and L. M. Collins), the reduction of
Gannotetal.,wheretheacoustictransferfunctionsbetween speech recognition performance in the presence of noise is
thedesiredspeechsourceandthemicrophonearrayarees- discussed for patients with cochlear implants. In the paper,
timatedandusedinthedesignofthefixedbeamformerand listening tests using normal-hearing subjects are conducted
theblockingmatrixoftheGSC.Insteadofusingoneofthe on noisy consonant and vowels processed by two different
microphone signals as the reference signal, this paper pro- acousticmodelsofcochlearimplantprocessors.Anextensive
posestouseanoptimalcombinationofallavailablemicro- analysisoftheresultsisgivenalongwithstatisticalmodelsfor
phone signals as the reference signal. Hence, by increasing predictingpatternsofvowelandconsonantconfusionbased
thesignal-to-noiseratioofthereferencesignal,theaccuracy ontheprocessedspeechtokensinthelisteningtest.
oftheestimatedacoustictransferfunctionsisimproved. “Soundclassificationinhearingaidsinspiredbyauditory
In“Anauditory-masking-threshold-basednoisesuppres- sceneanalysis”(M.Bu¨chleretal.)presentsasystematiceval-
sion algorithm GMMSE-AMT[ERB] for listeners with sen- uation ofclassifiersandfeaturesforspeechandmusic clas-
sorineural hearing loss” (A. Natarajan et al.), a new noise sification in a hearing-aid application. The considered fea-
suppression algorithm for hearing aid applications is de- turescomprise,forexample,amplitudemodulationandhar-
scribed. The algorithm is based on an approach that uses monicity,andacomparisonbetweenperformanceofsimple
the auditory masked threshold (AMT) in conjunction with classifiersandcomplexclassifierslikehiddenMarkovmodels
amodifiedgeneralizedminimummeansquareerrorestima- is given. It is illustrated that good performance can be ob-
tor (GMMSE) to adjust enhancement parameters based on tained even with simple classifiers in many situations, but
themaskedthresholdofthenoiseacrossthefrequencyspec- alsothatmostclassifiersyieldpoorperformanceforspeech
trum. The new algorithm also establishes a framework for innoise.
customizationoftheAMTestimationtoindividualsubjects In“Multichanneldynamic-rangecompressionusingdig-
withhearing loss. Therepresentation of cochlearfrequency ital frequency warping” (J. M. Kates and K. H. Arehart), a
resolution is achieved in terms of auditory filter equiva- novel multichannel dynamic-range compressor system us-
lent rectangular bandwidths (ERBs). The estimation of the ing digital frequency warping is described. The frequency-
AMT and spreading functions for masking is implemented warpedfilterisrealizedbyreplacingthefilterunitdelayswith
in two ways: with normal auditory thresholds and normal all-passfilters,andthewarpedcompressorisshowntohave
auditory filter bandwidths and with the elevated thresholds substantiallyreducedgroupdelayincomparisonwithacon-
and broader auditory filters characteristic of cochlear hear- ventionaldesignhavingcomparablefrequencyresolution.
ingloss. In “A low-power two-digit multi-dimensional logarith-
In “Speech enhancement with natural sounding resid- micnumbersystemfilterbankarchitectureforadigitalhear-
ual noise based on connected time-frequency speech pres- ingaid”(R.Muscedereetal.),theimplementationofafilter-
ence regions” (K. V. Sørensen and S. V. Andersen), a low- bankfordigitalhearingaidsusingamultidimensionalloga-
complexitysingle-microphonespeechenhancementmethod rithmicnumbersystem(MDLNS)isaddressed.Byexploiting
is presented. To achieve natural sounding attenuated back- various properties of the MDLNS, an improved design for
ground noise, this method uses a generalized spectral sub- atwo-digit2DMDLNSfilterbankimplementationreducing
Editorial 2913
thepowerandareabyover2timesfromtheoriginaldesign In “Design of low-cost FPGA hardware for real-time
ispresented. ICA-basedblindsourceseparationalgorithm”(C.Charoen-
“An intrinsically digital amplification scheme for hear- sak and F. Sattar), a real-time implementation of a modi-
ing aids” (P. Blamey et al.) suggests a new intrinsically dig- fied version of Torkkola’s convolutive blind source separa-
italamplificationschemeforhearingaids.Contrarytosome tionalgorithmisdescribed.Adiscussionofthetradeoffsbe-
existingamplificationstrategieslikelinearamplificationand tweenseparationperformanceandefficientimplementation
compression,thesuggestedmethodisnotadigital“reimple- isgivenanditisshownhowthealgorithmcanbemappedto
mentation”ofanestablishedtechniquefromananaloghear- anefficientimplementationonalow-costFPGAplatform.
ingaid.Thenewamplificationstrategyisbasedonstatistical
analysis of the signal and aims at maximizing the dynamic SimonDoclo
rangeineachfrequencybandinamultibandhearingaid.The SørenHoldtJensen
methodhasbeenimplementedonacommerciallyavailable PhilippeA.Pango
DSP. A comparison to existing schemes indicates improved SørenK.Riis
audibilityofsoundinnarrowfrequencybands. JanWouters
In “Effects of instantaneous multiband dynamic com-
pression on speech intelligibility” (T. Herzke and V.
Hohmann),instantaneousmultibanddynamiccompression
Simon Doclo was born in Wilrijk, Bel-
basedonanauditoryfilterbankisinvestigated.Instantaneous
gium, in 1974. Simon Doclo received the
envelopecompressionisperformedineachfrequencyband
M.S. degree in electrical engineering and
of a gammatone filterbank, which provides a combination
the Ph.D. degree in applied sciences from
oftimeandfrequencyresolutioncomparabletothenormal
the Katholieke Universiteit Leuven, Bel-
healthy cochlea. The gain characteristics used for dynamic gium,in1997and2003,respectively.Cur-
compressionarededucedfromcategoricalloudnessscaling. rently, he is a postdoctoral fellow of the
Bymeansofspeechintelligibilitytests,theinstantaneousdy- FundforScientificResearch-Flanders,af-
namic compression scheme is compared with a linear am- filiatedwiththeElectricalEngineeringDe-
plification scheme, which uses the same filterbank for fre- partment, Katholieke Universiteit Leuven.
quency analysis, but employs constant gain factors that re- During2005hewasaVisitingResearcherattheAdaptiveSystems
Laboratory, McMaster University, Canada. His research interests
storedthesoundlevelformediumperceivedloudnessineach
are in microphone array processing for acoustic noise reduction,
frequencyband.
dereverberationandsoundlocalization,adaptivefiltering,speech
In“Apsychoacoustic“NofM”-typespeechcodingstrat-
enhancement,andhearingaidtechnology.Dr.Docloreceivedthe
egy for cochlear implants” (W. Nogueira et al.), a new sig-
first prize “KVIV-Studentenprijzen” (with E. De Clippel) for his
nalprocessingtechniqueisdescribedforcochlearimplants.
M.S. thesis in 1997, a Best Student Paper Award at the Interna-
The scheme is based on the ACE strategy, as applied in de- tionalWorkshoponAcousticEchoandNoiseControlin2001,and
vicesofCochlear,butusesapsychoacousticmaskingmodel theEURASIPSignalProcessingBestPaperAward2003(withM.
inadditiontodeterminetheessentialcomponentsinthein- Moonen).HewasSecretaryoftheIEEEBeneluxSignalProcessing
putaudiosignal.Theyhavebeenabletoshowwithcochlear Chapter(1998–2002)andservesasaGuestEditoroftheEURASIP
implant users that improvements in speech understanding JournalonAppliedSignalProcessing.
areobtainedwhenasmallnumberofchannelsisstimulated
withinthesamecycle. SørenHoldtJensen receivedtheM.S.de-
gree in electrical engineering from Aal-
Thepaper“SPAIDE:areal-timeresearchplatformforthe
borg University, Denmark, in 1988, and
ClarionCII/90Kcochlearimplant”(L.VanImmerseeletal.)
the Ph.D. degree from the Technical Uni-
describesaplatformfortestsandexperimentswithcochlear
versity of Denmark in 1995. He has been
implantsoftheAdvancedBionicsCorporation.Itfacilitates
with the Telecommunications Laboratory
advancedresearchonsoundprocessingandelectricalstim-
of Telecom Denmark, the Electronics In-
ulation strategies with the Clarion CII and 90K implants. stitute of the Technical University of Den-
SPAIDE allows for real-time sound capturing, sound pro- mark, the Scientific Computing Group of
cessing,applicationofstimulationstrategy,andstreamingof theDanishComputingCenterforResearch
the outcome to the implant. This experimental platform is andEducation(UNI-C),theElectricalEngineeringDepartmentof
beingusedbydifferentresearchgroups. Katholieke Universiteit Leuven, Belgium, the Center for Person-
In “Ultra wideband transceivers for cochlear implants” Kommunikation(CPK)ofAalborgUniversity,andiscurrentlyan
Associate Professor at the Department of Communication Tech-
(T.Bucheggeretal.),thepracticalimplementationofanul-
nology,AalborgUniversity.Hisresearchactivitiesareindigitalsig-
trawideband(UWB)transceiverforcochlearimplantsisde-
nalprocessing,communicationsignalprocessing,andspeechand
scribed.AUWBlinkforadatarateof1.2Mbpsandaprop-
audio processing. Dr. Jensen is a Member of the Editorial Board
agation distance up to 500mm as well as transmitters with
of EURASIP Journal on Applied Signal Processing, an Associate
steprecoverydiodeandtransistorpulsegeneratorsarepro- Editor of IEEE Transactions on Signal Processing, and a former
posed.Moreover,twotypesofantennasandtheirfilterchar- ChairmanoftheIEEEDenmarkSectionanditsSignalProcessing
acteristicsintheUWBspectrumarediscussed. Chapter.