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Acoustics, Speech, and Signal Processing, IEEE International Conference on (2000)
Istanbul, Turkey
June 5, 2000 to June 9, 2000
ISBN: 0-7803-6293-4
TABLE OF CONTENTS

Author Index (Abstract)

pp. A1-A11

Tied posteriors: an approach for effective introduction of context dependency in hybrid NN/HMM LVCSR (Abstract)

J. Rottland , Dept. of Comput. Sci., Gerhard-Mercator-Univ. Duisburg, Germany
pp. 1241-1244

Discriminative resolution enhancement in acoustic modelling (Abstract)

J. Duchateau , ESAT, Katholieke Univ., Leuven, Heverlee, Belgium
pp. 1245-1248

A segmental-feature HMM using parametric trajectory model (Abstract)

Young-Sun Yun , Dept. of Comput. Sci., Korea Adv. Inst. of Sci. & Technol., Taejon, South Korea
pp. 1249-1252

Soft GPD for minimum classification error rate training (Abstract)

B.E. Shi , Dept. of Electr. & Electron. Eng., Hong Kong Univ. of Sci. & Technol., Kowloon, China
pp. 1253-1256

Heterogeneous lexical units for automatic speech recognition: preliminary investigations (Abstract)

I. Bazzi , Lab. for Comput. Sci., MIT, Cambridge, MA, USA
pp. 1257-1260

An effective acoustic modeling of names based on model induction (Abstract)

Taeyoon Kim , Dept. of Electron. Eng., Korea Univ., Seoul, South Korea
pp. 1265-1268

Agglomerative vs. tree-based clustering for the definition of multilingual set of triphones (Abstract)

B. Imperl , Fac. of Electr. Eng. & Comput. Sci., Maribor Univ., Slovenia
pp. 1273-1276

Integrating dynamic speech modalities into context decision trees (Abstract)

C. Fugen , Interactive Syst. Lab., Karlsruhe Univ., Germany
pp. 1277-1280

Automatic learning of numeral grammars for multi-lingual speech synthesizers (Abstract)

G. Flach , Lab. of Acoust. & Speech Commun., Tech. Univ. Dresden, Germany
pp. 1291-1294

Time and frequency scale modification of speech signals (Abstract)

B. Ninness , Dept. of Electr. & Comput. Eng., Newcastle Univ., Callaghan, NSW, Australia
pp. 1295-1298

Improving the robustness of wavelet transform for epoch detection (Abstract)

Y.Y. Lam , Dept. of Comput., Hong Kong Polytech. Univ., China
pp. 1303-1306

A weighted autocorrelation method for pitch extraction of noisy speech (Abstract)

H. Kobayashi , Dept. of Inf. & Comput. Sci., Saitama Univ., Urawa, Japan
pp. 1307-1310

Performance of the pitch-scaled harmonic filter and applications in speech analysis (Abstract)

P.J.B. Jackson , Dept. of Electron. & Comput. Sci., Southampton Univ., UK
pp. 1311-1314

Speech parameter generation algorithms for HMM-based speech synthesis (Abstract)

K. Tokuda , Dept. of Comput. Sci., Nagoya Inst. of Technol., Japan
pp. 1315-1318

Unsupervised estimation of the human vocal tract length over sentence level utterances (Abstract)

B.F. Necioglu , Sch. of Electr. & Comput. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1319-1322

Multivariate-state hidden Markov models for simultaneous transcription of phones and formants (Abstract)

M. Hasegawa-Johnson , Dept. of Electr. & Comput. Eng., Illinois Univ., Urbana, IL, USA
pp. 1323-1326

On the mutual information between frequency bands in speech (Abstract)

M. Nilsson , Dept. of Speech Music & Hearing, R. Inst. of Technol., Stockholm, Sweden
pp. 1327-1330

Study of talker individuality by using ARX speech analysis-synthesis-editing system (Abstract)

Weizhong Zhu , Adv. Technol. Res. Lab., Matsushita Electr. Ind. Co. Ltd., Kyoto, Japan
pp. 1331-1334

Linguistic properties of non-native speech (Abstract)

L.M. Tomokiyo , Language Technol. Inst., Carnegie Mellon Univ., Pittsburgh, PA, USA
pp. 1335-1338

Visual approach for automatic pitch period estimation (Abstract)

Zhang Sen , Dept. of Inf. & Comput. Sci., Waseda Univ., Tokyo, Japan
pp. 1339-1342

Robust pitch tracking for prosodic modeling in telephone speech (Abstract)

Chao Wang , Lab. for Comput. Sci., MIT, Cambridge, MA, USA
pp. 1343-1346

Perceptual effects of coarticulation in fricatives (Abstract)

S. Fernandez , Dept. of Appl. Phys., Santiago de Compostela Univ., Spain
pp. 1347-1350

Mel-scaled discrete wavelet coefficients for speech recognition (Abstract)

J.N. Gowdy , Dept. of Electr. & Comput. Eng., Clemson Univ., SC, USA
pp. 1351-1354

On-line speaking rate estimation using Gaussian mixture models (Abstract)

R. Faltlhauser , Inst. for Human-Machine-Commun., Tech. Univ. of Munich, Germany
pp. 1355-1358

Waveform extraction for perfect reconstruction in WI coding (Abstract)

V.T. Ruoppila , Dept. of Electr. Eng., Sherbrooke Univ., Que., Canada
pp. 1359-1362

High quality enhanced waveform interpolative coding at 2.8 kbps (Abstract)

O. Gottesman , Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA
pp. 1363-1366

Analysis-by-synthesis multimode harmonic speech coding at 4 kb/s (Abstract)

Chunyan Li , Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA
pp. 1367-1370

Speech coding with an analysis-by-synthesis sinusoidal model (Abstract)

C.O. Etemoglu , Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA
pp. 1371-1374

A 1200 bps speech coder based on MELP (Abstract)

Tian Wang , SignalCom Inc., Goleta, CA, USA
pp. 1375-1378

A 4 kb/s hybrid MELP/CELP coder with alignment phase encoding and zero-phase equalization (Abstract)

J. Stachurski , DSP Solutions R&D Center, Texas Instrum., Dallas, TX, USA
pp. 1379-1382

Perceptual phase redundancy in speech (Abstract)

Doh-Suk Kim , Human & Comput. Interaction Lab., Samsung Adv. Inst. of Technol., Kyonggi, South Korea
pp. 1383-1386

A combined WI and MELP coder at 5.2 kbps (Abstract)

J. Skoglund , AT&T Labs.-Res., Florham Park, NJ, USA
pp. 1387-1390

Variable rate multi-mode excitation coding of speech at 2.4 kbps (Abstract)

Shihua Wang , Atmel Corp., Berkeley, CA, USA
pp. 1395-1398

Hands-free speech recognition using a filtered clean corpus and incremental HMM adaptation (Abstract)

M. Matassoni , Centro per la Ricerca Sci. e Technol., Povo di Trento, Italy
pp. 1407-1410

HMM adaptation and microphone array processing for distant speech recognition (Abstract)

J. Kleban , CAIP Center, Rutgers Univ., Piscataway, NJ, USA
pp. 1411-1414

Comparing acoustic features for robust ASR in fixed and cellular network applications (Abstract)

F. de Wet , Dept. of Language & Speech, Nijmegen Univ., Netherlands
pp. 1415-1418

Anchoring hypothesis and its application to tone recognition of Chinese continuous speech (Abstract)

J.-S. Zhang , Dept. of Inf. & Commun. Eng., Tokyo Univ., Japan
pp. 1419-1422

Strategies for automatic segmentation of audio data (Abstract)

T. Kemp , Interactive Syst. Labs., Karlsruhe Univ., Germany
pp. 1423-1426

Conversational speech recognition using acoustic and articulatory input (Abstract)

K. Kirchhoff , Dept. of Electr. Eng., Washington Univ., Seattle, WA, USA
pp. 1435-1438

Harmonic exponential modeling of transitional speech segments (Abstract)

J. Jensen , Center for PersonKommunikation, Aalborg Univ., Denmark
pp. 1439-1442

Variable dimensional algebraic CELP coding of prototype waveforms (Abstract)

Jongseo Sohn , Sch. of Electr. Eng., Seoul Nat. Univ., South Korea
pp. 1443-1446

Low-rate quantization of spectrum parameters (Abstract)

T. Eriksson , Dept. of Signals & Syst., Chalmers Univ. of Technol., Goteborg, Sweden
pp. 1447-1450

Recursive LPC spectrum coding-a classified VQ approach (Abstract)

F. Norden , Inf. Theory Lab., Chalmers Univ. of Technol., Goteborg, Sweden
pp. 1451-1454

Encoding sinusoidal amplitudes with a minimum phase rational model (Abstract)

N. Malik , Sch. of Electr. Eng. & Telecommun., New South Wales Univ., Sydney, NSW, Australia
pp. 1463-1466

Phase and transient modeling for harmonic+noise speech coding (Abstract)

E.W.M. Yu , Dept. of Electr. Eng., City Univ. of Hong Kong, Kowloon, China
pp. 1467-1470

Linear prediction incorporating simultaneous masking (Abstract)

J. Lukasiak , Whisper Labs., Wollongong Univ., NSW, Australia
pp. 1471-1474

A frame interpretation of sinusoidal coding and waveform interpolation (Abstract)

W. Bastiaan Kleijn , Dept. of Speech, Music & Hearing, R. Inst. of Technol., Stockholm, Sweden
pp. 1475-1478

Optimized estimation of spectral parameters for the coding of noisy speech (Abstract)

R. Martin , Inst. of Commun. Syst. & Data Process., Tech. Hochschule Aachen, Germany
pp. 1479-1482

Improved frame erasure concealment for CELP-based coders (Abstract)

J.C. de Martin , DSPS R&D, Texas Instrum. Inc., Dallas, TX, USA
pp. 1483-1486

A CELP-based hybrid digital-analog (HDA) joint source-channel speech coder (Abstract)

N. Phamdo , Dept. of Electr. & Comput. Eng., State Univ. of New York, Stony Brook, NY, USA
pp. 1487-1490

Pitch-synchronous linear-prediction analysis by synthesis with reduced pulse densities (Abstract)

D. Guerchi , INTS-Telecommun., Univ. du Quebec, Verdun, Que., Canada
pp. 1491-1494

Shaped fixed codebook search for CELP coding at low bit rates (Abstract)

E. Erzin , Lucent Technol. Bell. Labs., Whippany, NJ, USA
pp. 1495-1498

Digital watermarking of speech signals for the National Gallery of the Spoken Word (Abstract)

F.J. Ruiz , Dept. of Electr. Eng., Michigan State Univ., East Lansing, MI, USA
pp. 1499-1502

Dispersed-pulse codebook and its application to a 4 kb/s speech coder (Abstract)

K. Yasunaga , Mobile Commun. Res. Lab., Matsushita Res. Inst. Tokyo Inc., Kawasaki, Japan
pp. 1503-1506

Joint source-channel MMSE-decoding of speech parameters (Abstract)

S. Heinen , Inst. of Commun. Syst. & Data Process., Tech. Hochschule Aachen, Germany
pp. 1507-1510

Speech quality objective assessment using neural network (Abstract)

Qiang Fu , Nat. Key Lab. of ISN, Xidian Univ., Xi'an, China
pp. 1511-1514

Rapid likelihood calculation of subspace clustered Gaussian components (Abstract)

A. Aiyer , IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA
pp. 1519-1522

Pitch tracking and tone features for Mandarin speech recognition (Abstract)

M.A. Picheny , Philips Innovation Center, Taipei, Taiwan
pp. 1523-1526

Boosting Gaussian mixtures in an LVCSR system (Abstract)

G. Zweig , IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA
pp. 1527-1530

Using SIMD instructions for fast likelihood calculation in LVCSR (Abstract)

S. Kanthak , Lehrstuhl fur Inf. VI, Tech. Hochschule Aachen, Germany
pp. 1531-1534

Fast decoding in large vocabulary name dialing (Abstract)

J. Suontausta , Speech & Audio Syst. Lab., Nokia Res. Center, Tampere, Finland
pp. 1535-1538

On the incremental addition of regression classes for speaker adaptation (Abstract)

J. McDonough , Center for Language & Speech Processing, Johns Hopkins Univ., Baltimore, MD, USA
pp. 1539-1542

Inter-class MLLR for speaker adaptation (Abstract)

S.-J. Doh , Carnegie Mellon Univ., Pittsburgh, PA, USA
pp. 1543-1546

Model adaptation in line spectrum domain (Abstract)

An-Tze Yu , Dept. of Electr. Eng., Nat. Tsing Hua Univ., Hsinchu, Taiwan
pp. 1547-1550

MAP adaptation with subspace regression classes and tying (Abstract)

Kwok-Man Wong , Dept. of Comput. Sci., Hong Kong Univ. of Sci. & Technol., Clear Water Bay, China
pp. 1551-1554

DUcoder-the Duisburg University LVCSR stackdecoder (Abstract)

D. Willett , Dept. of Comput. Sci., Gerhard-Mercator-Univ., Duisburg, Germany
pp. 1555-1558

Turkish LVCSR: towards better speech recognition for agglutinative languages (Abstract)

K. Carki , Interactive Syst. Labs., Karlsruhe Univ., Germany
pp. 1563-1566

Employing heterogeneous information in a multi-stream framework (Abstract)

H. Christensen , Center for PersonKommunikation, Aalborg Univ., Denmark
pp. 1571-1574

Dictation of multiparty conversation using statistical turn taking model and speaker model (Abstract)

N. Murai , Dept. of Electr., Electron., & Comput. Eng., Waseda Univ., Tokyo, Japan
pp. 1575-1578

Automatic speech summarization based on word significance and linguistic likelihood (Abstract)

C. Hori , Dept. of Comput. Sci., Tokyo Inst. of Technol., Japan
pp. 1579-1582

Statistical knowledge based frame synchronous search strategies in continuous speech recognition (Abstract)

Zhanjiang Song , Dept. of Comput. Sci. & Technol., Tsinghua Univ., Beijing, China
pp. 1583-1586

Using posterior word probabilities for improved speech recognition (Abstract)

F. Wessel , Lehrstuhl fur Inf., Tech. Hochschule Aachen, Germany
pp. 1587-1590

Variable word rate N-grams (Abstract)

Y. Gotoh , Dept. of Comput. Sci., Sheffield Univ., UK
pp. 1591-1594

Integrating detailed information into a language model (Abstract)

Ruiqiang Zhang , ATR Interpreting Telephony Res. Labs., Kyoto, Japan
pp. 1595-1598

Ear-model derived features for automatic speech recognition (Abstract)

R. De Mori , LIA CERI-IUP, Univ. of Avignon, France
pp. 1603-1606

Bitstream-based feature extraction for wireless speech recognition (Abstract)

F. Mana , AT&T Labs.-Res., Florham Park, NJ, USA
pp. 1607-1610

A fuzzy approach for the equalization of cepstral variances (Abstract)

Wei-Wen Hung , Dept. of Electr. Eng., Ming-Chi Inst. of Technol., Taishan, Taiwan
pp. 1611-1614

A new approach to discriminative feature extraction using model transformation (Abstract)

M. Thomae , Res. & Technol., DaimlerChrysler AG, Ulm, Germany
pp. 1615-1618

A Markov random field based multi-band model (Abstract)

G. Gravier , ENST/TSI, CNRS, Paris, France
pp. 1619-1622

Auditory-based speech processing based on the average localized synchrony detection (Abstract)

A.M. Abdelatty Ali , Dept. of Electr. Eng., Pennsylvania Univ., Philadelphia, PA, USA
pp. 1623-1626

Data-driven RASTA filters in reverberation (Abstract)

M.L. Shire , Int. Comput. Sci. Inst., California Univ., Berkeley, CA, USA
pp. 1627-1630

Speech feature extraction using independent component analysis (Abstract)

Jong-Hwan Lee , Dept. of Electr. Eng., Korea Adv. Inst. of Sci. & Technol., Taejon, South Korea
pp. 1631-1634

Tandem connectionist feature extraction for conventional HMM systems (Abstract)

H. Hermansky , Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA
pp. 1635-1638

A unified context-free grammar and n-gram model for spoken language processing (Abstract)

Ye-Yi Wang , Speech Technol. Group, Microsoft Corp., Redmond, WA, USA
pp. 1639-1642

Integrating a context-dependent phrase grammar in the variable n-gram framework (Abstract)

Manhung Siu , Dept. of Electr. & Electron. Eng., Hong Kong Univ. of Sci. & Technol., Clearwater Bay, China
pp. 1643-1646

Putting it all together: language model combination (Abstract)

J.T. Goodman , Speech Technol. Group, Microsoft Corp., Redmond, WA, USA
pp. 1647-1650

Transcription and indexation of broadcast data (Abstract)

J.-L. Gauvnin , Spoken Language Process. Group, LIMSI-CNRS, Orsay, France
pp. 1663-1666

A baseline for the transcription of Italian broadcast news (Abstract)

F. Brugnara , Centro per la Ricerca Sci. e Tecnol., ITC-irst, Trento, Italy
pp. 1667-1670

Pronunciation ambiguity vs. pronunciation variability in speech recognition (Abstract)

M. Saraclar , Centre for Language & Speech Processing, Johns Hopkins Univ., Baltimore, MD, USA
pp. 1679-1682

Lexical modeling of non-native speech for automatic speech recognition (Abstract)

K. Livescu , Spoken Language Res. Group, MIT, Cambridge, MA, USA
pp. 1683-1686

Automatic generation of phone sets and lexical transcriptions (Abstract)

R. Singh , Dept. of Electr. & Comput. Eng., Carnegie Mellon Univ., Pittsburgh, PA, USA
pp. 1691-1694

Selecting articles from the language model training corpus (Abstract)

D. Klakow , Philips GmbH Forschungslab., Aachen, Germany
pp. 1695-1698

Syntactic heads in statistical language modeling (Abstract)

Jun Wu , Center for Language & Speech Process., Johns Hopkins Univ., Baltimore, MD, USA
pp. 1699-1702

Polyphone decision tree specialization for language adaptation (Abstract)

T. Schultz , Interactive Syst. Labs., Karlsruhe Univ., Germany
pp. 1707-1710

Enhanced language modelling with phonologically constrained morphological analysis (Abstract)

A.C. Fang , Dept. of Phonetics & Linguistics, Univ. Coll. London, UK
pp. 1711-1714

Long range language models for free spelling recognition (Abstract)

F. Thiele , Philips Res. Lab., Aachen, Germany
pp. 1715-1718

Model-based feature enhancement for noisy speech recognition (Abstract)

C. Couvreur , Lernout & Hauspie Speech Products, Wemmel, Belgium
pp. 1719-1722

Robust speech recognition using near-field superdirective beamforming with post-filtering (Abstract)

I.A. McCowan , Speech Res. Lab., Queensland Univ. of Technol., Brisbane, Qld., Australia
pp. 1723-1726

Noisy speech recognition using noise reduction method based on Kalman filter (Abstract)

M. Fujimoto , Dept. of Electron. & Inf., Ryukoku Univ., Ohtsu, Japan
pp. 1727-1730

Statistical estimation of unreliable features for robust speech recognition (Abstract)

P. Renevey , Signal Process. Lab., Swiss Fed. Inst. of Technol., Lausanne, Switzerland
pp. 1731-1734

Hands-free voice activation of personal communication devices (Abstract)

S.E. Bou-Ghazale , Media Access Speech Technol. Dev., Conexant Syst. Inc., Newport Beach, CA, USA
pp. 1735-1738

An acoustic measure for predicting recognition performance degradation (Abstract)

K. Takeda , Grad. Sch. of Eng., Nagoya Univ., Japan
pp. 1739-1742

Low complexity speaker independent command word recognition in car environments (Abstract)

S.K. Riis , Digital Signal Processing Group, Nokia Mobile Phones, Copenhagen, Denmark
pp. 1743-1746

A novel approach to robust speech endpoint detection in car environments (Abstract)

Liang-Sheng Huang , Panasonic Taiwan Labs. Co., Taipei, Taiwan
pp. 1751-1754

Decision tree based Mandarin tone model and its application to speech recognition (Abstract)

Yang Cao , Nat. Lab. of Pattern Recog., Acad. Sinica, Beijing, China
pp. 1759-1762

Parameter optimization for vocal tract length normalization (Abstract)

P. Dognin , Dept. of Electr. Eng., Pittsburgh Univ., PA, USA
pp. 1767-1770

Word-level rate of speech modeling using rate-specific phones and pronunciations (Abstract)

Jing Zheng , Speech Technol. & Res. Lab., SRI Int., Menlo Park, CA, USA
pp. 1775-1778

Specialized acoustic models for hyperarticulated speech (Abstract)

H. Soltau , Karlsruhe Univ., Germany
pp. 1779-1782

On the use of variable frame rate analysis in speech recognition (Abstract)

Qifeng Zhu , Dept. of Electr. Eng., California Univ., Los Angeles, CA, USA
pp. 1783-1786

A probabilistic union model for sub-band based robust speech recognition (Abstract)

Ji Ming , Sch. of Comput. Sci., Queen's Univ., Belfast, UK
pp. 1787-1790

Robust speech recognition over IP networks (Abstract)

B. Milner , BT Adastral Park, Adv. Communs. Eng., Martlesham Heath, UK
pp. 1791-1794

Fast speaker adaptation of artificial neural networks for automatic speech recognition (Abstract)

S. Dupont , TCTS-MULTITEL, Faculte Polytech. de Mons, Belgium
pp. 1795-1798

Word and phone level acoustic confidence scoring (Abstract)

S.O. Kamppari , Spoken Language Syst. Group, MIT, Cambridge, MA, USA
pp. 1799-1802

Contextual confidence measures for continuous speech recognition (Abstract)

G. Hernandez-Abrego , Dept. de Teoria del Senyal i Comunicacions, Univ. Politecnica de Catalunya, Barcelona, Spain
pp. 1803-1806

Confidence measure and incremental adaptation for the rejection of incorrect data (Abstract)

N. Moreau , CNET/DIH/DIPS, France Telecom, Lannion, France
pp. 1807-1810

Robust out-of-vocabulary rejection for low-complexity speaker independent speech recognition (Abstract)

C.C. Broun , Human Interface Lab., Motorola Inc., Tempe, AZ, USA
pp. 1811-1814

Meta-models for confidence estimation in speech recognition (Abstract)

S. Dasmahapatra , Sch. of Inf. Syst., Univ. of East Anglia, Norwich, UK
pp. 1815-1818

Evaluation of various confidence-based strategies for isolated word rejection (Abstract)

E. Tsiporkova , Speech Recognition Res., Lernout & Hauspie, Wemmel, Belgium
pp. 1819-1822

Use of word level side information to improve speech recognition (Abstract)

D. Vergyri , Center for Language & Speech Process., Johns Hopkins Univ., Baltimore, MD, USA
pp. 1823-1826

Confidence measure based language identification (Abstract)

F. Metze , Interactive Syst. Lab., Karlsruhe Univ., Germany
pp. 1827-1830

A new keyword spotting approach based on iterative dynamic programming (Abstract)

M. Silaghi , Artificial Intelligence Lab., Swiss Fed. Inst. of Technol., Lausanne, Switzerland
pp. 1831-1834

Frame-discriminative and confidence-driven adaptation for LVCSR (Abstract)

F. Wallhoff , Dept. of Comput. Sci., Gerhard-Mercator Univ. Duisberg, Germany
pp. 1835-1838

An adaptive subspace approach for speech enhancement (Abstract)

S. Gazor , Dept. of Electr. & Comput. Eng., Queen's Univ., Kingston, Ont., Canada
pp. 1839-1842

Narrowband to wideband conversion of speech using GMM based transformation (Abstract)

Kun-Youl Park , Dept. of Electron. Eng., Pusan Nat. Univ., South Korea
pp. 1843-1846

Signal/noise KLT based approach for enhancing speech degraded by colored noise (Abstract)

U. Mittal , Dept. of Electr. & Comput. Eng., State Univ. of New York, Stony Brook, NY, USA
pp. 1847-1850

Low-band extension of telephone-band speech (Abstract)

G. Miet , Philips Consumer Commun., Le Mans, France
pp. 1851-1854

A hybrid speech enhancement system based on HMM and spectral subtraction (Abstract)

M.H. Ghoreishi , Dept. of Electr. Eng., Amirkabir Univ. of Technol., Tehran, Iran
pp. 1855-1858

Enhancement of speech based on non-parametric estimation of a time varying harmonic representation (Abstract)

S. Dubost , Dept. TSI/LTCI, Ecole Nat. Superieure des Telecommun., Paris, France
pp. 1859-1862

Integrated noise reduction and echo cancellation for IS-136 systems (Abstract)

F. Basbug , Hughes Network Syst. Inc., Germantown, MD, USA
pp. 1863-1866

An improved cumulant-based blind speech separation method (Abstract)

Yeping Su , Hughes Network Syst. Inc., Germantown, MD, USA
pp. 1867-1870

Impulsive noise suppression using neural networks (Abstract)

I. Potamitis , Dept. of Electr. & Comput. Eng., Patras Univ., Greece
pp. 1871-1874
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