Abstract is missing.
- Numerical investigation of the Euler equations by means of wave digital filtersRonald Bernhardt, Dirk Dahlhaus. 1-4 [doi]
- A Gabor sampling theorem and some time-bandwidth implicationsRichard S. Orr. 1-4 [doi]
- Novel multirate processing of beamspace noise eigenvectorsMichael D. Zoltowski, Gregory M. Kautz. 1-4 [doi]
- Speech enhancement based on a new set of auditory constrained parametersSrinivas Nandkumar, John H. L. Hansen. 1-4 [doi]
- Phonological parsing for reversible letter-to-sound/sound-to-letter generationHelen M. Meng, Stephanie Seneff, Victor W. Zue. 1-4 [doi]
- Unsupervised segmentation of radar images using wavelet decomposition and cumulantsJean-Marc Boucher, Stephane Pleihers. 1-4 [doi]
- Morphological scale-space fingerprints and their use in object recognition in range imagesPaul T. Jackway, Wageeh W. Boles, Mohamed Deriche. 5-8 [doi]
- Computation and conditioning of the finite-discrete Gabor transformNicholas J. Redding, Garry N. Newsam. 5-8 [doi]
- A stochastic language model for speech recognition integrating local and global constraintsRyosuke Isotani, Shoichi Matsunaga. 5-8 [doi]
- Wave digital filters for the migration of seismic dataMichael Bolle. 5-8 [doi]
- Low residual noise speech enhancement utilizing time-frequency filteringGary H. Whipple. 5-8 [doi]
- Recursive CR bounds: algebraic and statistical accelerationMohammad Usman, Alfred O. Hero III. 5-8 [doi]
- Cepstrum based deconvolution for speech dereverberationAthina P. Petropulu, Suresh Subramaniam. 9-12 [doi]
- Improving sentence recognition in stochastic context-free grammarsAna L. N. Fred, José M. N. Leitão. 9-12 [doi]
- Multidimensional reactive elements on curvilinear coordinate systems and their MDWDF discretizationMichael Fries. 9-12 [doi]
- Detection of point targets in high resolution synthetic aperture radar imagesYing Wang, Rama Chellappa, Qinfen Zheng. 9-12 [doi]
- Multiuser array beamforming based on a neural network mappingAna I. Pérez-Neira, Miguel Angel Lagunas. 9-12 [doi]
- A class of wavelet kernels associated with wave propagationLevent Onural, Mefharet Kocatepe, Haldun M. Özaktas. 9-12 [doi]
- The optimal wavelet transform and translation invarianceFeng Bao, Nurgun Erdol. 13-16 [doi]
- Comparative performance of spectral subtraction and HMM-based speech enhancement strategies with application to hearing and designHamid Sheikhzadeh, Hossein Sameti, Li Deng, Robert L. Brennan. 13-16 [doi]
- An energy minimization approach to building detection in aerial imagesSanthana Krishnamachari, Rama Chellappa. 13-16 [doi]
- Stability verification of multidimensional Kirchoff circuits by suitable energy functionsGerald Hemetsberger. 13-16 [doi]
- Efficient chart parsing of speech recognition candidatesToshihisa Tashiro, Toshiyuki Takezawa, Tsuyoshi Morimoto, Masaaki Nagata. 13-16 [doi]
- Adaptive time-varying parametric modelingAydin Akan, Luis F. Chaparro. 13-16 [doi]
- Use of M-band wavelet transform for multidirectional and multiscale edge detectionTurgut Aydin, Yücel Yemez, Bülent Sankur, Emin Anarim, Oktay Alkin. 17-20 [doi]
- A new WDF three-port adaptor suitable for floating-point arithmetic and/or DSP implementationsXiaojian Liu, Leonard T. Bruton. 17-20 [doi]
- Integrating semantic constraints into the Sphinx-II recognition searchWayne Ward, Sunil Issar. 17-20 [doi]
- Fast parameter estimation algorithms for linear FM signalsPeter O'Shea. 17-20 [doi]
- Algebraic design of discrete multiwavelet transformsPeter Rieder, Jürgen Götze, Josef A. Nossek. 17-20 [doi]
- Noise suppression using a wavelet modelAnthony Teolis, John J. Benedetto. 17-20 [doi]
- Discrete modelling of plasma equations with ion motion using technique of wave digital filtersToshio Utsunomiya, Alfred Fettweis. 21-24 [doi]
- Detecting misrecognitions and out-of-vocabulary wordsSheryl R. Young. 21-24 [doi]
- Rank determination in time-series analysisDonald W. Tufts, Abhijit A. Shah. 21-24 [doi]
- A wavelet based mammographic systemAndrew F. Laine, Michael Lewis 0002, Fred J. Taylor. 21-24 [doi]
- Adaptation techniques for ambience and microphone compensation in the IBM Tangora speech recognition systemSubrata K. Das, Arthur Nádas, David Nahamoo, Michael Picheny. 21-24 [doi]
- Second-order statistics of the wavelet transform of multiplicative white stochastic processCheng Youn Lu. 21-24 [doi]
- Adaptive wave digital fan filtersAlfredo C. Tan, Sheng-Tsai Chen, Sankar Basu. 25-28 [doi]
- Orientation selective operators for ridge, valley, edge, and line detection in imageryJianxin Hou, Roberto H. Bamberger. 25-28 [doi]
- Arbitrary tilings of phase spaceThomas Frederick, Nurgun Erdol. 25-28 [doi]
- Co-channel speaker separation based on maximum-likelihood deconvolutionMichael I. Savic, Huiqin Gao, Jeffrey S. Sorensen. 25-28 [doi]
- Heuristic search integrating syntactic, semantic and dialog-level constraintsTatsuya Kawahara, Masahiro Araki, Shuji Doshita. 25-28 [doi]
- Fast algorithms for exponential data modelingHaesun Park, Sabine Van Huffel, Lars Eldén. 25-28 [doi]
- A reduced complexity ESPRIT method and its generalization to an antenna of partially unknown shapeAlain Marsal, Sylvie Marcos. 29-32 [doi]
- A review of signal processing education in ScotlandTariq S. Durrani. 29-32 [doi]
- A new speech enhancement technique with application to speaker identificationMichael A. Ramalho, Richard J. Mammone. 29-32 [doi]
- Time-frequency distribution seriesShie Qian, Dapang Chen. 29-32 [doi]
- On jointly learning the parameters in a character-synchronous integrated speech and language modelTung-Hui Chiang, Yi-Chung Lin, Keh-Yih Su. 29-32 [doi]
- Periodicity estimation in textured images using to ML approachJorge S. Marques. 29-32 [doi]
- Engineering education for the 21st centuryDelores M. Etter, Joseph Bordogna. 33-36 [doi]
- Acoustic modeling for speech recognition based on spotting of phonetic unitsLes T. Niles. 33-36 [doi]
- Detecting scene changes and activities in video databasesP. Robert Hsu, Hiroshi Harashima. 33-36 [doi]
- A multimodal, keyword-based spoken dialogue system-MultiksDialHiroshima Matsu'ura, Yasuyuki Masai, Jun'ichi Iwasaki, Shin'ichi Tanaka, Hiroyuki Kamio, Tsuneo Nitta. 33-36 [doi]
- An adaptive algorithm of linear computational complexity for both rank and subspace trackingBin Yang, Frank Gersemsky. 33-36 [doi]
- Performance analysis of the dual sign algorithm with contaminated-Gaussian noiseSeung Chan Bang, Hong Sub Choi, SouGuil Ann. 33-36 [doi]
- Local minima escape transients of CMAMichael R. Frater, C. Richard Johnson Jr.. 37-40 [doi]
- IPA: improved phone modelling with recurrent neural networksTony Robinson, Mike Hochberg, Steve Renals. 37-40 [doi]
- The Signal Processing Information Base project: the present and the futureDon H. Johnson, Sally Wood. 37-40 [doi]
- Error criteria analysis and robust data fusionXing Li, Weiru Fang, Qi Tian. 37-40 [doi]
- Learning consistent semantics from training dataRoland Kuhn, Renato de Mori, Evelyne Millien. 37-40 [doi]
- On vector quantization for fast facet edge detectionMysore Y. Jaisimha, Jill R. Goldschneider, Alexander E. Mohr, Eve A. Riskin, Robert M. Haralick. 37-40 [doi]
- A fast convergence algorithm for sparse-tap adaptive FIR filters for an unknown number of multiple echoesShigeji Ikeda, Akihiko Sugiyama. 41-44 [doi]
- Adaptive separation of independent sources: a deflation approachNathalie Delfosse, Philippe Loubaton. 41-44 [doi]
- Phoneme recognition in continuous speech using large inhomogeneous hidden Markov modelsR. N. V. Sitaram, Thippur V. Sreenivas. 41-44 [doi]
- The design and development of an undergraduate signal processing laboratorySridha Sridharan, Vinod Chandran, M. Dawson. 41-44 [doi]
- A contour-based part segmentation algorithmMohammed Bennamoun. 41-44 [doi]
- Parsing word graphs using a linguistic grammar and a statistical language modelLudwig A. Schmid. 41-44 [doi]
- Stabilization of tracks with multiple model filtersMichael O. Kolawole. 45-48 [doi]
- Phonetic classification and recognition using HMM representation of overlapping articulatory features for all classes of English soundsLi Deng, Don X. Sun. 45-48 [doi]
- Computing and signal processing: an experimental multidisciplinary courseEdward A. Lee. 45-48 [doi]
- A tapered SVD without rank determination for estimation of multipulse input time series from noisy outputHiroshi Kanai, Kazuhiro Ikikame, Noriyoshi Chubachi. 45-48 [doi]
- Minimum cost based phoneme class detection for improved iterative speech enhancementLevent M. Arslan, John H. L. Hansen. 45-48 [doi]
- An image processing algorithm for a super high definition imaging scheme with multiple different-aperture camerasTakahiro Saito, Takashi Komatsu, Kiyoharu Aizawa. 45-48 [doi]
- Segmental phoneme recognition using piecewise linear regressionS. Krishnan, P. V. S. Rao. 49-52 [doi]
- On bias in transfer functions estimated with stochastic excitationPiet M. T. Broersen. 49-52 [doi]
- An improvement on the reception of Pal-M television signal using an additional simple delay filterYuzo Iano. 49-52 [doi]
- On the limitations of cepstral features in noiseJohn P. Openshaw, John S. Mason. 49-52 [doi]
- A one stage self-adaptive algorithm for source separationEric Moreau, Odile Macchi. 49-52 [doi]
- "Digital signal processing with applications: " a new and successful approach to undergraduate DSP educationJan P. Allebach, Michael D. Zoltowski, Charles A. Bouman. 49-52 [doi]
- Programmable blind adaptive spatial filteringWilliam A. Gardner, Jeffrey L. Schenck, Stephan V. Schell. 53-56 [doi]
- A new method for segmenting continuous speechBasavaraj I. Pawate, Eric M. Dowling. 53-56 [doi]
- Performance analysis of subband adaptive systems using an equivalent modelYoshihiro Ono, Hitoshi Kiya. 53-56 [doi]
- An update on the CMU virtual laboratoryVirginia L. Stonick. 53-56 [doi]
- Image visual quality restoration by cancellation of the unmasked noiseBenoit M. Macq, Marco Mattavelli, Olivier Van Calster, Emmanuel Van der Plancke, Serge Comes, Wei Li 0046. 53-56 [doi]
- A family of MLP based nonlinear spectral estimators for noise reductionFei Xie, Dirk Van Compernolle. 53-56 [doi]
- Stochastic trajectory modeling for speech recognitionYifan Gong, Jean-Paul Haton. 57-60 [doi]
- On the use of decorrelation in scalar signal separationStefaan Van Gerven, Dirk Van Compernolle. 57-60 [doi]
- DSP and communication engineering education with graphical block diagram simulation toolsSasan H. Ardalan, Tülay Adali. 57-60 [doi]
- A 2-D IIR neural hybrid filter for image processingMitsuji Muneyasu, Satoshi Tsujii, Takao Hinamoto. 57-60 [doi]
- Markov model based noise modeling and its application to noisy speech recognition using dynamical features of speechTetsunori Kobayashi, Ryuji Mine, Katsuhiko Shirai. 57-60 [doi]
- A new cumulant based parameter estimation method for noncausal autoregressive systemsChong-Yung Chi, Jian-Lin Hwang. 57-60 [doi]
- Hierarchical stochastic modelling for speech compressionKie-Bum Eom, Rama Chellappa. 61-64 [doi]
- Adaptive α-trimmed mean filters with excellent detail-preservingAkira Taguchi. 61-64 [doi]
- Signal processing education in the French systemMessaoud Benidir, Guy Demoment, Bernard C. Picinbono. 61-64 [doi]
- A globally admissible off-line modulus restoral algorithm for low-order adaptive channel equalisersKutluyil Dogancay, Rodney A. Kennedy. 61-64 [doi]
- Using multiple vector quantization and semicontinuous hidden Markov models for speech recognitionAntonio M. Peinado, José C. Segura, Antonio J. Rubio, M. Carmen Benítez. 61-64 [doi]
- Environment normalization for robust speech recognition using direct cepstral comparisonFu-Hua Liu, Richard M. Stern, Alejandro Acero, Pedro J. Moreno. 61-64 [doi]
- An efficient combination of acoustic and supra-segmental informations in a speech recognition systemNelly Suaudeau, Régine André-Obrecht. 65-68 [doi]
- Blind wideband source separationMontse Nájar, Miguel Angel Lagunas, Ignasi Bonet. 65-68 [doi]
- Adaptive filtering in data communications with self improved error referenceJosé Manuel Páez-Borrallo, Iván A. Pérez Alvarez, Santiago Zazo-Bello. 65-68 [doi]
- A short history of the bootstrapPeter Hall. 65-68 [doi]
- Design of optimal median-type filters under structural constraintsBing Zeng. 65-68 [doi]
- Noisy speech recognition using cepstral-time features and spectral-time filtersSaeed Vaseghi, Ben P. Milner, Jason J. Humphries. 65-68 [doi]
- Speech recognition in noisy car environment based on OSALPC representation and robust similarity measuring techniquesJavier Hernando, Climent Nadeu. 69-72 [doi]
- Multiple bootstrap tests and their applicationAbdelhak M. Zoubir. 69-72 [doi]
- Fractionally spaced equalizers with adaptive samplingMiwa Sakai, Kiyoharu Aizawa, Mitsutoshi Hatori. 69-72 [doi]
- Statistical analysis of the median based multi-shell order-statistics filtersJ. S. Jimmy Li, Anand Ramsingh. 69-72 [doi]
- Word accent patterns modelling by concatenation of mora hidden Markov modelsTakashi Yoshimura, Satoru Hayamizu, Kazuyo Tanaka. 69-72 [doi]
- AR parameter estimation from noisy data using the EM algorithmMohamed Deriche. 69-72 [doi]
- Phonemic segmentation of fluent speechDavid B. Grayden, Michael S. Scordilis. 73-76 [doi]
- A new general global array calibration methodNikolaos Fistas, Athanassios Manikas. 73-76 [doi]
- GMLOS and a comparative study of nonlinear filtersHamid R. Rabiee, R. L. Kashyap. 73-76 [doi]
- Steady-state performance limitations of full-band acoustic echo cancellersMichael E. Knappe, Rafik Goubran. 73-76 [doi]
- Parallel adaptation for enhanced RLS trackingS. Douglas Peters, Andreas Antoniou. 73-76 [doi]
- Jackknifing multiple-window spectraDavid J. Thomson. 73-76 [doi]
- EEG dipole localization bounds for head models with parameter uncertaintiesBill M. Radich, Kevin M. Buckley. 77-80 [doi]
- Knowledge based approach to consonant recognitionAra Samouelian. 77-80 [doi]
- Exact analysis of the finite precision error generation and propagation in the FAEST and the fast transversal algorithms. A general methodology for developing stable a posteriori RLS computational schemesConstantin Papaodysseus, Cristos C. Halkias, Costas N. Triantafyllou, V. Asimakopoulos. 77-80 [doi]
- Auditory model inversion for sound separationMalcolm Slaney, Daniel Naar, Richard F. Lyon. 77-80 [doi]
- Bootstrapping coherency functionKiheon Choi, Myoungshic Jhun, Jae Chang Lee. 77-80 [doi]
- Statistical morphological filters for binary image processingCarlo S. Regazzoni, Anastasios N. Venetsanopoulos, Gian Luca Foresti, Gianni Vernazza. 77-80 [doi]
- Macrophone: an American English telephone speech corpus for the Polyphone projectJared Bernstein, Kelsey Taussig, John J. Godfrey. 81-84 [doi]
- Speech parameter extraction in noisy environment using a masking modelTsuyoshi Usagawa, Makoto Iwata, Masanao Ebata. 81-84 [doi]
- Bootstrap sampling applied to image analysisFaouzi Ghorbel, Calvin Banga. 81-84 [doi]
- Fixed and floating point error analysis of QRD-RLS and STAR-RLS adaptive filtersKalavai J. Raghunath, Keshab K. Parhi. 81-84 [doi]
- Array shape reconstruction for a nominally linear arrayJean-Jacques Fuchs. 81-84 [doi]
- A human-machine interactive system for efficient image restorationHong Tang. 81-84 [doi]
- A Bayesian approach to direction finding with parametric array uncertaintyMats Viberg, A. Lee Swindlehurst. 85-88 [doi]
- Constructing telephone acoustic models from a high-quality speech corpusMitchel Weintraub, Leonardo Neumeyer. 85-88 [doi]
- Non-linear regression based feature extraction for connected-word recognition in noiseFrank Seide, Alfred Mertins. 85-88 [doi]
- Blind superresolving image recovery from blur-invariant edgesKazuki Nishi, Shigeru Ando. 85-88 [doi]
- A bootstrap approach to estimating fractal dimensionsThomas Mikosch, David Vere-Jones, Qiang Wang. 85-88 [doi]
- New interference robust block adaptive filter with correlated signal inputShigenori Kinjo. 85-88 [doi]
- A new algorithm of realizing arbitrary nonlinear filters-adaptive neural filtersJian Zhan, Fu Li. 89-92 [doi]
- Multiple testing for seismic data using bootstrapDirk Maiwald, Johann F. Böhme. 89-92 [doi]
- Color quantization of images based on human vision perceptionNavin Chaddha, Wee-Chiew Tan, Teresa H. Y. Meng. 89-92 [doi]
- Efficient parameter estimation of partially polarized electromagnetic wavesJian Li, Petre Stoica. 89-92 [doi]
- A hierarchical LPNN network for noise reduction and noise degraded speech recognitionYuqing Gao, Jean-Paul Haton. 89-92 [doi]
- The voice across Japan database-the Japanese language contribution to PolyphoneThomas Staples, Joseph Picone, Nozomi Arai. 89-92 [doi]
- Jackknifed multiple-window spectra and coherence applied to seismic dataFrank L. Vernon. 93-96 [doi]
- Implementation and performance of an 8-kbit/s conjugate structure CELP speech coderAkitoshi Kataoka, Takehiro Moriya, Shinji Hayashi. 93-96 [doi]
- Towards automatic collection of the US censusRonald A. Cole, David G. Novick, Daniel C. Burnett, Brian Hansen, Stephen Sutton, Mark Fant. 93-96 [doi]
- Color quantization error in terms of perceived image qualityAlain Trémeau, Maurice Calonnier, Bernard Laget. 93-96 [doi]
- Parameter estimation for periodic discrete event processesDoug A. Gray, Benjamin J. Slocumb, Stephen D. Elton. 93-96 [doi]
- Multichannel L-filter design based on marginal data orderingConstantine Kotropoulos, Ioannis Pitas. 93-96 [doi]
- Greedy tree growing for color image quantizationTsann-Shyong Liu, Long-Wen Chang. 97-100 [doi]
- Spectrum reuse using transmitting antenna arrays with feedbackDerek Gerlach, Arogyaswami Paulraj. 97-100 [doi]
- The Australian National Database of Spoken LanguageJ. Bruce Millar, Julie Vonwiller, Jonathan Harrington, Phillip Dermody. 97-100 [doi]
- Minimum mean square error filtering over the class of extended threshold Boolean filtersKi-Dong Lee, Yong-Hoon Lee. 97-100 [doi]
- 8 kbit/s ACELP coding of speech with 10 ms speech-frame: a candidate for CCITT standardizationRedwan Salami, Claude Laflamme, Jean-Pierre Adoul. 97-100 [doi]
- Bootstrapping autoregressions with infinite orderJens-Peter Kreiss, Gordon Lien. 97-100 [doi]
- High frequency atmospheric noise mitigationJ. D. R. Kramer Jr., Ronald T. Williams. 101-104 [doi]
- A stable and globally convergent OEM IIR ADFJinhui Chao, Teruyuki Sato, Shigeo Tsujii. 101-104 [doi]
- A comparative analysis of Japanese and English digit recognitionKazuhiro Kondo, Joseph Picone, Barbara Wheatley. 101-104 [doi]
- Designing the color palette for textile material printing processGabriel Marcu, Kansei Iwata. 101-104 [doi]
- A computationally efficient wavelet transform CELP coderJames M. Ooi, Vishu Viswanathan. 101-104 [doi]
- Adaptive channel equalization for TDMA digital cellular communications using antenna arraysAyman F. Naguib, Babak Hossein Khalaj, Arogyaswami Paulraj, Thomas Kailath. 101-104 [doi]
- Fast convergent genetic search for adaptive IIR filteringSin Chun Ng, Chi Yin Chung, Shu Hung Leung, Andrew Luk. 105-108 [doi]
- Perceptual enhancement of CELP speech codersDipanjan Sen, W. Harvey Holmes. 105-108 [doi]
- Exterior noise adaptive rejection for OTH radar implementationsYuri I. Abramovich, Alexei Gorokhov, V. N. Mikhaylyukov, I. P. Malyavin. 105-108 [doi]
- Halftoning technique using genetic algorithmNaoki Kobayashi II, Hideo Saito. 105-108 [doi]
- Wavelet based detectorsRussell D. Priebe, Kevin W. Baugh. 105-108 [doi]
- Application of vector quantized hidden Markov modeling to telephone network based connected digit recognitionEric R. Buhrke, Régis Cardin, Yves Normandin, Mazin G. Rahim, Jay G. Wilpon. 105-108 [doi]
- Conversion of scanned documents to the open document architectureGary S. D. Farrow, Costas S. Xydeas, John P. Oakley. 109-112 [doi]
- Wavelet domain bearing estimation in unknown correlated noiseAhmed H. Tewfik. 109-112 [doi]
- CELP speech coding with almost no codebook searchChristian G. Gerlach. 109-112 [doi]
- Factorization approach to time-varying filter banks and waveletsRamesh A. Gopinath. 109-112 [doi]
- STUDIO-monostatic backscatter sounder father of NOSTRADAMUS French O.T.H. radar project-new resultsClaude Goutelard, Julien Caratori, L. Barthes. 109-112 [doi]
- Sources of degradation of speech recognition in the telephone networkPedro J. Moreno, Richard M. Stern. 109-112 [doi]
- 2-dimensional recursive orthogonal wavelet transformationHiromichi Yasuoka, Masaaki Ikehara. 113-116 [doi]
- Error diffusion with dynamically adjusted kernelPing Wah Wong. 113-116 [doi]
- Statistical signal processing for application to over-the-horizon radarAbdelhak M. Zoubir. 113-116 [doi]
- Statistical optimization of PR-QMF banks and waveletsAlfred Mertins. 113-116 [doi]
- The development of file formats for very large speech corpora: SPHERE and SHORTENJohn S. Garofolo, Tony Robinson, Jonathan G. Fiscus. 113-116 [doi]
- A pitch synchronous innovation CELP (PSI-CELP) coder for 2-4 kbit/sSatoshi Miki, Kazunori Mano, Takehiro Moriya, Kumiko Oguchi, Hitoshi Ohmuro. 113-116 [doi]
- A distortion measure for image artifacts based on human visual sensitivityShanika A. Karunasekera, Nick G. Kingsbury. 117-120 [doi]
- Multi-band residual coding of CELP codecs at 8 kb/sPaul Mermelstein, Ping Zheng, M. Saikaly. 117-120 [doi]
- Toward vocabulary independent telephone speech recognitionYu-Hung Kao, Charles T. Hemphill, Barbara Wheatley, Raja Rajasekaran. 117-120 [doi]
- Wavelets of composite typeJohn E. Gilbert, Joseph D. Lakey. 117-120 [doi]
- Signal processing of sky wave OTH-B radarWenyu Zhou, Peinan Jiao. 117-120 [doi]
- Wavelet extrema and zero-crossings representations: properties and consistent reconstructionZoran Cvetkovic, Martin Vetterli. 117-120 [doi]
- A real-time wideband CELP coder for a videophone applicationErik Harborg, Jan E. Knudsen, Arild Fuldseth, Finn Tore Johansen. 121-124 [doi]
- Analysis of multirate components and application to multirate filter designRam G. Shenoy. 121-124 [doi]
- Application of directional statistics in vector direction estimationNikos Nikolaidis, Ioannis Pitas. 121-124 [doi]
- Optimal receiver design with wavelet basesNurgun Erdol, Feng Bao, Filiz Basbug. 121-124 [doi]
- On the use of data-driven clustering technique for identification of poly- and mono-phonemes for four European languagesOve Andersen, Paul Dalsgaard, William J. Barry. 121-124 [doi]
- Space and time statistical data processing in HF ionospheric soundingV. G. Bezrodny, P. V. Ponomarenko, Yuri M. Yampolski. 121-124 [doi]
- Phase correction of an HF multireceiver antenna array using a radar transponderAlain Bourdillon, J. Delloue. 125-128 [doi]
- Large vocabulary continuous speech recognition using HTKPhilip C. Woodland, J. J. Odell, V. Valtchev, Steve J. Young. 125-128 [doi]
- Speaker adaptation of tied-mixture-based phoneme models for text-prompted speaker recognitionTomoko Matsui, Sadaoki Furui. 125-128 [doi]
- Regular M-band modulated orthogonal transformsJoël Mau. 125-128 [doi]
- Signal modeling using increments of extended self-similar processesLance M. Kaplan, C. C. Jay Kuo. 125-128 [doi]
- On-line cursive handwriting recognition using speech recognition methodsThad Starner, John Makhoul, Richard M. Schwartz, George Chou. 125-128 [doi]
- Robust cepstral features for speaker identificationKhaled T. Assaleh, Richard J. Mammone. 129-132 [doi]
- Large vocabulary continuous speech recognition of Wall Street Journal dataXavier L. Aubert, Christian Dugast, Hermann Ney, Volker Steinbiss. 129-132 [doi]
- Phase linearization of filters in analysis/synthesis filter banksTsuhan Chen, P. P. Vaidyanathan. 129-132 [doi]
- A new Wold ordering for image similarityRosalind W. Picard, Fang Liu. 129-132 [doi]
- On model identification for distortion correction of OTH radar signalsStuart J. Anderson, Susan E. Godfrey, S. M. Voigt. 129-132 [doi]
- A polynomial phase parameter estimation-phase unwrapping algorithmBenjamin J. Slocumb, John Kitchen. 129-132 [doi]
- Border recovery for subband processing of finite-length signals. Application to time-varying filter banksFrançois Déprez, Olivier Rioul, Pierre Duhamel. 133-136 [doi]
- Gaussian mixture model classifiers for machine monitoringLarry P. Heck, Kenneth C. Chou. 133-136 [doi]
- A quantitative assessment of the relative speaker discriminating properties of phonemesJulian P. Eatock, John S. D. Mason. 133-136 [doi]
- A novel architecture to model non-linear systemsAlba Pagès-Zamora, Miguel Angel Lagunas. 133-136 [doi]
- An initial study on a segmental probability model approach to large-vocabulary continuous Mandarin speech recognitionJia-lin Shen, Hsin-Min Wang, Bo-Ren Bai, Lin-Shan Lee. 133-136 [doi]
- Filter estimation maximization algorithm for image segmentationHocine Cherifi, Richard Grisel. 133-136 [doi]
- Volterra filtering with spectral constraintsRobert D. Nowak, Barry D. Van Veen. 137-140 [doi]
- Recognition of space curves based on the dyadic wavelet transformQuang Minh Tieng, Wageeh W. Boles. 137-140 [doi]
- Text-dependent speaker recognition using the information in the higher frequency bandShoji Hayakawa, Fumitada Itakura. 137-140 [doi]
- Cosine-modulated 2 dimensional FIR filter banks satisfying perfect reconstructionMasaaki Ikehara. 137-140 [doi]
- Large vocabulary word recognition based on tree-trellis searchJung-Kuei Chen, Frank K. Soong, Lin-Shan Lee. 137-140 [doi]
- Feature representations for monitoring of tool wearSiva Bala Narayanan, Jing Fang, Gary D. Bernard, Les E. Atlas. 137-140 [doi]
- Extraction of almost periodic signals using cyclostationarityAmod V. Dandawate, Georgios B. Giannakis. 141-144 [doi]
- Search algorithm that merges candidates in meaning level for very large vocabulary spontaneous speech recognitionYasuhiro Minami, Kiyohiro Shikano, Satoshi Takahashi, Tomokazu Yamada. 141-144 [doi]
- Modelling and classification of shapes in two-dimensions using vector quantizationSimon Lee, Brian Lovell. 141-144 [doi]
- Two dimensional nonseparable modulated filter banksShing-Chow Chan. 141-144 [doi]
- A distributed decision approach to speaker verificationMichael A. Lund, Chung-Tshuy Lee, Chung-Chieh Lee, Robert W. Bossemeyer. 141-144 [doi]
- A closed form design method for recursive 3-D cone filtersMichael Bolle. 141-144 [doi]
- Real time implementation of HMM speech recognition for telecommunications applicationsMichael J. Flaherty, Todd Sidney. 145-148 [doi]
- Modeling and optimal compensation of quantization in multidimensional M-band filter bankKyusik Park, Richard A. Haddad. 145-148 [doi]
- Context-free large-vocabulary connected speech recognition with evolutional grammarsMichael K. Brown, Stephen C. Glinski. 145-148 [doi]
- Separation of digital communication signals through joint space-time decorrelationJosep Sala-Alvarez, Gregori Vázquez-Grau. 145-148 [doi]
- A robust, segmental method for text independent speaker identificationHerbert Gish, Michael Schmidt, Angela Mielke. 145-148 [doi]
- Heuristic image decoding using separable source modelsAnthony C. Kam, Gary E. Kopec. 145-148 [doi]
- Modified DFT polyphase SBC filter banks with almost perfect reconstructionNorbert J. Fliege. 149-152 [doi]
- Non-uniform unit parsing for SSS-LR continuous speech recognitionHarald Singer, Jun-ichi Takami, Shoichi Matsunaga. 149-152 [doi]
- Supervised hidden Markov modeling for on-line handwriting recognitionJerome R. Bellegarda, David Nahamoo, Krishna S. Nathan, Eveline J. Bellegarda. 149-152 [doi]
- A statistical model for the simulation of time-varying multipath mobile radio propagation channelBouchra Senadji, A. J. Levy. 149-152 [doi]
- Investigations on speaker characterization from Orphee system techniquesJean-Luc Le Floch, Claude Montacié, Marie-José Caraty. 149-152 [doi]
- Application of cyclostationary and time-frequency signal analysis to car engine diagnosisDetlef König, Johann F. Böhme. 149-152 [doi]
- A generalized lapped orthonormal transform for asymmetrically overlapped windowsBruce W. Suter, Mark E. Oxley. 153-156 [doi]
- Self coupled harmonics: stationary and cyclostationary approachesGuotong Zhou, Georgios B. Giannakis. 153-156 [doi]
- Correlation filters for texture recognition and applications to terrain-delimitation in wide-area surveillanceHemant Singh, Abhijit Mahalanobis. 153-156 [doi]
- Shape estimation and self-noise analysis for towed arraysBrian G. Ferguson. 153-156 [doi]
- An algorithm of high resolution and efficient multiple string hypothesization for continuous speech recognition using inter-word modelsWu Chou, Tatsuo Matsuoka, Biing-Hwang Juang, Chin-Hui Lee. 153-156 [doi]
- A hybrid HMM-MLP speaker verification algorithm for telephone speechJay M. Naik, David M. Lubensky. 153-156 [doi]
- Noise behaviour in the modified Fourier transform for interferometric data inversionStephane Puechmorel, Pierre Thibaut. 157-160 [doi]
- Optimal configuration and weighting of nonuniform arrays according to a maximum ISLR criterionCarlo Boni, Mario Richard, Sergio Barbarossa. 157-160 [doi]
- Adaptive beamforming of cyclic signal and fast implementationQiang Wu, Kon Max Wong. 157-160 [doi]
- Hierarchical pattern classification for high performance text-independent speaker verification systemsJeffrey Sorensen, Michael I. Savic. 157-160 [doi]
- A novel structure for time-varying FIR filter banksIraj Sodagar, Kambiz Nayebi, Thomas P. Barnwell III, Mark J. T. Smith. 157-160 [doi]
- Fast match acoustic models in large vocabulary continuous speech recognitionHarinath Garudadri, Paul Labute, Gilles Boulianne, Patrick Kenny. 157-160 [doi]
- Target shifts due to modeling assumptions in inverse synthetic aperture radarHyeokho Choi, David C. Munson Jr.. 161-164 [doi]
- Location of a dragline bucket in space using machine vision techniquesDavid W. Hainsworth, Peter I. Corke, Graeme J. Winstanley. 161-164 [doi]
- Implementable orthogonal signal projections based on multirate filtersCormac Herley, Nguyen T. Thao. 161-164 [doi]
- Computational balance in real-time cyclic spectral analysisRandy S. Roberts, Herschel H. Loomis Jr.. 161-164 [doi]
- Segmentation of speech using speaker identificationLynn Wilcox, Francine Chen, Don Kimber, Vijay Balasubramanian. 161-164 [doi]
- A channel-bank-based phone detection strategyPonani S. Gopalakrishnan, David Nahamoo, Mukund Padmanabhan, Michael A. Picheny. 161-164 [doi]
- A neural network classifier for cyclostationary signalsBart F. Rice, Scott R. Smith, Richard A. Threlkeld. 165-168 [doi]
- Hybrid system combining expert-TDNNs and HMMs for continuous speech recognitionLaurence Devillers, Christian Dugast. 165-168 [doi]
- Speaker identification using neural tree networksKevin R. Farrell, Richard J. Mammone. 165-168 [doi]
- Comparative study of some algorithms for terrain classification using SAR imagesZiad Belhadj, Ali Saad, Safwen El Assad, Joseph Saillard, Dominique Barba. 165-168 [doi]
- An efficient algorithm for analytic signal generation for time-frequency distributionsAndrew Reilly, Gordon Frazer. 165-168 [doi]
- The processing of HF skywave radar signalsRobert K. Jarrott, Terry A. Soame. 165-168 [doi]
- A generalised model for utilising prosodic information in continuous speech recognitionAndrew Hunt. 169-172 [doi]
- Adaptive signal processing in subbands using sigma-delta modulation techniqueLu Lin, Tyseer Aboulnasr. 169-172 [doi]
- Optimum stuff threshold modulation schemes for digital data transmissionSaman S. Abeysekera. 169-172 [doi]
- Automatic target detection in dynamic clutter from incoherent ISAR dataMehrdad Soumekh, Michael Pollock, Robert Dinger. 169-172 [doi]
- Processing of HF signals scattered by artificial ionospheric turbulence using fractal analysisS. A. Bulgakov, P. V. Ponomarenko, V. G. Sinitsin, Yuri M. Yampolski. 169-172 [doi]
- Detecting an imposter in telephone speechJohan Schalkwyk, Etienne Barnard, Jeffrey R. Sachs. 169-172 [doi]
- Exact analysis of aliasing effects and non-stationary quantization noise in multirate systemsUte Petersohn, Norbert J. Fliege, Horst Unger. 173-176 [doi]
- Automatic classification of prosodically marked phrase boundaries in GermanRalf Kompe, Anton Batliner, Andreas Kießling, Ute Kilian, Heinrich Niemann, Elmar Nöth, Peter Regel-Brietzmann. 173-176 [doi]
- Imaging of multitargets with ISAR based on the time-frequency distributionAiyuan Wang, Yinfang Mao, Zongzhi Chen. 173-176 [doi]
- Low-bit-rate speech coding using a two-dimensional transform of residual signals and waveform interpolationYoshinori Tanaka, Hisanari Kimura. 173-176 [doi]
- Multidimensional wave-digital principles: from filtering to numerical integrationAlfred Fettweis. 173-182 [doi]
- Three receiver structures and their performance analyses for binary signalling in a mixture of Gaussian and α-stable impulsive noisesSachin Ambike, Dimitrios Hatzinakos. 173-176 [doi]
- Demodulation of noisy phase or frequency modulated signals with Kalman filtersOtmar Loffeld. 177-180 [doi]
- Suprasegmental features and continuous speech recognitionPierre Dumouchel. 177-180 [doi]
- Causal FIR matrices with anticausal FIR inverses, and application in characterization of biorthonormal filter banksP. P. Vaidyanathan. 177-180 [doi]
- ISARLAB: a radar signal processing toolBrett Haywood, Ross Kyprianou, Anthony Zyweck. 177-180 [doi]
- Transform trellis coded quantization of speech using small frame sizesVictoria E. Sánchez, José L. Pérez-Córdoba, Juan M. López-Soler, Antonio J. Rubio. 177-180 [doi]
- ∞ filter-a robust EKFGarry A. Einicke, Langford B. White. 181-184 [doi]
- A new methodology for Fourier synthesis-Fourier Interpolation and Reconstruction via Shannon-type Techniques: FIRSTAndré Lannes, Eric Anterrieu, Sylvie Roques, Geraldine Fitoussi. 181-184 [doi]
- An automatic intonation tone contour labelling and classification algorithmAdrian Grigoriu, Julie Vonwiller, Robin W. King. 181-184 [doi]
- On ladder structures and linear phase conditions for bi-orthogonal filter banksImran A. Shah, Ton Kalker. 181-184 [doi]
- High-quality harmonic coding at very low bit ratesGao Yang 0002, Henri Leich. 181-184 [doi]
- Fuzzy logic: issues, contentions and perspectivesLotfi A. Zadeh. 183-184 [doi]
- An overview of multiple-window and quadratic-inverse spectrum estimation methodsDavid J. Thomson. 185-194 [doi]
- Application of hidden Markov models to blind channel characterization and data detectionJosé A. R. Fonollosa, Josep Vidal. 185-188 [doi]
- Prosodic phrase segmentation by pitch pattern clusteringHiroshi Shimodaira, Mitsuru Nakai. 185-188 [doi]
- Non-linear short-term prediction in speech codingJes Thyssen, Henrik Nielsen, Steffen Duus Hansen. 185-188 [doi]
- Synthetic aperture technique used in intravascular imagingJiang Hui, Chao-Huan Hou. 185-188 [doi]
- Filter bank design based on time domain aliasing cancellation with non-identical windowsGreg Smart, Alan B. Bradley. 185-188 [doi]
- Identification techniques for the design of cascade forms perfect-reconstruction two-channel filter banksHervé Le Bihan, Pierre Siohan. 189-192 [doi]
- Estimation of the position of electrocortical generators via subspace techniquesDragan Klimovski, Alex A. Sergejew, Tony L. Cricenti, Greg K. Egan. 189-192 [doi]
- Fine pitch contour extraction by voice fundamental wave filtering methodHiroshi Ohmura. 189-192 [doi]
- Variable bit rate ADPCM via arithmetic codingCraig R. Watkins, Sam Crisafulli, Robert R. Bitmead, Robert Orsi. 189-192 [doi]
- Vector quantization of raw SAR dataJean-Marie Moureaux, Patricia Gauthier, Michel Barlaud, Pascale Bellemain. 189-192 [doi]
- A real time processor for the Australian synthetic aperture radarNick J. S. Stacy, Michael Burgess, J. J. Douglass, Marshall R. Muller, Murray Robinson. 193-196 [doi]
- Dynamic time-warp compensation for correlation of long sequencesRobert Prandolini, Miles Moody. 193-196 [doi]
- High quality coding of wideband audio signals using transform coded excitation (TCX)Roch Lefebvre, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul. 193-196 [doi]
- A single Kalman filtering algorithm for maneuvering target trackingChi-Min Liu, Kuo-Guan Wu, Jer-Heh Sheu. 193-196 [doi]
- Subband-Autocorrelation analysis and its application for speech recognitionShoji Kajita, Fumitada Itakura. 193-196 [doi]
- Generating non-Gaussian random fields for sea surface simulationsGarry N. Newsam, Michael Wegener. 195-198 [doi]
- Pulse train deinterleaving via the Hough transformJane E. Perkins, Ian Coat. 197-200 [doi]
- Robust estimation of mean Doppler frequency for the measurement of average wind velocity in a weather radarJonggil Lee. 197-200 [doi]
- Sign language image processing for intelligent communication by a communications satelliteYoshinao Aoki, Shin Tanahashi, Jun Xu. 197-200 [doi]
- Optimization of time-frequency masking filters using the minimum classification error criterionMichiel Bacchiani, Kiyoaki Aikawa. 197-200 [doi]
- Speech coding based on adaptive mel-cepstral analysisKeiichi Tokuda, Hidetoshi Matsumura, Takao Kobayashi, Satoshi Imai. 197-200 [doi]
- A computationally efficient self-calibrating direction-of-arrival estimatorDean McArthur, Jim P. Reilly. 201-204 [doi]
- A direct digital frequency synthesizer using an IIR filter implemented with a DSP microprocessorLetizia Lo Presti, Giuseppe Cardamone. 201-204 [doi]
- A new paradigm for reliable automatic formant trackingYves Laprie, Marie-Odile Berger. 201-204 [doi]
- A new approach to motion estimation for ISAR imagingStephen Simmons, Robin Evans. 201-204 [doi]
- Spectral entropy: an alternative indicator for rate allocation?Stan A. McClellan, Jerry D. Gibson. 201-204 [doi]
- Recursive tracking of formants in speech signalsMahesan Niranjan, Ingemar J. Cox, Sunita L. Hingorani. 205-208 [doi]
- Mixed-phase AR models for voiced speech and perceptual cost functionsWilliam R. Gardner, Bhaskar D. Rao. 205-208 [doi]
- Wavelet applications to structural analysisRussell D. Priebe, Gary R. Wilson. 205-208 [doi]
- Motion compensated video sequence interpolation using digital image warpingJacek Nieweglowski, Timo Moisala, Petri Haavisto. 205-208 [doi]
- Unknown signal wavelength and array processingAntony P.-C. Ng, Gim Pew Quek. 205-208 [doi]
- Low rate audio coder with hierarchical filterbanks and lattice vector quantizationPeter Monta, Shiufun Cheung. 209-212 [doi]
- Alternatives for the definition and evaluation of resolution thresholds of signal-subspace parameter estimatorsWenyuan Xu, Mostafa Kaveh. 209-212 [doi]
- Motion estimation using multiple image sensorsKiyoharu Aizawa, Kenichi Iwata, Takahiro Saito, Mitsutoshi Hatori. 209-212 [doi]
- A technique for transient suppression of IIR notch filterSoo-Chang Pei, Chien-Cheng Tseng. 209-212 [doi]
- Speech compression using ARMA model and wavelet transformSun-Won Park. 209-212 [doi]
- SLHMM: a continuous speech recognition system based on Alphanet-HMMJesús E. Díaz-Verdejo, José C. Segura, Pedro García Teodoro, Antonio J. Rubio. 213-216 [doi]
- Performance analysis of MUSIC and Pencil-MUSIC algorithms for diversely-polarized arrayChi Cheng, Yingbo Hua. 213-216 [doi]
- A two-channel approach to the removal of impulsive noise from archived recordingsChristopher M. Hicks, Simon J. Godsill. 213-216 [doi]
- Spanning the gap between motion estimation and morphingMichele Covell, Margaret Withgott. 213-216 [doi]
- Echo cancellation with reference signal generator and reliable receiving schemes for intersymbol interferenceHsiang-Feng Chi, Ja-Ling Wu. 213-216 [doi]
- Improved musical pitch tracking using principal decomposition analysisErkan Dorken, S. Hamid Nawab. 217-220 [doi]
- Optimal localization of partially known signals in unknown noise fieldsPetre Stoica, Mats Viberg, Björn E. Ottersten, Thomas Kailath. 217-220 [doi]
- Learning state-dependent stream weights for multi-codebook HMM speech recognition systemsIvica Rogina, Alex Waibel. 217-220 [doi]
- A new technique for block-based motion compensationShinichi Kozu, Sanjeev R. Kulkarni. 217-220 [doi]
- A noise cancelling filter for the digital Holter monitoring systemTakashi Kohama, Shogo Nakamura, Hiroshi Hoshino. 217-220 [doi]
- Arrival angle estimation in non-Gaussian noiseDebasis Sengupta, Sarbani Palit. 221-224 [doi]
- Bayesian learning of the SCHMM parameters for speech recognitionQiang Huo, Chorkin Chan, Chin-Hui Lee. 221-224 [doi]
- Real-time echo cancellation using a new fast LMS-based algorithmMohammed A. Khasawneh, Tariq Haddad. 221-224 [doi]
- An algorithm for simultaneous motion estimation and scene segmentationMichael M. Chang, M. Ibrahim Sezan, A. Murat Tekalp. 221-224 [doi]
- Real-time, loose-harmonic matching fundamental frequency estimation for musical signalsFrancisco Javier Casajús-Quirós, Pablo Fernandez-Cid. 221-224 [doi]
- Further study of the SDMP method for parameter estimation of multiple transientsJames B. Evans, Yingbo Hua. 225-228 [doi]
- Adaptive DOA tracking by rank-revealing QR updating and exponential sliding windows techniquesJeng-Kuang Hwang, Shun-Tai Wei. 225-228 [doi]
- Real time speech rate converting system for elderly peopleAkira Nakamura, Nobumasa Seiyama, Ryou Ikezawa, Tohru Takagi, Eiichi Miyasaka. 225-228 [doi]
- Non-linear input transformations for discriminative HMMsFinn Tore Johansen, Magne Hallstein Johnsen. 225-228 [doi]
- Digital video standards conversion in the presence of accelerated motionAndrew J. Patti, M. Ibrahim Sezan, A. Murat Tekalp. 225-228 [doi]
- Using MAP estimated parameters to improve HMM speech recognition performanceYoshihiko Gotoh, Michael M. Hochberg, Harvey F. Silverman. 229-232 [doi]
- Nonuniform image motion estimation using Kalman filteringNader M. Namazi, Pablo B. Penafiel, Chieh-Min Fan. 229-232 [doi]
- Automatic model parameter generation for the speech training of deaf childrenHector R. Javkin, Elizabeth Keate, Norma Antonanzas-Barroso, Yoshinori Yamada, Karen Youdelman. 229-232 [doi]
- Unitary ESPRIT: how to exploit additional information inherent in the relational invariance structureMartin Haardt, Markus E. Ali-Hackl. 229-232 [doi]
- Parametric technique for P wave delineationB. Madhukar, K. Rajgopal, Lalit M. Patnaik. 229-232 [doi]
- High-resolution direction of arrival estimation using minimum-norm method without eigendecompositionArnab K. Shaw, Wei Xia. 233-236 [doi]
- Discriminative training for improved neural prediction systemsAbdelhamid Mellouk, Patrick Gallinari. 233-236 [doi]
- Blind channel estimation and equalization with partial-response input signalsUma Gummadavelli, Jitendra K. Tugnait. 233-236 [doi]
- Intensity scale invariant motion estimation with rotation and spatial scaling informationColin Bussiere, Dimitrios Hatzinakos. 233-236 [doi]
- Recursive restoration of pitch variation defects in musical recordingsSimon J. Godsill. 233-236 [doi]
- Restoration of low bit rate compressed images using mean field annealingJames C. Brailean, Taner Özcelik, Aggelos K. Katsaggelos. 237-240 [doi]
- An evaluation of cross-language adaptation for rapid HMM development in a new languageBarbara Wheatley, Kazuhiro Kondo, Wallace W. Anderson, Yeshwant K. Muthusamy. 237-240 [doi]
- Detection of human speech in structured noiseJohn D. Hoyt, Harry Wechsler. 237-240 [doi]
- Implementation of a parallel DFE using residue number systemStephen Oh, Domingo Garcia. 237-240 [doi]
- Performances analysis of the propagator method for source bearing estimationSylvie Marcos, Alain Marsal, Messaoud Benidir. 237-240 [doi]
- Array pattern estimation from amplitude measurements on arbitrary near-field surfacesGeoff Poulton. 241-244 [doi]
- Automatic training of phoneme dictionary based on mutual information criterionShigeki Okawa, Tetsunori Kobayashi, Katsuhiko Shirai. 241-244 [doi]
- Active control using IIR filters-a second lookScott D. Snyder. 241-244 [doi]
- Adaptive multi-feature motion estimationRegis J. Crinon, Wojciech J. Kolodziej. 241-244 [doi]
- Computational and performance analysis of Radon transform based constellation identificationSally L. Wood, John R. Treichler. 241-244 [doi]
- Tree-structured speaker clustering for fast speaker adaptationTetsuo Kosaka, Shigeki Sagayama. 245-248 [doi]
- DSP implementation and performance evaluation of a compact stereo echo cancellerAkihiro Hirano, Akihiko Sugiyama, Yasuhiro Arasawa, Noboru Kawayachi. 245-248 [doi]
- Replacement noise in image sequences-detection and interpolation by motion field segmentationRobin D. Morris, William J. Fitzgerald. 245-248 [doi]
- On the performance of multicarrier modulation in a broadcast multipath environmentJulien J. Nicolas, Jae S. Lim. 245-248 [doi]
- Low rank detection of multichannel Gaussian signals using a constrained inversePeter Strobach. 245-248 [doi]
- Linearly-constrained adaptive beamforming using cyclostationary signal propertiesLuis Castedo, Ching-Yih Tseng, Aníbal R. Figueiras-Vidal, Lloyd J. Griffiths. 249-252 [doi]
- Multi band sigma delta analog to digital conversionPervez M. Aziz, Henrik V. Sorensen, Jan Van der Spiegel. 249-252 [doi]
- A time varying step size normalized LMS echo canceler algorithmHéctor M. Pérez Meana, Luís Nino de Rivera, Mariko Nakano-Miyatake, Fausto Casco-Sanchez, Juan Carlos Sanchez-Garcia. 249-252 [doi]
- All-phoneme ergodic hidden Markov network for unsupervised speaker adaptationYasunaga Miyazawa, Jun-ichi Takami, Shigeki Sagayama, Shoichi Matsunaga. 249-252 [doi]
- Minimum generalised quadratic error quantisation for image and video codingJohn Princen, Ming H. Chan. 249-252 [doi]
- Entropy-constrained predictive trellis coded quantization: application to hyperspectral image compressionGlen P. Abousleman, Michael W. Marcellin, Bobby R. Hunt. 253-256 [doi]
- Fully adaptive generalized recursive control system for active acoustic attenuationLarry J. Eriksson, Mark C. Allie, Douglas E. Melton, Steven R. Popovich, Trevor A. Laak. 253-256 [doi]
- Phoneme recognition improvement by restricting training section in concatenated HMM trainingYasuo Ariki, Keisuke Doi. 253-256 [doi]
- A DSP-based research prototype reverse channel transmitter/receiver for ADSLAlbert M. Gottlieb. 253-256 [doi]
- A subspace GLRT for vector-sensor array detectionKeith A. Burgess, Barry D. Van Veen. 253-256 [doi]
- Array signal number detection for coherent and incoherent signals in unknown noise environmentsQiang Wu, Kon Max Wong. 257-260 [doi]
- Image coding using adaptive recursive interpolative DPCM with entropy-constrained trellis coded quantizationEric A. Gifford, Bobby R. Hunt, Michael W. Marcellin. 257-260 [doi]
- Characterization of talker radiation pattern using a microphone arrayPaul C. Meuse, Harvey F. Silverman. 257-260 [doi]
- A RLS bilinear filter for channel equalizationGin-Kou Ma, Junghsi Lee, V. John Mathews. 257-260 [doi]
- A codec candidate for the GSM half rate speech channelJörg-Martin Müller, Bertram Wächter. 257-260 [doi]
- Deterministic annealing for trellis quantizer and HMM design using Baum-Welch re-estimationDavid J. Miller 0001, Kenneth Rose, Philip A. Chou. 261-264 [doi]
- Complexity reduction for FS-1016 with multistage searchMichel Mauc, Geneviève Baudoin, Milan Jelinek. 261-264 [doi]
- Source number estimator using Gerschgorin disksHsien-Tsai Wu, Jar-Ferr Yang, Fwu-Kuen Chen. 261-264 [doi]
- Effects of room reverberation on time-delay estimation performanceStéphane Bédard, Benoît Champagne, Alex Stephenne. 261-264 [doi]
- Use of the DFT for synchronization in packetized data communicationsSelim U. Zaman, K. Warren Yates. 261-264 [doi]
- New adaptive-filtering techniques applied to speech echo cancellationMarcio G. Siqueira, Abeer Alwan. 265-268 [doi]
- MMSE design of polyphase components for generalized interpolatorsLuc Vandendorpe, Paul Delogne, Benoît Maison, Laurent Cuvelier. 265-268 [doi]
- Speech and channel codec candidate for the half rate digital cellular channelKumar Swaminathan, Kalyan Ganesan, Yi-Sheng Wang, Prabhat K. Gupta. 265-268 [doi]
- Improved signal copy with partially known or unknown array responseJiankan Yang, Soumendra Daas, A. Lee Swindlehurst. 265-268 [doi]
- Optimal entropy constrained scalar quantization for exponential and Laplacian random variablesGary J. Sullivan. 265-268 [doi]
- Colouration in assisted reverberation systemsMark A. Poletti. 269-272 [doi]
- Robust adaptive beamforming via LMS-like target trackingSofiène Affes, Saeed Gazor, Yves Grenier. 269-272 [doi]
- A method for non-parametric waveform estimation based on filter banksFarook Sattar, Göran Salomonsson. 269-272 [doi]
- M-LCELP speech coding at 4 kbpsKazunori Ozawa, Masahiro Serizawa, Toshiki Miyano, Toshiyuki Nomura. 269-272 [doi]
- Lattice vector quantization of image wavelet coefficient vectors using a simplified form of entropy codingAndrew Woolf, Glynn Rogers. 269-272 [doi]
- Acoustic event localization using a crosspower-spectrum phase based techniqueMaurizio Omologo, Piergiorgio Svaizer. 273-276 [doi]
- A new sensor array signal processing techniqueAlan J. Coulson, Rodney G. Vaughan. 273-276 [doi]
- A new composite criterion for adaptive and iterative blind source separationJean-François Cardoso, Adel Belouchrani, Beate H. Laheld. 273-276 [doi]
- Object-oriented video coding employing dense motion fieldsChristoph Stiller. 273-276 [doi]
- Comparison of ARMA modelling methods for low bit rate speech codingSusan Yim, Dipanjan Sen, W. Harvey Holmes. 273-276 [doi]
- Asymptotic performance of second order blind separationKarim Abed-Meraim, Adel Belouchrani, Jean-François Cardoso, Eric Moulines. 277-280 [doi]
- Robust signal extrapolation using waveletsLi-Chien Lin, C. C. Jay Kuo. 277-280 [doi]
- 3D contour image coding based on morphological filters and motion estimationChuang Gu. 277-280 [doi]
- A 5.6 kb/s speech codec using a pulse codebook and improved Viterbi decodingYoshiaki Asakawa, Hidetoshi Sekine, Makoto Takashima, Nobuyoshi Ishikawa, Toshiyuki Matsuda, Tom Okamoto, Ryujiro Muramatsu. 277-280 [doi]
- Parametric separation applied to underwater acoustic multipath propagationAbderrahman Essebbar. 277-280 [doi]
- Far field array processing with neural networksBrigitte Colnet, Jean-Paul Haton. 281-284 [doi]
- The generalized Backus-Gilbert inversion method for signal recovery in multiresolution spacesXiang-Gen Xia, C. C. Jay Kuo, Zhen Zhang. 281-284 [doi]
- A VR-CELP codec implementation for CDMA mobile communicationsLuca Cellario, Daniele Sereno, Mario Giani, Peter Blöcher, Karl Hellwig. 281-284 [doi]
- Adaptive subspace selection using subband decompositions for sensor array processingJ. Scott Goldstein, Mary Ann Ingram, E. Jeff Holder, Richard N. Smith. 281-284 [doi]
- An analog interpretation of compression for digital communication systemsJohn M. Lervik, Tor A. Ramstad. 281-284 [doi]
- Adaptive waveform selection for active sonarDavid J. Kershaw, Robin J. Evans. 285-288 [doi]
- Reed-Solomon coding for CELP EDAC in land mobile radioDavid E. Ray, Douglas J. Rahikka. 285-288 [doi]
- A fast algorithm for region-oriented texture codingMarco Cermelli, Fabio Lavagetto, Matteo Pampolini. 285-288 [doi]
- Sampling rate conversion using fractional-sample delayAndrzej Tarczynski, Wojciech Kozinski, Gerald D. Cain. 285-288 [doi]
- Evolutionary maximum entropy spectral analysisSyed I. Shah, Luis F. Chaparro, A. Salim Kayhan. 285-288 [doi]
- The transitory evolutionary spectrumC. S. Detka, Amro El-Jaroudi. 289-292 [doi]
- Address predictive color quantization image compression for multimedia applicationsLai-Man Po, Wen-Tao Tan, Chi-Ho Chan. 289-292 [doi]
- Least squares signal reconstruction under normalized autocorrelation constraintsOrhan Arikan, Bülent Baygün. 289-292 [doi]
- A comparison between spectral and bispectral analysis for ship detection from acoustical time seriesCarlo S. Regazzoni, Alessandra Tesei, Giorgio Tacconi. 289-292 [doi]
- Analysis of phoneme-based features for language identificationKay M. Berkling, Takayuki Arai, Etienne Barnard. 289-292 [doi]
- Language identification using phone-based acoustic likelihoodsLori Lamel, Jean-Luc Gauvain. 293-296 [doi]
- On the error concealment technique for DCT based image codingJong-Wook Park, Dong Sik Kim, Sang Uk Lee. 293-296 [doi]
- The use of non-supervised neural networks to detect lines in lofargramJean-Claude Di Martino, Brigitte Colnet, Marc Di Martino. 293-296 [doi]
- Multiple window spectrogram and time-frequency distributionsGordon Frazer, Boualem Boashash. 293-296 [doi]
- Genetic algorithms for neuromagnetic source reconstructionPaul S. Lewis, John C. Mosher. 293-296 [doi]
- Channel equalization with perceptrons: an information-theoretic approachTülay Adali, M. Kemal Sönmez. 297-300 [doi]
- Automatic language identification using syllabic spectral featuresKung-Pu Li. 297-300 [doi]
- Optimal kernels for Wigner-Ville spectral estimationAkbar M. Sayeed, Douglas L. Jones. 297-300 [doi]
- Image reconstruction from contour data using a back-propogation neural networkKarim Faez, Mohamed Kamel. 297-300 [doi]
- Adaptive detection of moving signal using shallow sea hydroacoustic dataAlex B. Gershman, Vitaly A. Zverev. 297-300 [doi]
- A new positive time-frequency distributionJavier Rodríguez Fonollosa, Chrysostomos L. Nikias. 301-304 [doi]
- Sampling of two-dimensional signals below Nyquist density with application to computer aided tomographyKai-Kou R. Yu, Sze-Fong Yau. 301-304 [doi]
- Inverse filtering of room impulse response for binaural recording playback through loudspeakersChristian Bourget, Tyseer Aboulnasr. 301-304 [doi]
- Automatic language identification using sub-word modelsRoger C. F. Tucker, Michael J. Carey 0002, Eluned S. Parris. 301-304 [doi]
- GASTOM 90 tomography experiment data inversionYann Stéphan, François-Regis Martin-Lauzer. 301-304 [doi]
- A fast tomographic reconstruction algorithm in the 2-D wavelet transform domainLaure Blanc-Féraud, Pierre Charbonnier, Pierre Lobel, Michel Barlaud. 305-308 [doi]
- Probabilistic ray identification: a new tool for ocean acoustic tomographyFrançois-Regis Martin-Lauzer, Didier Mauuary, Yann Stéphan. 305-308 [doi]
- Adaptive HMM filters for signals in noisy fading channelsIain B. Collings, John B. Moore. 305-308 [doi]
- Automatic language identification of telephone speech messages using phoneme recognition and N-gram modelingMarc A. Zissman, Elliot Singer. 305-308 [doi]
- Non-orthogonal Gabor representation of biological signalsMark L. Brown, William J. Williams, Alfred O. Hero III. 305-308 [doi]
- Quantisation error modelling of narrowband adaptive arrays using projected perturbation sequencesSteven Ivandich. 309-312 [doi]
- Distance measures for text-independent speaker recognition based on MAR modelChintana Griffin, Tomoko Matsui, Sakaoki Furui. 309-312 [doi]
- Time-frequency kernel design via point and derivative constraintsMoeness G. Amin, James F. Carroll. 309-312 [doi]
- Functional-link models for adaptive channel equaliserWoon-Seng Gan, John J. Soraghan, Tariq S. Durrani. 309-312 [doi]
- Tomographic reconstruction of time-varying object from linear time-sequential sampled projectionsYing Ha Chiu, Sze-Fong Yau. 309-312 [doi]
- Effects of input data correlation on the convergence of blind adaptive equalizersJames P. LeBlanc, Kutluyil Dogancay, Rodney A. Kennedy, C. Richard Johnson Jr.. 313-316 [doi]
- An optimal window length for the PWVD with application to passive acoustic parameter estimationDavid C. Reid, Jonathon C. Ralston. 313-316 [doi]
- The reconstruction of subsurface property maps using projection onto convex setsAlberto Malinverno, David J. Rossi, Michael M. Daniel. 313-316 [doi]
- Discriminating semi-continuous HMM for speaker verificationMark E. Forsyth, Mervyn A. Jack. 313-316 [doi]
- An integrated approach to passive target classificationR. Rajagopal, K. Anoop Kumar, P. Ramakrishna Rao. 313-316 [doi]
- Maneuvering target motion analysis using hidden Markov modelOlivier Trémois, Jean-Pierre Le Cadre. 317-320 [doi]
- A model distance measure for talker clustering and identificationJonathan Foote, Harvey F. Silverman. 317-320 [doi]
- Simultaneous confidence intervals for image reconstruction problemsYong Zhang, Alfred O. Hero III, W. Leslie Rogers. 317-320 [doi]
- Generalization of the reassignment method to all bilinear time-frequency and time-scale representationsFrançois Auger, Patrick Flandrin. 317-320 [doi]
- A blind spatio-temporal equalizer for a radio-mobile channel using the constant modulus algorithm (CMA)Sylvie Mayrargue. 317-320 [doi]
- Improved voice identification using a nearest-neighbor distance measureLawrence G. Bahler, Jack E. Porter, Alan L. Higgins. 321-324 [doi]
- Optimum Costas-like decompositions of Costas arrays for channel characterization and communicationsSanjay K. Mehta, Edward L. Titlebaum. 321-324 [doi]
- Iterative reconstruction of multidimensional objects buried in inhomogeneous elastic mediaTarek M. Habashy, Eveline J. Bellegarda. 321-324 [doi]
- Periodic uncertainty in periodic spectral analysis of processes associated with periodic phenomenaLangford B. White, Peter J. Sherman. 321-324 [doi]
- New adaptive blind equalization algorithms for constant modulus constellationsConstantinos B. Papadias, Dirk T. M. Slock. 321-324 [doi]
- Speaker recognition based on minimum error discriminative trainingChi-shi Liu, Chin-Hui Lee, Biing-Hwang Juang, Aaron E. Rosenberg. 325-328 [doi]
- Time-varying higher-order cumulant spectra: application to the analysis of composite FM signals in multiplicative and additive noiseBoualem Boashash, Branko Ristic. 325-328 [doi]
- On globally convergent blind equalization for QAM systemsKen Yamazaki, Rodney A. Kennedy. 325-328 [doi]
- Transient sonar signal classification using hidden Markov model and neural netAmlan Kundu, George C. Chen, Charles E. Persons. 325-328 [doi]
- Discrete multichannel orthogonal transformsIoannis Pitas, Anestis Karasaridis. 325-328 [doi]
- Statistics of the phase delays between array receivers estimated from the eigendecomposition of the signal correlation matrixJustin J. Smith, Yee Hong Leung, Antonio Cantoni. 329-332 [doi]
- Time-frequency complexity and informationPatrick Flandrin, Richard G. Baraniuk, Olivier J. J. Michel. 329-332 [doi]
- An edge classification based approach to the post-processing of transform encoded imagesJohn D. McDonnell, Robert N. Shorten, Anthony D. Fagan. 329-332 [doi]
- Speaker recognition in tactical communicationsRichard Ricart, Jim Cupples, Laurie Fenstermacher. 329-332 [doi]
- Instantaneous spectral estimation of nonstationary signalsYumi Takizawa, Keisuke Oda, Atsushi Fukasawa. 329-332 [doi]
- Perceptual benchmarks for automatic language identificationYeshwant K. Muthusamy, Neena Jain, Ronald A. Cole. 333-336 [doi]
- Block predictive transform coding of still imagesJianzhong Huang, Sam Liu. 333-336 [doi]
- A simple calculation of the joint moments of hidden Markov modelsMehmet Karan, Brian D. O. Anderson, Robert C. Williamson. 333-336 [doi]
- Extraction of pertinent parameters in oceanographic dataMichel Bouvet, Laurent Kerleguer, Philippe Mennecier, Michel Cresp. 333-336 [doi]
- Kernel design techniques for alias-free time-frequency distributionsJohn R. O'Hair, Bruce W. Suter. 333-336 [doi]
- Demonstrations and applications of spoken language technology: highlights and perspectives from the 1993 ARPA Spoken Language Technology and Applications DayClifford J. Weinstein. 337-340 [doi]
- Image coding with discrete cosine transforms using efficient energy-based adaptive zonal filteringAlessandro M. Palau, Gagan Mirchandani. 337-340 [doi]
- Adaptive estimation of hidden nearly completely decomposable Markov chainsVikram Krishnamurthy. 337-340 [doi]
- Matched field processing in shallow ocean: signal arrival identification using EM algorithmChristoph F. Mecklenbräuker, Johann F. Böhme. 337-340 [doi]
- Analysis of multicomponent signals by multilinear time-frequency representationsSergio Barbarossa, Giuseppe Schiappa. 337-340 [doi]
- Adaptive kernel design in the generalized marginals domain for time-frequency analysisSrivathsan Krishnamachari, William J. Williams. 341-344 [doi]
- The estimation and HMM tracking of weak narrowband signalsBarry G. Quinn, Ross F. Barrett, Stephen J. Searle. 341-344 [doi]
- Unanswerable queries in a spontaneous speech taskSunil Issar, Wayne Ward. 341-344 [doi]
- Bayesian time delay estimation for ocean acoustic tomographyDidier Mauuary, Geneviève Jourdain. 341-344 [doi]
- A new method for block effect removal in low bit-rate image compressionJiebo Luo, Chang Wen Chen, Kevin J. Parker, Thomas S. Huang. 341-344 [doi]
- Resolving ambiguities in estimating spatial frequencies in sparse linear arrayDonald W. Tufts, Hongya Ge, Ramdas Kumaresan. 345-348 [doi]
- Filtering real signals through frequency modulation and peak detection in the time-frequency planeMorgan J. Arnold, Mark Roessgen, Boualem Boashash. 345-348 [doi]
- An efficient Bayes solution to AR signal modelling for short sequencesDouglas E. Johnston, Petar M. Djuric. 345-348 [doi]
- Fast segmented image coding using weakly separable basesWilfried Philips, Charilaos A. Christopoulos. 345-348 [doi]
- JANUS 93: towards spontaneous speech translationMonika Woszczyna, Naomi Aoki-Waibel, Finn Dag Buø, Noah Coccaro, Keiko Horiguchi, Thomas Kemp, Alon Lavie, Arthur E. McNair, Thomas Polzin, Ivica Rogina, Carolyn Penstein Rosé, Tanja Schultz, Bernhard Suhm, Masaru Tomita, Alex Waibel. 345-348 [doi]
- Robust recursive estimation for linear systems with non-Gaussian state and measurement noisesKi Yong Lee, Byung-Gook Lee, SouGuil Ann, Iickho Song. 349-352 [doi]
- Correcting complex false starts in spontaneous speechDouglas D. O'Shaughnessy. 349-352 [doi]
- Warped linear prediction (WLP) in speech and audio processingUnto K. Laine, Matti Karjalainen, Toomas Altosaar. 349-352 [doi]
- Estimation of aliasing error in layered coding systemMasahiro Iwahashi, Koichi Ohyama, Noriyoshi Kambayashi. 349-352 [doi]
- A deconvolution approach to moving sources localizationHerve Chuberre, Jean-Jacques Fuchs. 349-352 [doi]
- Detection of nonstationary random signals in colored noiseWayne T. Padgett, Douglas B. Williams. 353-356 [doi]
- Subband coding of color images with limited palette sizePatrick Waldemar, Tor Audun Ramstad. 353-356 [doi]
- Reducing the computational complexity for inferring stochastic context-free grammar rules from example textHelmut Lucke. 353-356 [doi]
- Non-recursive FM demodulation of laser radar backscatter using time-frequency distributionsAshruf S. El-Dinary, Timothy D. Cole. 353-356 [doi]
- High-resolution bathymetric simulations based on Kirchhoff scattering theory and anisotropic seafloor modelingDimitris Pantzartzis, Dimitri Alexandrou, Vincent Premus. 353-356 [doi]
- Geolocalization by combined range difference and range rate difference measurementsGeorge Henry Niezgoda, K. C. Ho. 357-360 [doi]
- Robust approximate likelihood ratio tests for nonlinear dynamic systemsLangford B. White. 357-360 [doi]
- Ergodic hidden Markov models and polygrams for language modelingThomas Kuhn, Heinrich Niemann, Ernst Günter Schukat-Talamazzini. 357-360 [doi]
- Inversion of large-support ill-conditioned linear operators using a Markov model with a line processMila Nikolova, Ali Mohammad-Djafari, Jérôme Idier. 357-360 [doi]
- Beyond time-frequency analysis: energy densities in one and many dimensionsRichard G. Baraniuk. 357-360 [doi]
- Analysis of an echo canceller with an adaptive FIR filter using the Sign algorithm for Gaussian transmit and receive signalsShin'ichi Koike. 361-364 [doi]
- Extraction of multiple periodic waveforms from noisy dataPartha Pratim Kanjilal, Sarbani Palit. 361-364 [doi]
- A robust language model incorporating a substring parser and extended n-gramsJerry H. Wright, Gareth J. F. Jones, Harvey Lloyd-Thomas. 361-364 [doi]
- Detection of weak stochastic signals in non-Gaussian noise: a general resultPeter M. Schultheiss, Lal C. Godara. 361-364 [doi]
- Estimation of qth-order fractal dimensionsDaniele D. Giusto, Stefano Fioravanti. 361-364 [doi]
- An interactive solution to adaptive phase jitter cancellationRamin A. Nobakht. 365-368 [doi]
- Statistical signal processing in broadband reflectometryFernando D. Nunes, José M. N. Leitão. 365-368 [doi]
- Learning complex output representations in connectionist parsing of spoken languageFinn Dag Buø, Thomas Polzin, Alex Waibel. 365-368 [doi]
- Deconvolution of sensor array signals using time-scalingTayfun Akgül, Amro El-Jaroudi, Marwan A. Simaan. 365-368 [doi]
- On the distribution of the DCT coefficientsThierry Eude, Richard Grisel, Hocine Cherifi, Roland Debrie. 365-368 [doi]
- Echo cancellation with the gamma filterMahlar Palkar, Jose C. Principe. 369-372 [doi]
- Self-similarity modeling for interpolation and extrapolation of multi-viewpoint image setsTakeshi Naemura, Hiroshi Harashima. 369-372 [doi]
- A frequency domain filtering method for generation of long complex Gaussian sequences with required spectraWeimin Zhang. 369-372 [doi]
- Correction of I/Q errors in homodyne step frequency radar refocuses range profilesDavid A. Noon, Dennis Longstaff, Glen F. Stickley. 369-372 [doi]
- Sentence spotting applied to partial sentences and unknown wordsYoshiaki Itoh 0001, Jiro Kiyama, Ryuichi Oka. 369-372 [doi]
- Adaptation of memory depth in the gamma filterJyh-Ming Kuo, Samel Çelebi. 373-376 [doi]
- Optimizing recognition and rejection performance in wordspotting systemsHervé Bourlard, Bart D'hoore, Jean-Marc Boite. 373-376 [doi]
- Stochastic modeling and estimation of multispectral image dataRichard R. Schultz, Robert L. Stevenson. 373-376 [doi]
- Radar target recognition using range profilesAnthony Zyweck, Robert E. Bogner. 373-376 [doi]
- A new RLS algorithm based on the variation characteristics of a room impulse responseShoji Makino, Yutaka Kaneda. 373-376 [doi]
- A fast lattice-based approach to vocabulary independent wordspottingDavid A. James, Steve J. Young. 377-380 [doi]
- Approximate covariance functions for gray-level Gibbs random fieldsIbrahim M. Elfadel. 377-380 [doi]
- The WRLS algorithm for speech processingKyung Y. Yoo, Nancy Hubing. 377-380 [doi]
- Parametrization of stable systems from impulse response dataS. Unnikrishna Pillai, Won-Cheol Lee. 377-380 [doi]
- Weak radar signal detection based on wavelet transformNaoki Ehara, Iwao Sasase, Shinsaku Mori. 377-380 [doi]
- Cluster-based probability model applied to image restoration and compressionKris Popat, Rosalind W. Picard. 381-384 [doi]
- Coherence analysis of multichannel time series applying conditioned multivariate autoregressive spectraHeli Väätäjä, Risto Suoranta, Seppo Rantala. 381-384 [doi]
- Spotting events in continuous speechPhilippe Jeanrenaud, Man-Hung Siu, Jan Robin Rohlicek, Marie Meteer, Herbert Gish. 381-384 [doi]
- Detection, identification and tracking of active scatterersZouak Mouhcine, Joseph Saillard. 381-384 [doi]
- A robust FLS algorithm for linearly-constrained adaptive filteringLeonardo S. Resende, João Marcos Travassos Romano, Maurice G. Bellanger. 381-384 [doi]
- On the equivalence between Gamma and Laguerre filtersTomas Oliveira e Silva. 385-388 [doi]
- A very robust, fast, parallelizable adaptive least squares algorithm with excellent tracking abilitiesConstantin Papaodysseus, D. Gorgoyannis, Elias Koukoutsis, Panayiotis Rousopoulos. 385-388 [doi]
- Approaches to topic identification on the switchboard corpusJohn W. McDonough, Kenney Ng, Philippe Jeanrenaud, Herbert Gish, Jan Robin Rohlicek. 385-388 [doi]
- Covariance matrix matching for multi-spectral image classificationPaul J. Whitbread. 385-388 [doi]
- Estimation of the causal impulse response of underwater targetPhilippe Delachartre, Didier Vray, ZhiGang Sun, Gérard Gimenez, Albin Dziedzic. 385-388 [doi]
- Wordspotter training using figure-of-merit back propagationRichard Lippmann, Eric I. Chang, Charles R. Jankowski Jr.. 389-392 [doi]
- An adaptive linear multiuser receiver for deep water acoustic local area networksZoran Zvonar, David Brady, Josko Catipovic. 389-392 [doi]
- Tracking subspace representations of face imagesHsi-Jung Wu, Dulce B. Ponceleon, Katherine S. Wang, James O. Normile. 389-392 [doi]
- A vectorized systolic array for block constrained RLSHideaki Sakai, Manabu Kuroda. 389-392 [doi]
- A criterion for the separation of a linear-quadratic mixture of independent componentsMichel Krob, Messaoud Benidir. 389-392 [doi]
- A fault tolerant FIR adaptive filter based on the FFTBernard A. Schnaufer, W. Kenneth Jenkins. 393-396 [doi]
- An efficient closed-form localization solution from time difference of arrival measurementsYiu-Tong Chan, K. C. Ho. 393-396 [doi]
- Bayesian model order selection for the Karhunen-Loeve transform and the singular value decompositionJebu J. Rajan, Peter J. W. Rayner. 393-396 [doi]
- Texture class assignment in texscale: an evaluation studyJane You, Harvey A. Cohen. 393-396 [doi]
- Rejection for connected digit recognition based on GPD segmental discriminationRafid A. Sukkar. 393-396 [doi]
- The performance prediction method on sentence recognition system using a finite state automatonTakashi Otsuki, Akinori Ito, Shozo Makino, Teruhiko Otomo. 397-400 [doi]
- Least-squares lattice interpolation filtersJenq-Tay Yuan. 397-400 [doi]
- Pseudo roots and the effects of model order overestimation on the ESPRIT algorithmSeth D. Silverstein. 397-400 [doi]
- Time delay estimation using integrated polyspectrumYisong Ye, Jitendra K. Tugnait. 397-400 [doi]
- A CAD driven multiscale approach to automated inspectionDaniel Tretter, Khalid W. Khawaja, Charles A. Bouman, Anthony A. Maciejewski. 397-400 [doi]
- New ways to use LVQ-codebooks together with hidden Markov modelsKari Torkkola. 401-404 [doi]
- A VLSI architecture for real-time hierarchical encoding/decoding of video using the wavelet transformMohan Vishwanath, Chaitali Chakrabarti. 401-404 [doi]
- Blind maximum likelihood sequence estimation for fading channelsDavid J. Reader, Jason B. Scholz. 401-404 [doi]
- Adaptive post-processing algorithms for low bit rate video signalsTsann-Shyong Liu, Nikil Jayant. 401-404 [doi]
- Almost-sure convergence of the non-homogeneous DNLMS algorithm with decreasing step sizeSang-Sik Ahn, Peter J. Voltz. 401-404 [doi]
- An efficient implementation of a nonlinear predictor using a zero-memory nonlinearity followed by a second order Volterra filterM. Anisur Rahman, Kai-Bor Yu. 405-408 [doi]
- 3D subband coder for very low bit ratesWeng Leong Chooi, King Ngi Ngan. 405-408 [doi]
- Parallel distributed binary mapping models for speech recognitionJianmin Li, Ditang Fang. 405-408 [doi]
- The curse of dimension on the learning rate of the LMS adaptive FIR filterJohn Homer. 405-408 [doi]
- Digit pipelined discrete wavelet transformChetana Nagendra, Mary Jane Irwin, Robert Michael Owens. 405-408 [doi]
- Continuous speech recognition in noise using spectral subtraction and HMM adaptationJuan Arturo Nolazco-Flores, Steve J. Young. 409-412 [doi]
- A low power subband video decoder architectureBenjamin M. Gordon, Teresa H. Y. Meng. 409-412 [doi]
- Generalized URV subspace tracking LMS algorithmSrinath Hosur, Ahmed H. Tewfik, Daniel Boley. 409-412 [doi]
- Recursive Bayes risk parameter estimation from the cyclic autocorrelation matrixJaume Riba-Sagarra, Gregori Vázquez-Grau. 409-412 [doi]
- Performance evaluation of video coding schemes working at very low bit ratesLaura Contin, Stefano Battista. 409-412 [doi]
- Duration and spectral based stress token generation for HMM speech recognition under stressSahar E. Bou-Ghazale, John H. L. Hansen. 413-416 [doi]
- An LS based new gradient type adaptive algorithm-least squares gradientKiyoshi Nishikawa, Hitoshi Kiya. 413-416 [doi]
- Simultaneous 3-D motion estimation and wire-frame model adaptation including photometric effects for knowledge-based video codingGozde Bozdagi, A. Murat Tekalp, Levent Onural. 413-416 [doi]
- A hierarchical multiprocessor architecture based on heterogeneous processors for video coding applicationsWinfried Gehrke, Richard Hoffer, Peter Pirsch. 413-416 [doi]
- Application of higher order spectra to multi-scale deconvolution of sensor array signalsAmro El-Jaroudi, Tayfun Akgül, Marwan A. Simaan. 413-416 [doi]
- Probabilistic optimum filtering for robust speech recognitionLeonardo Neumeyer, Mitchel Weintraub. 417-420 [doi]
- Video DSP architecture for MPEG2 codecToshiyuki Araki, Masaki Toyokura, Toshihide Akiyama, Hiroshi Takeno, Brent Wilson, Kunitoshi Aono. 417-420 [doi]
- Rate-distortion analysis of variable block size VQ-based motion compensated video codecsSam Liu. 417-420 [doi]
- Two dimensional adaptive filter using Laguerre functionChien-Cheng Tseng, Soo-Chang Pei. 417-420 [doi]
- Identification of parametric linear models with cyclostationary inputsShankar Prakriya, Dimitrios Hatzinakos. 417-420 [doi]
- On the optimality of convergence behaviour for transform-domain split-path adaptive filterK. F. Wan, P. C. Ching. 421-424 [doi]
- A lattice structure for the detection of a non-Gaussian signal in Gaussian AR noiseMariano García Otero, Jose A. Moral-Beneitez. 421-424 [doi]
- Arithmetic codec from behavioral description based LSI-CAD for fully programmable image coding systemJunji Suzuki, Florent Colin, Sadayasu Ono. 421-424 [doi]
- Use of steerable viewing window (SVW) to improve the visual sensation in face to face teleconferencingLiyanage C. De Silva, Kiyoharu Aizawa, Mitsutoshi Hatori. 421-424 [doi]
- Integrating RASTA-PLP into speech recognitionJoachim Köhler, Nelson Morgan, Hynek Hermansky, Hans-Günter Hirsch, Grace Tong. 421-424 [doi]
- A new two-dimensional block adaptive FIR filtering algorithmTerence Wang, Chin-Liang Wang. 425-428 [doi]
- Two-layered DCT based coding scheme for recording digital HDTV signalsJae-Hyun Kim, Goo-Man Park. 425-428 [doi]
- "Whiter than white" noiseTariq S. Durrani, Abdul Rahim Leyman, John J. Soraghan. 425-428 [doi]
- Secure speech and data communication over the public switching telephone networkLuís Díez del Río, Sofia Moreno Perez, Rafael Sarmiento de Sotomayor, José Parera, Marcelino Veiga-Perez, Ramón García Gómez. 425-428 [doi]
- Degraded word recognition based on segmental signal-to-noise ratio weightingHidefumi Kobatake, Yousuke Matsunoo. 425-428 [doi]
- Nonlinear system identification using a Hammerstein model and a cumulant-based Steiglitz-McBride algorithmJohn M. M. Anderson. 429-432 [doi]
- On the importance of the microphone position for speech recognition in the carJohan Smolders, Tom Claes, Gert Sablon, Dirk Van Compernolle. 429-432 [doi]
- Mean-square analysis of the multiple-error and block LMS adaptive algorithmsScott C. Douglas. 429-432 [doi]
- A DSP core for speech coding applicationsJari Nurmi, Ville Eerola, Erwin Ofner, Andreas Gierlinger, Jürgen Jernej, Teppo Karema, Tommi Raita-aho. 429-432 [doi]
- Noise reduction for MPEG type of codecLi Yan. 429-432 [doi]
- The application of subband coding in MPEG for prioritized ATM networksBrian DeCleene, Henrik V. Sorensen. 433-436 [doi]
- Adaptation to new microphones using tied-mixture normalizationTasos Anastasakos, Francis Kubala, John Makhoul, Richard M. Schwartz. 433-436 [doi]
- Performance analysis of some cumulant-based estimators: harmonics in noiseAnanthram Swami. 433-436 [doi]
- A poly-phase structure for system identification and adaptive filteringUmashankar Iyer, Majid Nayeri, Hiroshi Ochi. 433-436 [doi]
- Concurrency characteristics in DSP programsLisa M. Guerra, Miodrag Potkonjak, Jan M. Rabaey. 433-436 [doi]
- An investigation of JPEG image and video compression using parallel processingGregory W. Cook, Edward J. Delp. 437-440 [doi]
- Application-driven design of DSP architectures and compilersMazen A. R. Saghir, Paul Chow, Corinna G. Lee. 437-440 [doi]
- Noise independent speech recognition for a variety of noise typesWilliam C. Treurniet, Yifan Gong. 437-440 [doi]
- Efficient formulation for the realization of discrete cosine transform using recursive structureLap-Pui Chau, Wan-Chi Siu. 437-440 [doi]
- The utilization of higher-order spectra to determine nonlinear radar cross sectionsEdward J. Powers, Sungbin Im. 437-440 [doi]
- Effective nearly lossless compression of digital video sequences via motion-compensated filteringBalas K. Natarajan, Vasudev Bhaskaran. 441-444 [doi]
- Behavioral synthesis of low-cost partial scan designs for DSP applicationsSujit Dey, Miodrag Potkonjak, Rabindra K. Roy. 441-444 [doi]
- Parameter identifiability of multichannel ARMA models of linear non-Gaussian signals via cumulant matchingJitendra K. Tugnait. 441-444 [doi]
- Monitoring the stage of diagonalization in Jacobi-type methodsJürgen Götze. 441-444 [doi]
- Root adaptive homomorphic deconvolution schemes for speech recognition in noisePhilip Lockwood, Patrice Alexandre. 441-444 [doi]
- Automatic code generation for heterogeneous multiprocessorsJosé Luis Pino, Thomas M. Parks, Edward A. Lee. 445-448 [doi]
- Signal bias removal for robust telephone based speech recognition in adverse environmentsMazin G. Rahim, Biing-Hwang Juang. 445-448 [doi]
- On the computation and interpretation of auto- and cross-trispectraVinod Chandran. 445-448 [doi]
- Blind deconvolution for multidimensional imagesRick P. Millane, Philip J. Bones, H. Jiang. 445-448 [doi]
- The quick discrete Fourier transformHaitao Guo, Gary A. Sitton, C. Sidney Burrus. 445-448 [doi]
- Frame-adaptive techniques for quality versus efficiency tradeoffs in STFT analysisErkan Dorken, S. Hamid Nawab. 449-452 [doi]
- p restoration of blurred imagesWai Ho Pun, Brian D. Jeffs. 449-452 [doi]
- Dynamic data flow and control flow in high level DSP code synthesisMatthias Pankert, Oliver Mauss, Sebastian Ritz, Heinrich Meyr. 449-452 [doi]
- Adaptive noise immunity learning for word spottingYoichi Takebayashi, Hiroshi Kanazawa. 449-452 [doi]
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- Minimizing memory requirements for chain-structured synchronous dataflow programsPraveen K. Murthy, Shuvra S. Bhattacharyya, Edward A. Lee. 453-456 [doi]
- Maximum entropy algorithm for spectral estimation problem with gapsRoman Ugrinovsky. 453-456 [doi]
- Decimation-in-time-frequency FFT algorithmAli Saidi. 453-456 [doi]
- Applying generalised cross-validation to image restorationRobert Whatmough. 453-456 [doi]
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- On the equivalence of normalized convolution and normalized differential convolutionCarl-Fredrik Westin, Klas Nordberg, Hans Knutsson. 457-460 [doi]
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- Word-length determination and scaling software for a signal flow block diagramWonyong Sung, Ki-Il Kum. 457-460 [doi]
- A robust sequential parameter estimation for time-varying speech signal analysisHong Sub Choi, Seung Chan Bang, SouGuil Ann. 457-460 [doi]
- Non-linear system identification using Bayesian inferenceKenneth J. Pope, Peter J. W. Rayner. 457-460 [doi]
- Sparse colour and grey scale image restoration using a morphological methodAlan L. Harvey, Harvey A. Cohen. 461-464 [doi]
- An efficient subspace algorithm for 2-D harmonic retrievalFiliep Vanpoucke, Marc Moonen, Yannick Berthoumieu. 461-464 [doi]
- Speech spectrum transformation by speaker interpolationNaoto Iwahashi, Yoshinori Sagisaka. 461-464 [doi]
- Efficient implementation of composite length fast FIR filtering on the "ADSP-2100"Anissa Zergaïnoh-Mokraoui, Pierre Duhamel, Jean Pierre Vidal. 461-464 [doi]
- RECALS II: a new list scheduling algorithmMinjoong Rim, Rajiv Jain. 461-464 [doi]
- Complex interpolation for rational orthogonal signal approximation with applicationsAnthony G. Constantinides, Tania Stathaki. 465-468 [doi]
- On efficient software realization of the prime factor discrete cosine transformDaniel Pak-Kong Lun. 465-468 [doi]
- A novel approach for classifying continuous speech into visible mouth-shape related classesSuhuai Luo, Robin W. King. 465-468 [doi]
- Retiming of DSP programs for optimum vectorizationVojin Zivojnovic, Sebastian Ritz, Heinrich Meyr. 465-468 [doi]
- Image reconstruction from zeros of the z-transformCharles R. Parker, Brenda L. Satherley, Philip J. Bones. 465-468 [doi]
- A regular recursive algorithm for the discrete sine transformZhongde Wang, Graham A. Jullien, William C. Miller. 469-472 [doi]
- Generating efficient loop code for programmable DSPsH. John Reekie, John M. Potter. 469-472 [doi]
- Robust signal modeling through nonlinear least squaresYasemin Yardimci, James A. Cadzow, A. Enis Çetin. 469-472 [doi]
- Voice conversion based on piecewise linear conversion rules of formant frequency and spectrum tiltHideyuki Mizuno, Masanobu Abe. 469-472 [doi]
- A subspace decomposition method for point source localization in blurred imagesMetin Gunsay, Brian D. Jeffs. 469-472 [doi]
- Specification and support for multidimensional DSP in the SILAGE languageIngrid Verbauwhede, Chris J. Scheers, Jan M. Rabaey. 473-476 [doi]
- A fast parallel projection algorithm for set theoretic image recoveryPatrick L. Combettes, Hong Puh. 473-476 [doi]
- Accurate estimation of AR model by tapered SVD without rank determinationHiroshi Kanai, Noriyoshi Chubachi. 473-476 [doi]
- Fast algorithms for interpolation and decimation filter banksRyszard Stasinski, Tor A. Ramstad. 473-476 [doi]
- Voiced-speech analysis based on the residual interfering signal canceler (RISC) algorithmC. S. Ramalingam, Ramdas Kumaresan. 473-476 [doi]
- Enhanced state-space method for high resolution estimation of multiple 2-D coherent sinusoidsJiun-Horng Deng, Jeng-Kuang Hwang. 477-480 [doi]
- Convergence, convergence point and convergence rate for Steiglitz-McBride method; a unified approachMu-Huo Cheng, Virginia L. Stonick. 477-480 [doi]
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- Frequency scalable video coding using the MDCTAndrew W. Johnson, John Princen, Ming H. Chan. 477-480 [doi]
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- Hybrid video coding for low bit-rate applicationsFeng-Ming Wang, Sam Liu. 481-484 [doi]
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- Filter bank design based on discriminative feature extractionAlain Biem, Shigeru Katagiri. 485-488 [doi]
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- Fuzzy based system identificationRobert H. T. Jongwe. 485-488 [doi]
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- A novel efficient parallel algorithm for RNS to binary conversion for arbitrary moduli setAmitabha Das. 485-488 [doi]
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- Rate and resolution scalable subband coding of videoDavid Taubman, Avideh Zakhor. 493-496 [doi]
- A `Jacobi' signal processing unit for time-adaptive SVDEd F. Deprettere, Hylke W. van Dijk, Gerben J. Hekstra. 493-496 [doi]
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- Robust recursive spectral estimation based on AR model excited by a t-distribution processJunibakti Sanubari, Keiichi Tokuda, Mahoki Onoda. 497-500 [doi]
- Efficient vector quantisation of LPC parameters for noisy channelsHarald Skinnemoen, Andrew Perkis. 497-500 [doi]
- Generalized Cramer-Rao bound and the location parameter caseStella N. Batalama, Demitrios Kazakos. 497-500 [doi]
- Subband video coding with temporally adaptive motion interpolationJungwoo Lee, Bradley W. Dickinson. 497-500 [doi]
- Design of high throughput, low latency and low cost structures for linear systemsMiodrag Potkonjak, Mani B. Srivastava. 497-500 [doi]
- Adaptive IIR filtering: composite prefiltered regressor methodVirginia L. Stonick, Mu-Huo Cheng. 501-504 [doi]
- Efficient prediction of uncovered background in interframe coding using spatial extrapolationAndré Kaup, Til Aach. 501-504 [doi]
- Detection and estimation of multiple cisoids in colored noise by Bayesian predictive densitiesChao-Ming Cho, Petar M. Djuric. 501-504 [doi]
- Efficient interblock noiseless coding of speech LPC parametersStefan Bruhn. 501-504 [doi]
- A novel VLSI array design for the discrete Hartley transform using cyclic convolutionJiun-In Guo, Chi-Min Liu, Chein-Wei Jen. 501-504 [doi]
- Using preliminary decisions to reduce complexity in adaptive equalizersPhillip M. S. Burt. 505-508 [doi]
- Rate buffered fractal videoDavid L. Wilson, Jeremy A. Nicholls, Donald M. Monro. 505-508 [doi]
- Variable dimension vector quantization of linear predictive coefficients of speechPhilip A. Chou, Tom D. Lookabaugh. 505-508 [doi]
- Genetic algorithm adaptation of non-linear filter structures for active sound and vibration controlCorey T. Wangler, Colin H. Hansen. 505-508 [doi]
- A MAP solution to off-line segmentation of signalsPetar M. Djuric. 505-508 [doi]
- Optimal VLSI architecture for distributed arithmetic-based algorithmsKamal Nourji, Nicolas Demassieux. 509-512 [doi]
- Orthonormal functions for nonlinear signal processing and adaptive filteringBernard Mulgrew. 509-512 [doi]
- Fast design of optimal array filtersPhillip Musumeci. 509-512 [doi]
- Spatial smoothing for arrays with arbitrary geometryHongyi Wang, K. J. Ray Liu, Henry Anderson. 509-512 [doi]
- Spectral quantization of cepstral coefficientsRoar Hagen. 509-512 [doi]
- 3-D band limitation by motion adaptive spatial filteringHyun Soo Kang, Jong-Hun Kim, Jung Hee Lee, Seong Dae Kim. 513-516 [doi]
- Vector quantization-lattice vector quantization of speech LPC coefficientsJianping Pan, Thomas R. Fischer. 513-516 [doi]
- An analysis of FPGA-based custom computers for DSP applicationsNeil W. Bergmann, J. Craig Mudge. 513-516 [doi]
- Recursive Bayesian location of a discontinuity in time seriesJoseph Ó. Ruanaidh, William J. Fitzgerald, Kenneth J. Pope. 513-516 [doi]
- Sampling frequency requirements for identification and compensation of nonlinear systemsJohn Tsimbinos, Kenneth V. Lever. 513-516 [doi]
- A multi-channel decimator for a tri-level delta-sigma modulatorSal Bernadas, Charles Thompson. 517-520 [doi]
- Segmental quantization of speech spectral informationTorbjørn Svendsen. 517-520 [doi]
- Wideband multiple target trackingA. Satish, Rangasami L. Kashyap. 517-520 [doi]
- The automatic generation of 3D object model from range imageWentao Zheng, Hiroshi Harashima. 517-520 [doi]
- MMD-an efficient approximation to the 2nd order Volterra filterWalter A. Frank. 517-520 [doi]
- Chaotic signal processing via unstable cycle extractionMaciej Ogorzalek, Hervé Dedieu. 521-524 [doi]
- Real-time determination of the signal-to-noise ratio of partly coherent seismic time seriesPeter K. Moller. 521-524 [doi]
- Square-root, reciprocal, sine/cosine, arctangent cell for signal and image processingVijay K. Jain, Lei Lin. 521-524 [doi]
- Low-complexity encoding of speech LSF parameters using constrained-storage TSVQWai-Yip Chan, David Chemla. 521-524 [doi]
- Nonlinear phase estimators based on the Kullback distanceJosé M. N. Leitão, José M. F. Moura. 521-524 [doi]
- Design of an extended Kalman filter frequency trackerBarbara F. La Scala, Robert R. Bitmead. 525-528 [doi]
- A focusing algorithm for the optical array imaging systemOsamu Ikeda. 525-528 [doi]
- Single stage spectral quantization at 20 bitsPer Hedelin. 525-528 [doi]
- A self-structuring algorithm for artificial neural networksA. David M. Garvin. 525-528 [doi]
- ∞ criteriaSanjeev Tavathia, John F. Doherty. 525-528 [doi]
- Broadband maximum energy array with user imposed spatial and frequency constraintsDaniel Korompis, Kung Yao, Flavio Lorenzelli. 529-532 [doi]
- On the design of nonlinear speech predictors with recurrent netsLizhong Wu, Mahesan Niranjan. 529-532 [doi]
- High-order allpole modelling of the spectral envelopeTerrence G. Champion, Robert J. McAulay, Thomas F. Quatieri. 529-532 [doi]
- Reducing the computational requirement of the orthogonal least squares algorithmEngsiong Chng, Sheng Chen 0001, Bernard Mulgrew. 529-532 [doi]
- Robust N-dimensional orientation estimation using quadrature filters and tensor whiteningHans Knutsson, Hans Andersson. 529-532 [doi]
- Kernel invariance method for relating continuous-time with discrete-time nonlinear parametric modelsXiao Zhao, Vasilis Z. Marmarelis. 533-536 [doi]
- Robust methods for using context-dependent features and models in a continuous speech recognizerLalit R. Bahl, Peter V. de Souza, Ponani S. Gopalakrishnan, David Nahamoo, Michael A. Picheny. 533-536 [doi]
- Noise reduction in state space using the focused gamma neural networkJosé Carlos Príncipe, Jyh-Ming Kuo. 533-536 [doi]
- Global non-linear multigrid optimization for image analysis tasksJean-Marc Laferté, Patrick Pérez, Fabrice Heitz. 533-536 [doi]
- New sets of constraints for maximally flat optimum broadband antenna arraysIan L. Thng, Antonio Cantoni, Yee Hong Leung. 533-536 [doi]
- Multiscale difference equation signal modeling and analysis techniquesMurtaza Ali, Ahmed H. Tewfik. 537-540 [doi]
- Adaptive antenna array pattern synthesis using recursive minimum-norm solution with constraint selection and phase optimizationKai-Bor Yu. 537-540 [doi]
- Maximum likelihood scale estimation for a class of Markov random fieldsCharles A. Bouman, Ken D. Sauer. 537-540 [doi]
- Genones: optimizing the degree of mixture tying in a large vocabulary hidden Markov model based speech recognizerVassilios Digalakis, Hy Murveit. 537-540 [doi]
- Design of neural estimators for multisensors: second order backpropagation, initialization and generalizationPascale Hirschauer, Pascal Larzabal, Henri Clergeot. 537-540 [doi]
- An invariance property of neural networksHerbert Gish, Man-Hung Siu. 541-544 [doi]
- A comparison of phoneme decision tree (PDT) and context adaptive phone (CAP) based approaches to vocabulary-independent speech recognitionRoger K. Moore, Martin J. Russell, Peter Nowell, Simon Downey, Sue Browning. 541-544 [doi]
- A sparse approach in partially adaptive linearly constrained arraysIain Scott, Bernard Mulgrew. 541-544 [doi]
- Look-ahead decision-feedback ΣΔ modulationJohn T. Stonick, Jim L. Rulla, Sasan H. Ardalan. 541-544 [doi]
- Divergence penalty for image regularizationJoseph A. O'Sullivan. 541-544 [doi]
- Towards large vocabulary Mandarin Chinese speech recognitionHsiao-Wuen Hon, Baosheng Yuan, Yen-Lu Chow, Shankar Narayan, Kai-Fu Lee. 545-548 [doi]
- A constrained neural network with complex activation function: application to time-frequency analysisMohamed Ibnkahla, Stephane Puechmorel, Francis Castanie. 545-548 [doi]
- p-complex approximation using iterative reweighted least squares for FIR digital filtersJose Antonio Barreto, C. Sidney Burrus. 545-548 [doi]
- New prospects in line detection for remote sensing imagesNicolas Merlet, Josiane Zerubia. 545-548 [doi]
- A construction of arrays free of rank ambiguitiesKah-Chye Tan, Zenton Goh. 545-548 [doi]
- Bayesian restoration of millimeter wave imageryBobby R. Hunt, David DeKruger. 549-552 [doi]
- Improving speech recognition performance via phone-dependent VQ codebooks and adaptive language models in SPHINX-IIMei-Yuh Hwang, Ronald Rosenfeld, Eric H. Thayer, Mosur Ravishankar, Lin Chase, Robert Weide, Xuedong Huang, Fil Alleva. 549-552 [doi]
- Study of ambiguities of linear arraysChristos Proukakis, Athanassios Manikas. 549-552 [doi]
- The conditional expectation via a general class of nonlinear networksMang Zhu, James A. Cadzow. 549-552 [doi]
- A complex Chebyshev approximation algorithm for FIR filter designAshraf Alkhairy. 549-552 [doi]
- Bimodal recognition experiments with recurrent neural networksPiero Cosi, Emanuela Magno Caldognetto, Kyriaki Vagges, Gian Antonio Mian, Matteo Contolini. 553-556 [doi]
- Residual image coding using mathematical morphologyJosep R. Casas, Luis Torres. 553-556 [doi]
- New graph search techniques for speech recognitionPatrick Kenny, Paul Labute, Zhishun Li, Douglas D. O'Shaughnessy. 553-556 [doi]
- A generalized Remez multiple exchange algorithm for complex FIR filter designChing-Yih Tseng. 553-556 [doi]
- Modelling and estimation of mutual coupling between array elementsAthanassios Manikas, Nikolaos Fistas. 553-556 [doi]
- Pareto optimal designs of low sensitivity digital filters: parallel and cascade form structuresTarek Tutunji, Victor E. DeBrunner. 557-560 [doi]
- A new competitive learning algorithm for vector quantizationCe Zhu, Lihua Li 0002, Zhenya He, Jun Wang 0002. 557-560 [doi]
- Optimal positioning of sensors for a microphone arraySaeed Gazor, Yves Grenier. 557-560 [doi]
- Fractal image compression without searchingDonald M. Monro, Stuart J. Woolley. 557-560 [doi]
- The LIMSI continuous speech dictation system: evaluation on the ARPA Wall Street Journal taskJean-Luc Gauvain, Lori Faith Lamel, Gilles Adda, Martine Adda-Decker. 557-560 [doi]
- Comparative experiments on large vocabulary speech recognitionFrancis Kubala, Anastasios Anastasakos, John Makhoul, Long Nguyen, Richard M. Schwartz, George Zavaliagkos. 561-564 [doi]
- The design of neural network configuration for object recognitionBin Qiu, Paul Im, Anne Pleasants. 561-564 [doi]
- Maximum likelihood technique to quadrature parameter estimationJohn W. Pierre, Roger A. Green. 561-564 [doi]
- Design of FIR filters with powers of two coefficients based on a new quantization quality criterionTolga Çiloglu, Zafer Ünver. 561-564 [doi]
- On the convergence of fractal transformsBemd Hurtgen, Thomas Hain. 561-564 [doi]
- M-lattice: a novel non-linear dynamical system and its application to halftoningAlex Sherstinsky, Rosalind W. Picard. 565-568 [doi]
- High quality text-to-speech synthesis: a comparison of four candidate algorithmsThierry Dutoit. 565-568 [doi]
- A new improved collage theorem with applications to multiresolution fractal image codingGeir E. Øien, Zachi Baharav, Skjalg Lepsøy, Ehud D. Karnin. 565-568 [doi]
- Error criteria for filter designBeth A. Weisburn, Thomas W. Parks, Ram G. Shenoy. 565-568 [doi]
- Backward elimination procedures for testing multiple hypotheses: application to optimal sensor locationAbdelhak M. Zoubir. 565-568 [doi]
- An efficient method for IIR filter designAshraf Alkhairy. 569-572 [doi]
- Overlapped adaptive partitioning for image coding based on the theory of iterated functions systemsEmmanuel Reusens. 569-572 [doi]
- Development of a text-to-speech system for Japanese based on waveform splicingHisashi Kawai, Norio Higuchi, Tohru Shimizu, Seiichi Yamamoto. 569-572 [doi]
- A multiresolution probabilistic neural network for image segmentationStefanos D. Kollias, Dimitrios Kalogeras. 569-572 [doi]
- An ML/MMSE estimation approach to blind equalizationShao Min, Chrysostomos L. Nikias. 569-572 [doi]
- Subspace methods for the blind identification of multichannel FIR filtersEric Moulines, Pierre Duhamel, Jean-François Cardoso, Sylvie Mayrargue. 573-576 [doi]
- Echocardiogram structure and tissue classification using hierarchical fuzzy neural networksTom Brotherton, Tom Pollard, Pat Simpson, Anthony DeMaria. 573-576 [doi]
- New algorithm for spectral smoothing and envelope modification for LP-PSOLA synthesisFrancisco M. Gimenez de los Galanes, Mohammad Hasan Savoji, José Manuel Pardo. 573-576 [doi]
- Position-dependent encodingJohn G. Apostolopoulos, Aleksandar Pfajfer, Hae Mook Jung, Jae S. Lim. 573-576 [doi]
- An improvement to the Powell and Chau linear phase IIR filtersAlan N. Willson Jr., Henry John Orchard. 573-576 [doi]
- On channel estimation for RAKE receiver in a mobile multipath fading channelFu Li, Heng Xiao, Jin Yang. 577-580 [doi]
- A new waveform speech synthesis approach based on the COC speech spectrumKenzo Itoh, Shin'ya Nakajima, Tomohisa Hirokawa. 577-584 [doi]
- Allpass transfer functions with prescribed group delayMax Gerken. 577-580 [doi]
- A simulation environment for very large neural networksAndre Yakovleff, Mario Cavaiuolo. 577-580 [doi]
- 3-D image coding based on affine transformToshiaki Fujii, Hiroshi Harashima. 577-580 [doi]
- Minimization of pole/zero sensitivity in digital filter design with sparse structure considerationGang Li. 581-584 [doi]
- A study of convex coders with an application to image codingKohtaro Asai, Nguyen T. Thao, Martin Vetterli. 581-584 [doi]
- A deterministic approach to blind identification of multi-channel FIR systemsHui Liu, Guanghan Xu, Lang Tong. 581-584 [doi]
- Approach of using a density equalizing function to self-organizing learning for solving travelling salesman problemClifford Sze-Tsan Choy, Wan-Chi Siu. 581-584 [doi]
- Generalized magnitude and power complementary filtersS. Radhakrishnan Pillai, Gregory H. Allen. 585-588 [doi]
- Articulatory speech synthesis based on fractional delay waveguide filtersVesa Välimäki, Matti Karjalainen, Timo Kuisma. 585-588 [doi]
- Blind fractionally-spaced equalization, perfect-reconstruction filter banks and multichannel linear predictionDirk T. M. Slock. 585-588 [doi]
- An extension to the analytical Gabor expansion with applications in image codingAndreas Teuner, Per Asbeck Nielsen, Bedrich J. Hosticka. 585-588 [doi]
- A new cascaded projection pursuit network for nonlinear regressionShih-Shien You, Jenq-Neng Hwang, I-Chang Jou, Shyh-Rong Lay. 585-588 [doi]
- Neural receiver for CPM signalsPatrizia Bollini, Guido Castellini, Enrico Del Re, Giacomo Mangani, Laura Pierucci. 589-592 [doi]
- Perturbation effects on filters having multiple polesGregory H. Allen. 589-592 [doi]
- A new algorithm for fast blind equalization of wireless communication channelsGuanghan Xu, Lang Tong, Hui Liu. 589-592 [doi]
- 0 and duration modeling in text to speech conversion for SpanishEduardo López Gonzalo, Luis A. Hernández Gómez. 589-592 [doi]
- Comparison of "wavelet" filters and subband analysis structures for still image compressionJames P. Andrew, Philip Ogunbona, Frank John Paoloni. 589-592 [doi]
- Application of neural networks to detecting misfire in automotive enginesWilliam B. Ribbens, Jaehong Park, DaeEun Kim. 593-596 [doi]
- Adaptive blind equalization using weighted cumulant slicesJosep Vidal, José A. R. Fonollosa. 593-596 [doi]
- Automatic generation of prosodic rules for speech synthesisYoichi Yamashita, Riichiro Mizoguchi. 593-596 [doi]
- Critical comparison of Hankel-norm optimal approximation and balanced model truncation algorithms as vehicles for FIR-to-IIR filter order reductionBartlomiej Beliczynski, Jeremi Gryka, Izzet Kale. 593-596 [doi]
- Image coding with overlapped projection and pyramid vector quantizationRosa Lancini, Emanuele Marconetti, Stefano Tubaro. 593-596 [doi]
- Segmentation based coding of textures using stochastic vector quantizationLuis Torres, Josep R. Casas, S. de Diego. 597-600 [doi]
- Generalised running DHT and real-time (DHT) analysersJohnson Ihyeh Agnomua. 597-600 [doi]
- A MRF-based parallel processing algorithm for speech recognition using linear predictive HMMHideki Noda, Mehdi N. Shirazi. 597-600 [doi]
- Optimal, matching-score network for pattern classificationAndrew Luk, Wai-Fung Leung. 597-600 [doi]
- Multiple neural networks using the reduced input dimensionJongwan Kim, Jesung Ahn, Chong-Sang Kim, Heeyeung Hwang, Seongwon Cho. 601-604 [doi]
- Image coding using pyramid vector quantization of subband coefficientsEly K. Tsern, Teresa H. Y. Meng. 601-604 [doi]
- Speech modelling using cepstral-time feature matrices and hidden Markov modelsBen P. Milner, Saeed Vaseghi. 601-604 [doi]
- On the sensitivity of realizations of real coefficient digital filters using complex arithmeticChimin Tsai, Meng-Liang Lin, Adly T. Farn. 601-604 [doi]
- HMM with global path constraint in Viterbi decoding for isolated word recognitionWeon-Goo Kim, Jeung-Yoon Choi, Dae Hee Youn. 605-608 [doi]
- On-line handwritten character recognition using parallel neural networksEveline J. Bellegarda, Jerome R. Bellegarda, Jin H. Kim. 605-608 [doi]
- The sensitivity of envelope-constrained filters with uncertain inputWei Xing Zheng, Antonio Cantoni, Kok Lay Teo. 605-608 [doi]
- Vector quantization over a noisy channel using soft decision decodingMikael Skoglund, Per Hedelin. 605-608 [doi]
- Asymptotic behavior of digital filters with block floating point arithmeticPeter H. Bauer. 609-612 [doi]
- Predictive mean search algorithm for vector quantization of imagesKwok-Tung Lo, Jian Feng. 609-612 [doi]
- Invariant property of spatio-temporal feature maps using gated neuronal architectureV. Chandrasekaran, Marimuthu Palaniswani, Terri M. Caelli. 609-612 [doi]
- Pseudo-segment based speech recognition using neural recurrent whole-word recognizersPhilippe Le Cerf, Kris Demuynck, Jacques Duchateau, Dirk Van Compernolle. 609-612 [doi]
- An efficient neural prediction for vector quantizationRoberto Fioravanti, Stefano Fioravanti, Daniele D. Giusto. 613-616 [doi]
- Artifact elimination using fuzzy rule based adaptive nonlinear filterTohru Kiryu, Hidekazu Kaneko, Yoshiaki Saitoh. 613-616 [doi]
- Using Gaussian mixture modeling in speech recognitionYaxin Zhang, Michael D. Alder, Roberto Togneri. 613-616 [doi]
- A new composite feature vector for Arabic handwritten signature recognitionAhmed M. Darwish 0001, Gasser Auda. 613-616 [doi]
- Task independent and dependent training: performance comparison of HMM and hybrid HMM/MLP approachesJean-Marc Boite, Hervé Bourlard, Bart D'hoore, Sari Accaino, Johan Vantieghem. 617-620 [doi]
- On two-pattern classification and feature selection using neural networksLuan Ling Lee. 617-620 [doi]
- Next-state functions for finite-state vector quantizationNasser M. Nasrabadi, Syed A. Rizvi. 617-620 [doi]
- Windows with rapidly decaying sidelobes and steerable sidelobe dipsMagdy T. Hanna. 617-620 [doi]
- Using artificial neural networks to improve the mechanical signature analysis testVictor E. DeBrunner, Tod Bussert. 621-624 [doi]
- Lapped RVQ and alphabet and entropy constraintsFrancesco Nesci, Faouzi Kossentini, Mark J. T. Smith. 621-624 [doi]
- Evaluation of several variable FIR fractional-sample delay filtersGerald D. Cain, N. Paul Murphy, Andrzej Tarczynski. 621-624 [doi]
- The influence of speech coding algorithms on automatic speech recognitionStephan Euler, Joachim Zinke. 621-624 [doi]
- Discriminative training of high performance speech recognizer using N best candidatesJung-Kuei Chen, Frank K. Soong. 625-628 [doi]
- PDF estimation using order statistic filter bankRisto Suoranta, Kari-Pekka Estola, Seppo Rantala, Heli Väätäjä. 625-628 [doi]
- One-pass adaptive universal vector quantizationMichelle Effros, Philip A. Chou, Robert M. Gray. 625-628 [doi]
- Transformation of optimized prototypes for handwritten digit recognitionHong Yan. 625-628 [doi]
- Image reconstruction using vector quantized linear interpolationSheila S. Hemami, Robert M. Gray. 629-632 [doi]
- Model topology selection for isolated word recognitionPietro Laface, Luciano Fissore. 629-632 [doi]
- Incremental learning using the time delay neural networkMinh Tue Vo. 629-632 [doi]
- A connectionist recognizer for on-line cursive handwriting recognitionStefan Manke, Ulrich Bodenhausen. 633-636 [doi]
- Which model for future speech recognition systems: hidden Markov models or finite-state automata?Joseph di Martino, Jean-François Mari, Bruno Mathieu, Karine Perot, Kamel Smaïli. 633-636 [doi]
- On the fuzzy vector quantization based hidden Markov modelEiichi Tsuboka, Jun'ichi Nakahashi. 637-640 [doi]
- On-line cursive script recognition using time delay neural networks and hidden Markov modelsMarkus Schenkel, Isabelle Guyon, Don Henderson. 637-640 [doi]
- Performance of radar target recognition schemes using neural networks-a comparative studyNanda Nandagopal, N. M. Martin, R. P. Johnson, Peter Lozo, Marimuthu Palaniswami. 641-644 [doi]
- Mutual information neural networks: a new connectionist approach for dynamic speech recognition tasksGerhard Rigoll. 645-648 [doi]
- Isolated word recognition using a hybrid neural networkVahidi Tabarabaee, Babak Azimisadjadi, Seyed Bahram Zahir Azami, Caro Lucas. 649-652 [doi]
- Application of a generalized probabilistic descent method to recurrent neural network based speech recognitionSin-Horng Chen, Yuan-Fu Liao, Wen-Yuan Chen. 653-656 [doi]
- Hybrid model decomposition of speech and noise in a radial basis function neural model frameworkHelge B. D. Sørensen, Uwe Hartmann. 657-660 [doi]
- A neural architecture for hierarchical clusteringAbdesselam Bouzerdoum, Michael L. Southcott, Jihan Zhu, Robert E. Bogner. 661-664 [doi]
- Vowel classification using a neural predictive HMM: a discriminative training approachKhaled Hassanein, Li Deng, Mohamed I. Elmasry. 665-668 [doi]
- "Eigenlips" for robust speech recognitionChristoph Bregler, Yochai Konig. 669-672 [doi]
- Incorporating linguistic features in a hybrid HMM/MLP speech recognizerVictor Abrash, Michael Cohen, Horacio Franco, Isao Arima. 673-676 [doi]
- A new model-discriminant training algorithm for hybrid NN-HMM systemsWolfgang Reichl, Peter Caspary, Günther Ruske. 677-680 [doi]
- Combining stochastic trajectory model and discriminative feature in speech recognizerJun He, Henri Leich. 681-684 [doi]
- Noise immunization using neural net for speech recognitionRavi Sankar, Shrenik Patravali. 685-688 [doi]