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ORCIDICASSP 1990: Albuquerque, New Mexico, USA
- 1990 International Conference on Acoustics, Speech, and Signal Processing, ICASSP '90, Albuquerque, New Mexico, USA, April 3-6, 1990. IEEE 1990
- W. Bastiaan Kleijn
:
Source-dependent channel coding for CELP. 1-4 - Michael S. Brandstein, Peter A. Monta, John C. Hardwick, Jae S. Lim:
A real-time implementation of the improved MBE speech coder. 5-8 - Paul C. Meuse:
A 2400 bps multi-band excitation vocoder. 9-12 - Carmen García-Mateo, Francisco Javier Casajús-Quirós, Luis A. Hernández Gómez:
Multi-band vector excitation coding of speech at 4.8 kbps. 13-16 - Jorge S. Marques, Luís B. Almeida, José M. Tribolet:
Harmonic coding at 4.8 kb/s. 17-20 - Kazunori Mano, Takehiro Moriya:
4.8 kbit/s delayed decision CELP coder using tree coding. 21-24 - Jae H. Chung, Ronald W. Schafer:
Excitation modeling in a homomorphic vocoder. 25-28 - Masami Akamine, Kimio Miseki:
CELP coding with an adaptive density pulse excitation model. 29-32 - Rosario Drogo de Iacovo, Roberto Montagna, Daniele Sereno:
Vector quantization and perceptual criteria in SVD based CELP coders. 33-36 - Saeed Vaseghi:
Finite state CELP for variable rate speech coding. 37-40 - Jean-Claude Junqua:
ORION: a two pass hybrid system for isolated-words automatic speech recognition. 41-44 - Alexander I. Rudnicky, Michelle Sakamoto, Joseph Polifroni:
Spoken language interaction in a goal-directed task. 45-48 - Victor Zue, James R. Glass, David Goodine, Michael Philips, Stephanie Seneff:
The SUMMIT speech recognition system: phonological modelling and lexical access. 49-52 - Toshiyuki Hanazawa, Kenji Kita, Satoshi Nakamura, Takeshi Kawabata, Kiyohiro Shikano:
ATR HMM-LR continuous speech recognition system. 53-56 - Volker Steinbiss, Andreas Noll, A. Peaseler, Hermann Ney, Henning Bergmann, Christian Dugast, Hans-Hermann Hamer, H. Piotrowski, Horst Tomaschewski, Andrea Zielinski:
A 10000-word continuous-speech recognition system. 57-60 - Shinta Kimura:
100000-word recognition using acoustic-segment networks. 61-64 - Lin-Shan Lee, Chiu-yu Tseng, Hung-Yan Gu, Fu-hua Liu, Robert Chen-Hao Chang, Shew-Heng Hsieh, Chian-hung Chen:
A real-time Mandarin dictation machine for Chinese language with unlimited texts and very large vocabulary. 65-68 - Martin J. Russel, Keith Ponting, S. M. Peeling, Sue Browning, John S. Bridle, Roger K. Moore, Isabel Galiano, P. Howell:
The ARM continuous speech recognition system. 69-72 - Victor Zue, James R. Glass, David Goodine, Hong C. Leung, Michael S. Phillips, Joseph Polifroni, Stephanie Seneff:
The VOYAGER speech understanding system: preliminary development and evaluation. 73-76 - Michael Cohen, Hy Murveit, Jared Bernstein, Patti J. Price, Mitch Weintraub:
The decipher speech recognition system. 77-80 - Richard M. Schwartz, Yen-Lu Chow:
The N-best algorithms: an efficient and exact procedure for finding the N most likely sentence hypotheses. 81-84 - Lalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, Dimitri Kanevsky, David Nahamoo:
Constructing groups of acoustically confusable words. 85-88 - Abdulmesih Aktas, Otto Schmidbauer, K. H. Maier, Wolfgang Feix:
Classification of coarse phonetic categories in continuous speech: statistical classifiers vs. temporal flow connectionist network. 89-92 - Stephen E. Levinson, Andrej Ljolje, Laura G. Miller:
Continuous speech recognition from a phonetic transcription. 93-96 - David S. Pallett, William M. Fisher, Jonathan G. Fiscus:
Tools for the analysis of benchmark speech recognition tests. 97-100 - Edmund Lai, G. A. Carrijo, R. Bennett, Roberto Togneri, Michael D. Alder, Yianni Attikiouzel:
An English language speech database at the University of Western Australia. 101-104 - Joseph Picone:
The demographics of speaker independent digit recognition. 105-108 - Charles R. Jankowski Jr., Ashok Kalyanswamy, Sara Basson, Judith Spitz:
NTIMIT: a phonetically balanced, continuous speech, telephone bandwidth speech database. 109-112 - David P. Morgan, Christopher L. Scofield, Theresa M. Lorenzo, Edward C. Real, David P. Loconto:
A keyword spotter which incorporates neural networks for secondary processing. 113-116 - Herbert Gish, Yen-Lu Chow, Jan Robin Rohlicek:
Probabilistic vector mapping of noisy speech parameters for HMM word spotting. 117-120 - Noboru Sugamura:
Continuous speech recognition using large vocabulary word spotting and CV syllable spotting. 121-124 - Ayman Asadi, Richard M. Schwartz, John Makhoul:
Automatic detection of new words in a large vocabulary continuous speech recognition system. 125-128 - Richard C. Rose, Douglas B. Paul:
A hidden Markov model based keyword recognition system. 129-132 - Fritz Class, Alfred Kaltenmeier, Peter Regel-Brietzmann, Karl Trottler:
Fast speaker adaptation for speech recognition systems. 133-136 - Francis Kubala, Richard M. Schwartz, Chris Barry:
Speaker adaptation from a speaker-independent training corpus. 137-140 - Gerhard Rigoll:
Baseform adaptation for large vocabulary hidden Markov model based speech recognition systems. 141-144 - Chin-Hui Lee, Chih-Heng Lin, Biing-Hwang Juang:
A study on speaker adaptation of continuous density HMM parameters. 145-148 - Luc Mathan, Laurent Miclet:
Speaker hierarchical clustering for improving speaker-independent HMM word recognition. 149-152 - Hiroaki Hattori, Satoshi Nakamura, Kiyohiro Shikano:
Supplementation of HMM for articulatory variation in speaker adaptation. 153-156 - Satoshi Nakamura, Kiyohiro Shikano:
A comparative study of spectral mapping for speaker adaptation. 157-160 - S. J. Cox, John S. Bridle:
Simultaneous speaker normalisation and utterance labelling using Bayesian/neural net techniques. 161-164 - John B. Hampshire II, Alexander H. Waibel:
The Meta-Pi network: connectionist rapid adaptation for high-performance multi-speaker phoneme recognition. 165-168 - Katsuhiko Shirai, Naoki Hosaka, Eiichiro Kitagawa:
Speaker adaptive phoneme recognition by multi-level clustering based on mutual information criterion. 169-172 - Natalie Fournier, Yves Grenier:
A real-time vector excited adaptive predictive coder at 9.6 kbit/s. 173-176 - Claude Laflamme, Jean-Pierre Adoul, H. Y. Su, Sarto Morissette:
On reducing computational complexity of codebook search in CELP coder through the use of algebraic codes. 177-180 - Juin-Hwey Chen, Melvin J. Melchner, Richard V. Cox, Duane O. Bowker:
Real-time implementation and performance of a 16 kb/s low-delay CELP speech coder. 181-184 - Frank K. Soong, Biing-Hwang Juang:
Optimal quantization of LSP parameters using delayed decisions. 185-188 - Roar Hagen, Per Hedelin:
Low bit-rate spectral coding in CELP, a new LSP-method. 189-192 - Nam Phamdo, Nariman Farvardin:
Coding of speech LSP parameters using TSVQ with interblock noiseless coding. 193-196 - Philip A. Chou, Tom D. Lookabaugh:
Conditional entropy-constrained vector quantization of linear predictive coefficients. 197-200 - Richard A. Dean:
Global maximum likelihood voice decoding. 201-204 - Alessandro Falaschi, Massimo Giustiniani, Piero Pierucci:
A finite states Markov quantizer for speech coding. 205-208 - F. F. Tzeng:
An analysis-by-synthesis linear predictive model for narrowband speech coding. 209-212 - Masaaki Honda:
Speech coding using waveform matching based on LPC residual phase equalization. 213-216 - Ki Yong Lee, Byung-Gook Lee, Iickho Song, Souguil Ann:
On Bernoulli-Gaussian process modeling of speech excitation source. 217-220 - J. A. Naylor:
A neural network algorithm for enhancing delta modulation/LPC tandem connections. 221-224 - Mahesan Niranjan:
CELP coding with adaptive output-error model identification. 225-228 - S. A. Atungsiri, Ahmet M. Kondoz, Barry G. Evans:
Error detection and control for the parametric information in CELP coders. 229-232 - Renaud J. Di Francesco, Claude Lamblin, Alain Le Guyader, Dominique Massaloux:
Variable rate speech coding with online segmentation and fast algebraic codes. 233-236 - Jerry D. Gibson, Yoon-Chae Cheong, Hong Chae Woo, Wen-Whei Chang:
A comparison of backward adaptive prediction algorithms in low delay speech coders. 237-240 - Tomohiko Taniguchi, Yoshinori Tanaka, A. Sasama, Yasuji Ohta:
Principal axis extracting vector excitation coding: high quality speech at 8 kb/s. 241-244 - Eyal Yair, Kenneth Zeger, Allen Gersho:
Conjugate gradient methods for designing vector quantizers. 245-248 - Robert J. McAulay, Thomas F. Quatieri:
Pitch estimation and voicing detection based on a sinusoidal speech model. 249-252 - Alfredo Restrepo Palacios, Irwin W. Sandberg, Alan C. Bovik:
Non-Euclidean locally monotonic regression. 1201-1204 - Yoram Bresler, Scott P. Litke:
A parametric technique for superresolution image reconstruction. 1205-1208 - Vip Desai, K. S. Arun:
Robust parametric reconstruction of signals by rational modeling. 1209-1212 - Barry J. Sullivan:
Ill-conditioned signal restoration and the conjugate gradient method. 1213-1216 - M. A. Khasawneh, Winser E. Alexander:
Mean square error analysis for the fast LMS-sine algorithm. 1257-1260 - Heping Ding, Moustafa M. Fahmy:
Inverse of linear periodically time-varying filtering. 1217-1220 - Boualem Bouachache, Langford B. White:
Instantaneous frequency estimation and automatic time-varying filtering. 1221-1224 - Laith Naaman, Alan C. Bovik:
Least-squares order statistic filters for signal restoration in dependent noise. 1225-1228 - Francesco Palmieri:
Adaptive recursive order statistic filters. 1229-1232 - John L. Brown Jr., Sergio D. Cabrera:
Multi-channel signal reconstruction using noisy samples. 1233-1236 - Naoki Saito:
Superresolution of noisy band-limited data by data adaptive regularization and its application to seismic trace inversion. 1237-1240 - Bhaskar D. Rao:
Some new properties of the LMS FIR-ALE. 1241-1244 - Nam Ik Cho, Sang Uk Lee:
On the performance analysis of ALE using an IIR lattice notch filter. 1245-1248 - John J. Shynk, Christina K. Chan:
A comparative analysis of the stationary points of the constant modulus algorithm based on Gaussian assumptions. 1249-1252 - Peter J. Voltz, Frank Kozin:
Almost-sure sample stability of the continuous-time LMS algorithm. 1253-1256 - Victor E. DeBrunner, A. A. (Louis) Beex:
An informational approach to the convergence of output error adaptive IIR filter structures. 1261-1264 - Majid Nayeri, Hong Fan:
Asymptotic stability of SMM and RGM for insufficient order IIR adaptive filters. 1265-1268 - Tarek I. Haweel, Peter M. Clarkson:
Analysis and generalization of a median adaptive filter. 1269-1272 - Raziel Haimi-Cohen, Hanan Herzberg, Yair Be'ery:
Delayed adaptive LMS filtering: current results. 1273-1276 - Nancy E. Hubing, S. Thomas Alexander:
Statistical analysis of the soft constrained initialization of recursive least squares algorithms. 1277-1280 - Bowonkoon Chitprasert, K. R. Rao:
Discrete cosine transform filtering. 1281-1284 - Gregory W. Medlin:
A design technique for high order digital differentiators. 1285-1288 - Ali Zilouchian, Nurgun Erdol:
A minimum-order structure for fixed-point digital filters. 1289-1292 - Arnab K. Shaw, Pradeep Misra:
An exact realization of 2-D IIR filters using separable 1-D modules. 1293-1296 - Y. Z. Zhang, T. W. Cole, Y. F. Yao:
Optimum removal of LCOs in digital filters. 1297-1300 - Nevio Benvenuto, Michele Marchesi, Aurelio Uncini:
Results on the application of simulated annealing algorithm for the design of digital filters with powers-of-two coefficients. 1301-1304 - C. Sidney Burrus, Admadji W. Soewito, Ramesh A. Gopinath:
Least squared error FIR filter design with spline transition functions. 1305-1308 - M. Scott Andrews, Joseph Picone, Ronald D. DeGroat:
Robust pitch determination via SVD based cepstral methods. 253-256 - Mohammad Ali Masnadi-Shirazi, Nasir Ahmed:
Laguerre approximation of nonrecursive discrete-time systems. 1309-1312 - Hans Wilhelm Schüßler, Peter Steffen:
On the design of allpasses with prescribed group delay. 1313-1316 - L. Hodgson, M. E. Jernigan, Barry L. Wills:
Nonlinear multiplicative cepstral analysis for pitch extraction in speech. 257-260 - John Oglesby, John S. Mason:
Optimisation of neural models for speaker identification. 261-264 - Wei-Ping Zhu, Zhenya He:
Design of three-dimensional spherically symmetric digital filters. 1317-1320 - Younès Bennani, Françoise Fogelman-Soulié, Patrick Gallinari:
A connectionist approach for automatic speaker identification. 265-268 - Takao Kobayashi, Satoshi Imai:
Complex Chebyshev approximation for IIR digital filters using an iterative WLS technique. 1321-1324 - Aaron E. Rosenberg, Chin-Hui Lee, Frank K. Soong:
Sub-word unit talker verification using hidden Markov models. 269-272 - Ren-Hua Wang, Lingshen He, Hiroya Fujisaki:
A weighted distance measure based on the fine structure of feature space: application to speaker recognition. 273-276 - Virginia L. Stonick, S. Thomas Alexander:
Global optimal IIR filter design and ARMA estimation using homotopy continuation methods. 1325-1328 - Chi-Shi Liu, Wern-Jun Wang, Min-Tau Lin, Hsiao-Chuan Wang:
Study of line spectrum pair frequencies for speaker recognition. 277-280 - Paul T. Yang, Rajeev Jain, Toshiaki Yoshino, Wanda Gass, Ashwin Shah:
A functional silicon compiler for high speed FIR digital filters. 1329-1332 - Leland B. Jackson, Gerald J. Lemay:
A simple Remez exchange algorithm to design IIR filters with zeros on the unit circle. 1333-1336 - Richard J. Hartnett, Gloria Faye Boudreaux:
Design of efficient parallel hybrid FIR filters using dynamic programming and subset selection methods. 1337-1340 - Roy Chapman, M. A. Rahman:
Chebyshev polynomial based transfer functions for orthogonal lattice filters. 1341-1344 - Michael I. Savic, Sunil K. Gupta:
Variable parameter speaker verification system based on hidden Markov modeling. 281-284 - Jonathan S. Abel:
A bound on mean square estimate error. 1345-1348 - Constantin Papaodysseus, Constantin Triantafillou, Elias Koukoutsis, George Carayannis:
Error propagation and numerical recovery for a class of parametric DSP algorithms. 1349-1352 - Yifan Gong, Jean-Paul Haton:
Text-independent speaker recognition by trajectory space comparison. 285-288 - Don H. Johnson, Anand R. Kumar:
Modeling and analyzing fractal point processes. 1353-1356 - Brian H. Maranda, John A. Fawcett:
The performance analysis of a fourth-moment detector. 1357-1360 - Herbert Gish:
Robust discrimination in automatic speaker identification. 289-292 - Herbert Gish:
A probabilistic approach to the understanding and training of neural network classifiers. 1361-1364 - Victor C. Soon, Lang Tong, Y. F. Huang, R. Liu:
An extended fourth order blind identification algorithm in spatially correlated noise. 1365-1368 - Richard C. Rose, Douglas A. Reynolds:
Text independent speaker identification using automatic acoustic segmentation. 293-296 - Michael S. Scordilis, John N. Gowdy:
Neural network control for a cascade/parallel formant synthesizer. 297-300 - S. Kalson:
Adaptive detection with diagonal loading [for adaptive antenna array]. 1369-1372 - Johan de Veth, Bert Cranen, Helmer Strik, Lou Boves:
Extraction of control parameters for the voice source in a text-to-speech system. 301-304 - Georgios B. Giannakis, Amod V. Dandawate:
Linear and non-linear adaptive noise cancelers. 1373-1376 - John J. Shynk, Sumit Roy:
Analysis of a perceptron learning algorithm with momentum updating. 1377-1380 - Mahmood R. Azimi-Sadjadi, Stuart Citrin, Sassan Sheedvash:
Supervised learning process of multi-layer perceptron neural networks using fast least squares. 1381-1384 - V. John Mathews, Zhenhua Xie:
Stochastic gradient adaptive filters with gradient adaptive step sizes. 1385-1388 - Tetsuya Shimamura, Sho Takahashi:
Least squares adaptive equalizer using analytic signals. 1393-1396 - Ioannis Pitas, Anastasios N. Venetsanopoulos:
LMS and RLS adaptive L-filters. 1389-1392 - Johan de Veth, W. van Golstein Brouwers, H. Loman, Lou Boves:
Robust ARMA analysis as an aid in developing parameter control rules for a pole-zero cascade speech synthesizer. 305-308 - John F. Doherty, Richard J. Mammone:
A row-action projection algorithm for adaptive filtering. 1397-1400 - Eric Moulines, Françoise Emerard, Danielle Larreur, J. L. Le Saint-Milon, L. Le Faucheur, F. Marty, Francis Charpentier, Christel Sorin:
A real-time French text-to-speech system generating high-quality synthetic speech. 309-312 - A. M. Sequeira, Charles W. Therrien:
A new 2-D fast RLS algorithm. 1401-1404 - Jean-Christophe Pesquet, Georgios Tziritas, Odile Macchi:
Modified LMS algorithms for robust ADPCM. 1405-1408 - Xiao-Hu Yu:
A new numerically stable structure for fast RLS adaptive filtering. 1409-1412 - Wendy J. Holmes, John N. Holmes, Michael W. Judd:
Extension of the bandwidth of the JSRU parallel-formant synthesizer for high quality synthesis of male and female speech. 313-316 - Ki Man Kim, Il-Whan Cha, Dae Hee Youn:
Adaptive multichannel digital filter with lattice-escalator hybrid structure. 1413-1416 - K. C. Ho, P. C. Ching, Yiu Tong Chan:
Convergence speed up in adaptive time delay estimation. 1417-1420 - Rolf Carlson, Björn Granström, Lennart Nord:
Evaluation and development of the KTH text-to-speech system on the segmental level. 317-320 - Scott C. Douglas, Teresa H.-Y. Meng:
Optimum error nonlinearities for LMS adaptation. 1421-1424 - Tetsunori Kobayashi, Hidetoshi Sekine:
Statistical properties of fluctuation of pitch intervals and its modeling for natural synthetic speech. 321-324 - P. Bragard, Geneviève Jourdain:
A fast self-optimized LMS algorithm for non-stationary identification: application to underwater equalization. 1425-1428 - George-Othon Glentis, Nicholas Kalouptsidis:
Efficient order recursive algorithms for multichannel LS filtering. 1429-1432 - Yoshinori Sagisaka:
On the prediction of global F0 shape for Japanese text-to-speech. 325-328 - Louis B. Fertig, James H. McClellan:
A dual form adaptive filter. 1433-1436 - Andrew W. Hull, W. Kenneth Jenkins:
Transform domain adaptive filtering with the chirp Z transform. 1437-1440 - Tokunbo Ogunfunmi, Allen M. Peterson:
Alternative implementations for the frequency-domain LMS adaptive filter. 1441-1444 - Shinsuke Sakai, Kazunori Muraki:
From interlingua to speech: generating prosodic information from conceptual representation. 329-332 - Xiao-Hu Yu, Zhen-yang Wu:
A novel blind adaptive equalization technique. 1445-1448 - Marian J. Macchi, Murray F. Spiegel, Karen L. Wallace:
Modeling duration adjustment with dynamic time warping. 333-336 - Ching-Yih Tseng, Lloyd J. Griffiths:
On the implementation of adaptive filters with adjustable linear constraints. 1449-1452 - M. A. Arnao, Sathyanarayan S. Rao:
A variable-step LMS based on tap weight time variation. 1453-1456 - Jacob Benesty, Pierre Duhamel:
A fast exact least mean square adaptive algorithm. 1457-1460 - S. Parthasarathy, Cecil H. Coker:
Phoneme-level parameterization of speech using an articulatory model. 337-340 - Nacer K. M'Sirdi, Jean-Luc Zarader:
Adaptive comb filters for enhancement of quasi periodic signals. 1461-1464 - Kirk P. H. Sullivan, Robert I. Damper:
A psychologically-governed approach to novel-word pronunciation within a text-to-speech system. 341-344 - John B. Kenney, Mark D. Peek:
Mean square error in an IIR adaptive echo canceller. 1465-1468 - Masanobu Abe, Kiyohiro Shikano, Hisao Kuwabara:
Cross-language voice conversion. 345-348 - B. M. Keel, Ernest G. Baxa Jr.:
Adaptive least square complex lattice clutter rejection filters applied to the radar detection of low altitude windshear. 1469-1472 - Dan Chazan, Yoav Medan, N. Sharir:
Multi microphone techniques for noise reduction in speech recognition devices. 1473-1474 - Lynn D. Wilcox, Francine R. Chen:
Application of Markov random fields to formant extraction. 349-352 - Toshiyuki Shimada, Sanjit K. Mitra:
Blind adaptive equalization for digital storage systems using timing interpolation. 1475-1478 - Tülay Adali, Sasan H. Ardalan:
Convergence and error analysis of the fixed point RLS algorithm with correlated inputs. 1479-1482 - Krishna S. Nathan, Harvey F. Silverman:
High-resolution characterization of formants in vowel-consonant transitions. 353-356 - Geoffrey A. Williamson, William A. Sethares:
Robustness to quantization errors in LMS adaptation via degree of excitation. 1483-1486 - Malcolm Slaney, Richard F. Lyon:
A perceptual pitch detector. 357-360 - Per Hedelin, Dieter Huber:
Pitch period determination of aperiodic speech signals. 361-364 - Naftali Tishby:
A dynamical systems approach to speech processing. 365-368 - Kathleen E. Cummings, Mark A. Clements:
Analysis of glottal waveforms across stress styles. 369-372 - Hiroshi Tsubokawa, Hajime Kubota, Shigeo Tsujii:
Effect of floating-point error reduction with recursive least square for parallel architecture. 1487-1490 - Paul F. Titchener:
Transient behavior of the LMS algorithm: a deterministic approach. 1491-1494 - Bin Nong, Zhongde Wang:
A new nesting scheme of PFA [prime factor algorithm]. 1495-1498 - Daniel Pak-Kong Lun, Raymond Chan, Wan-Chi Siu:
Yet a faster address generation scheme for the computation of prime factor algorithms. 1499-1502 - Raimund Meyer, Karl Schwarz:
FFT implementation on DSP-chips-theory and practice. 1503-1506 - Zong Liang Wu, Jean-Luc Schwartz, Pierre Escudier:
Modeling spectral processing in the central auditory system. 373-376 - Henrik V. Sorensen, Charles A. Katz, C. Sidney Burrus:
Efficient FFT algorithms for DSP processors using tensor product decompositions. 1507-1510 - Jechang Jeong, William J. Williams:
A fast recursive bit-reversal algorithm. 1511-1514 - Pierre Duhamel, Christine Guillemot:
Polynomial transform computation of the 2-D DCT. 1515-1518 - Keiichi Tokuda, Takao Kobayashi, Shoji Shiomoto, Satoshi Imai:
Adaptive filtering based on cepstral representation-adaptive cepstral analysis of speech. 377-380 - Henrique S. Malvar:
Efficient signal coding with hierarchical lapped transforms. 1519-1522 - Weiping Li:
Fourier analysis using adaptive AFT. 1523-1526 - Gary A. Ray:
Residue computations in rings of algebraic integers. 1527-1530 - Elio D. Di Claudio, Gianni Orlandi, Francesco Piazza:
Fast RNS DSP algorithms implemented with binary arithmetic. 1531-1534 - James F. Kaiser:
On a simple algorithm to calculate the 'energy' of a signal. 381-384 - Giancarlo Calvagno, David C. Munson Jr.:
New results on Yen's approach to interpolation from nonuniformly spaced samples. 1535-1538 - Douglas Cochran, James J. Clark:
On the sampling and reconstruction of time-warped bandlimited signals. 1539-1541 - Les Atlas, William Koolman, Patrick J. Loughlin, Randy Cole:
New nonstationary techniques for the analysis and display of speech transients. 385-388 - Mark A. Richards:
A functional minimization interpretation of fast iterative reconstruction algorithms. 1543-1546 - Yves Goussard, Guy Demoment, Jérôme Idier:
A new algorithm for iterative deconvolution of sparse spike trains. 1547-1550 - Thomas F. Quatieri, Edward M. Hofstetter:
Short-time signal representation by nonlinear difference equations. 1551-1554 - Y. T. Ting, D. G. Childers:
Speech analysis using the weighted recursive least squares algorithm with a variable forgetting factor. 389-392 - Hamid Soltanian-Zadeh, Andrew E. Yagle:
A wavenumber-domain Kalman filtering approach to one-dimensional phase retrieval. 1555-1558 - Juergen Schroeter, Peter Meyer, S. Parthasarathy:
Evaluation of improved articulatory codebooks and codebook access distance measures. 393-396 - David H. Friedman:
Implementation of a nonlinear wave-digital-filter cochlear model. 397-400 - Jean-Pierre Martens, Luc Van Immerseel:
An auditory model based on the analysis of envelope patterns. 401-404 - Mohamad K. Asi, Bahaa E. A. Saleh:
A linear periodically time-varying filter for time-frequency scaling of speech. 405-408 - E. Hardam:
High quality time scale modification of speech signals using fast synchronized-overlap-add algorithms. 409-412 - Gérard Thomas, N. Souilah:
Use of penalty function or Lagrangian multipliers for constrained deconvolution. 1559-1562 - Wooshik Kim, Monson H. Hayes:
Iterative phase retrieval using two Fourier transform intensities. 1563-1566 - Athina P. Petropulu, Chrysostomos L. Nikias:
Signal reconstruction from the phase of the bispectrum. 1567-1570 - Marco Fratti:
A new iterative algorithm for minimum-phase signal reconstruction from its autocorrelation. 1571-1574 - Nelson Morgan, Hervé Bourlard:
Continuous speech recognition using multilayer perceptrons with hidden Markov models. 413-416 - Yu Hen Hu, Homer H. M. Chern:
An efficient VLSI CORDIC array structure implementation of Toeplitz eigensystem solvers. 1575-1578 - Andrew E. Yagle:
Multichannel coupled split algorithms for non-Hermitian block Toeplitz matrices. 1579-1582 - Kaushal K. Dhar:
Fast least squares algorithm based on partitioning technique. 1583-1586 - Les T. Niles, Harvey F. Silverman:
Combining hidden Markov model and neural network classifiers. 417-420 - Amrane Houacine:
Regularized fast recursive least squares algorithms. 1587-1590 - Dirk T. M. Slock:
Reconciling fast RLS lattice and QR algorithms. 1591-1594 - Richard M. Todd, J. R. Cruz:
A class of fast covariance least squares algorithms. 1595-1598 - Weiye Ma, Dirk Van Compernolle:
TDNN labeling for a HMM recognizer. 421-423 - Ramesh A. Gopinath, C. Sidney Burrus:
Efficient computation of the wavelet transforms. 1599-1601 - Jacqueline Bertrand, Pierre Bertrand, Jean Philippe Ovarlez:
Discrete Mellin transform for signal analysis. 1603-1606 - Franz Hlawatsch, Werner Kozek, Werner Krattenthaler:
Time-frequency subspaces and their application to time-varying filtering. 1607-1610 - Werner Krattenthaler, Franz Hlawatsch:
General signal synthesis algorithms for smoothed versions of Wigner distribution. 1611-1614 - Michael A. Franzini, Kai-Fu Lee, Alex Waibel:
Connectionist Viterbi training: a new hybrid method for continuous speech recognition. 425-428 - Unto K. Laine, Toomas Altosaar:
An orthogonal set of frequency and amplitude modulated (FAM) functions for variable resolution signal analysis. 1615-1618 - John M. Cioffi:
The fast Householder filters-RLS adaptive filter. 1619-1622 - Avinash L. Ghirnikar, S. Thomas Alexander:
Stable recursive least squares filtering using an inverse QR decomposition. 1623-1626 - Kuldip K. Paliwal:
Neural net classifiers for robust speech recognition under noisy environments. 429-432 - Phillip A. Regalia, Maurice G. Bellanger:
A hybrid lattice-QR fast algorithm for least squares adaptive filtering. 1627-1630 - K. J. Ray Liu, Shih-Fu Hsieh, Kung Yao:
Recursive LS filtering using block Householder transformations. 1631-1634 - Esther Levin:
Word recognition using hidden control neural architecture. 433-436 - Peter Strobach:
Recursive Givens reduction using a second-order window. 1635-1638 - Joe Tebelskis, Alex Waibel:
Large vocabulary recognition using linked predictive neural networks. 437-440 - Ravi Prakash Ramachandran, Peter Kabal:
Synthesis of bandwidth efficient OQAM and VSB transmultiplexers. 1639-1642 - Ken-ichi Iso, Takao Watanabe:
Speaker-independent word recognition using a neural prediction model. 441-444 - Frank Fallside, H. Lucke, Tim Marsland, P. J. O'Shea, Mark Owen, Richard W. Prager, Anthony J. Robinson, N. H. Russell:
Continuous speech recognition for the TIMIT database using neural networks. 445-448 - Masanori Miyatake, Hidefumi Sawai, Yasuhiro Minami, Kiyohiro Shikano:
Integrated training for spotting Japanese phonemes using large phonemic time-delay neural networks. 449-452 - Juin-Hwey Chen:
High-quality 16 kb/s speech coding with a one-way delay less than 2 ms. 453-456 - Jean E. Menez, Claude R. Galand, Michele M. Rosso:
A 2 ms-delay adaptive code excited linear predictive coder. 457-460 - Ira A. Gerson, Mark A. Jasiuk:
Vector sum excited linear prediction (VSELP) speech coding at 8 kbps. 461-464 - Douglas J. Rahikka, Thomas E. Tremain, Vanoy C. Welch, Joseph P. Campbell Jr.:
CELP coding for land mobile radio applications. 465-468 - Isabel Trancoso, Carlos M. Ribeiro, Luís B. Almeida, Luís C. Oliveira, Jorge S. Marques:
CELP: a candidate for GSM half-rate coding. 469-472 - Redwan Salami, K. H. H. Wong, Raymond Steele, Derek G. Appleby:
Performance of error protected binary pulse excitation coders at 11.4 kb/s over mobile radio channels. 473-476 - Tomohiko Taniguchi, Fumio Amano, Shigeyuki Unagami:
Combined source and channel coding based on multimode coding. 477-480 - Dale E. Veeneman, Baruch Mazor:
An efficient code structure and search strategy for stochastic coding at 8 kb/s. 481-484 - Przemyslaw Dymarski, Nicolas Moreau, A. Vigier:
Optimal and sub-optimal algorithms for selecting the excitation in linear predictive coders. 485-488 - Hitoshi Iwamida, Shigeru Katagiri, Erik McDermott, Yoh'ichi Tohkura:
A hybrid speech recognition system using HMMs with an LVQ-trained codebook. 489-492 - Les T. Niles, Harvey F. Silverman, Marcia A. Bush:
Neural networks, maximum mutual information training, and maximum likelihood training [speech recognition]. 493-496 - Don Kimber, Marcia A. Bush, Gary N. Tajchman:
Speaker-independent vowel classification using hidden Markov models and LVQ2. 497-500 - Yuqing Gao, Taiyi Huang, Daowen Chen:
HMM-based warping in neural networks. 501-504 - Yasuhiro Komori, Kaichiro Hatazaki, Takaharu Tanaka, Takeshi Kawabata:
Combining phoneme identification neural networks into an expert system using spectrogram reading knowledge. 505-508 - Seiichi Nakagawa, Yoshimitsu Hirata:
Comparison among time-delay neural networks, LVQ2 discrete parameter HMM and continuous parameter HMM. 509-512 - Nobuo Hataoka, Akio Amano, Toshiyuki Aritsuka, Akira Ichikawa:
Large vocabulary speech recognition using neural-fuzzy and concept networks. 513-516 - Pao-Chung Chang, San-Wei Sun, Sin-Horng Chen:
Mandarin tone recognition by multi-layer perceptron. 517-520 - Aage Bendiksen, Kenneth Steiglitz:
Neural networks for voiced/unvoiced speech classification. 521-524 - Hong C. Leung, Victor W. Zue:
Phonetic classification using multi-layer perceptrons. 525-528 - Ming Zhu, Klaus Fellbaum:
A connectionist model for speaker-independent isolated word recognition. 529-532 - Yeshwant K. Muthusamy, Ronald A. Cole, Malcolm Slaney:
Speaker-independent vowel recognition: spectrograms versus cochleagrams. 533-536 - Yoshua Bengio, Régis Cardin, Renato De Mori, Yves Normandin:
A hybrid coder for hidden Markov models using a recurrent neural network. 537-540 - Yu Qing Gao, Jin-Hui Xie:
A model block-training method for HMM-based speech recognition systems. 541-544 - Abdelaziz Kriouile, Jean-François Mari, Jean-Paul Haton:
Some improvements in speech recognition algorithms based on HMM. 545-548 - Markus Falkhausen, Stephan Euler, Dietrich Wolf:
Improved training and recognition algorithms with VQ-based hidden Markov models. 549-552 - Solomon Lerner, Baruch Mazor:
Issues related to the estimation of time variant hidden Markov models [speech recognition]. 553-556 - Eleftherios D. Frangoulis:
Vector quantization of the HMM distributions: further results. 557-560 - Fu-hua Liu, Jung-Kuei Chen, Eng-Fong Huang, Chi-Shi Liu:
Discriminant methods for improving the robustness of Mandarin syllables recognition based upon hidden Markov model. 561-564 - L. A. Smith, Brian L. Scott, L. S. Lin, J. M. Newell:
Template adaptation in a hypersphere word classifier. 565-568 - K. Matrouf, Jean-Luc Gauvain, Françoise D. Néel, Joseph Mariani:
Adapting probability-transitions in DP matching process for an oral task-oriented dialogue. 569-572 - Hy Murveit, Robert C. Moore:
Integrating natural language constraints into HMM-based speech recognition. 573-576 - Marc El-Bèze, Anne-Marie Derouault:
A morphological model for large vocabulary speech recognition. 577-580 - Kenji Kita, Takeshi Kawabata, Toshiyuki Hanazawa:
HMM continuous speech recognition using stochastic language models. 581-584 - Lee-Feng Chien, Keh-Jiann Chen, Lin-Shan Lee:
An augmented chart parsing algorithm integrating unification grammar and Markov language model for continuous speech recognition. 585-588 - Shoichi Matsunaga, Shigeki Sagayama, Shigeru Homma, Sadaoki Furui:
A continuous speech recognition system based on a two-level grammar approach. 589-592 - Ajay N. Jain, Alexander H. Waibel:
Robust connectionist parsing of spoken language. 593-596 - Kjell Elenius:
Comparing a connectionist and a rule based model for assigning parts-of-speech. 597-600 - Allen L. Gorin, Stephen E. Levinson, Laura G. Miller, A. N. Gertner, Andrej Ljolje, E. R. Goldman:
On adaptive acquisition of language. 601-604 - Keikichi Hirose, Hiroya Fujisaki, Sumio Ohno, Haruhiko Mio:
Spoken word recognition for multiple speakers based on path-limited DP matching and a method for speaker normalization. 605-608 - Francisco Javier Casajús-Quirós, Luis Alfonso Hernández Gómez, Carmen García-Mateo:
Analysis and quantization procedures for a real-time implementation of a 4.8 kb/s CELP coder. 609-612 - Biing-Hwang Juang, Frank K. Soong:
Speaker recognition based on source coding approaches. 613-616 - Hiroya Fujisaki, Keikichi Hirose, Hisashi Kawai, Yasuharu Asano:
A system for synthesizing Japanese speech from orthographic text. 617-620 - Frederick Jelinek, Robert L. Mercer, Salim Roukos:
Classifying words for improved statistical language models. 621-624 - S. Sridharan, Ed Dawson, Bruce Goldburg:
Speech encryption using discrete orthogonal transforms. 1647-1650 - Sally L. Wood, Michael G. Larimore, John R. Treichler:
Modem constellation identification: a performance comparison of two methods. 1651-1654 - Ramin A. Nobakht, David E. van den Bout, J. Keith Townsend, Sasan H. Ardalan:
The joint optimization of transmitter and receiver filters for data transmission systems using mean field annealing. 1655-1658 - Hiroyasu Sano, Masafumi Hagiwara, Masao Nakagawa:
Phase synchronizer for fading channels with the modified method of least squares. 1659-1662 - Leopold Bömer, Markus Antweiler:
Binary and biphase sequences and arrays with low periodic autocorrelation sidelobes. 1663-1666 - Jörn Thielecke:
A new nonlinear equalizer for mobile radio channels. 1667-1670 - Guozhu Long, Fuyun Ling:
A new complex system identification method and its application to echo canceller fast initialization. 1671-1674 - Neil J. Bershad, S. Roy:
Performance of the 2-2 constant modulus (CM) adaptive algorithm for Rayleigh fading sinusoids in Gaussian noise. 1675-1678 - Albert M. Gottlieb, Pedro M. Crespo, James L. Dixon, T. Russell Hsing:
The DSP implementation of a new timing recovery technique for high-speed digital data transmission. 1679-1682 - Stanton B. McMillan, Brian P. Flanagan, Tom K. Doong:
Determination of the modulation type of communication signals. 1683-1686 - Jaakko Astola, Yrjö Neuvo:
A median type matched filter. 1687-1690 - John T. Stonick, Sasan H. Ardalan, J. Keith Townsend:
An improved analysis of ΣΔ modulators with bandlimited Gaussian inputs. 1691-1694 - Walter Stammler, Frank Negretti:
Mobile radio bit synchronisation and time delay measurement with a digital phase locked loop. 1695-1698 - Michael J. Ready, Richard P. Gooch:
Blind equalization based on radius directed adaptation. 1699-1702 - Masayoshi Miyagi, Toshiaki Ogawa, Iwao Sasase, Shinsaku Mori:
Suppression of CW interference and filtered noise in QPRS systems using decision-feedback filters. 1703-1706 - Zhi Ding, C. Richard Johnson Jr., Rodney A. Kennedy:
On the admissibility of blind adaptive equalizers. 1707-1710 - Margarita Cabrera, Miguel Angel Lagunas:
Simultaneous frequency and phase estimation of digital modulated signals in burst mode transmission. 1711-1714 - James E. Gaby, Stanton B. McMillan:
Automatic bit pattern analysis of communication signals. 1715-1718 - M. Barton:
An alternative realization of the SVD-based FSE. 1719-1722 - Martin Vetterli, Cormac Herley:
Wavelets and filter banks: relationships and new results. 1723-1726 - Jean-Christophe Feauveau:
A new approach for subband processing. 1727-1730 - Bor-Rong Horng, Alan N. Wilson Jr.:
Lagrange multiplier approaches to the design of two-channel perfect-reconstruction linear-phase FIR filter banks. 1731-1734 - Kambiz Nayebi, Thomas P. Barnwell III, Mark J. T. Smith:
The time domain analysis and design of exactly reconstructing FIR analysis/synthesis filter banks. 1735-1738 - Roberto H. Bamberger, Mark J. T. Smith:
Narrow band analysis of a filter bank for the directional decomposition of images. 1739-1742 - Keshab K. Parhi, Gregory S. Munson, Lai Q. Pham:
Quantization effects in high-speed pipelined recursive filters. 1743-1746 - Robert J. McAulay, Terrence G. Champion:
Improved interoperable 2.4 kb/s LPC using sinusoidal transform coder techniques. 641-643 - Christine Uhl, Odile Macchi:
When is DPCM a stable system? 1747-1750 - Wu Chou, Teresa H. Meng, Robert M. Gray:
Time domain analysis of sigma delta modulation. 1751-1754 - Y. J. Liu:
A high-quality speech coder at 600 b.p.s. 645-648 - Ping Wah Wong:
FIR digital filter with sigma-delta modulated input and impulse response. 1755-1758 - Paul E. Beckmann, Bruce R. Musicus:
Fault-tolerant round robin A/D converter system. 1759-1762 - Yan Ming Cheng, Douglas D. O'Shaughnessy:
A 450 b.p.s. vocoder with natural-sounding speech. 649-652 - Chi-Hsin Wu, Andrew E. Yagle:
Performances of autoregressive spectrum estimators based on three-term recurrence. 1763-1766 - Patrick Peterson, Philippe Jeanrenaud, Jeffrey Vandegrift:
Improving intelligibility of a 300 b/s segment vocoder. 653-656 - Neeraj Magotra, J. Bitto, Wes McCoy:
Real time implementation of multidimensional fast Fourier transforms. 1767-1770 - Gagan Mirchandani, Robert B. Pegram III, Frederic L. Stone, Peter A. Twombly:
Scheduling and simulation of neural nets on a DSP accelerator. 1771-1774 - Yoshinao Shiraki, Masaaki Honda:
Speaker adaptation algorithms based on piecewise moving adaptive segment quantization method. 657-660 - Brian L. Evans, James H. McClellan, Wallace B. McClure:
Symbolic z-transforms using DSP knowledge bases. 1775-1778 - Michele Covell:
An algorithm design environment for signal processing. 1779-1782 - Peter Kroon, Bishnu S. Atal:
Pitch predictors with high temporal resolution. 661-664 - Raimund Meyer, R. Mehling:
Architecture and performance of a new arithmetic unit for the computation of elementary functions. 1783-1786 - Jorge S. Marques, Isabel Trancoso, José M. Tribolet, Luís B. Almeida:
Improved pitch prediction with fractional delays in CELP coding. 665-668 - William A. Weeks, Brian L. Curless:
A real-time, multichannel system with parallel digital signal processors. 1787-1790 - Long-Wen Chang, Jing-Ho Lin:
Bit level systolic arrays for real time median filters. 1791-1794 - Chi-Min Liu, Bor-Shyong Yang, Chein-Wei Jen:
Parallel adaptive algorithm for moving target indicator and its VLSI array realization. 1795-1798 - Petros Maragos, Kenneth L. Young:
Fractal excitation signals for CELP speech coders. 669-672 - Ian K. Proudler, T. J. Shepherd, John G. McWhirter:
Computationally efficient QRD-based wide-band beamforming. 1799-1802 - Yoshiaki Asakawa, Katsuya Yamasaki, Akira Ichikawa:
A 2.4 KBPS speech coding method based on fuzzy vector quantization. 673-676 - Dominique Vicard, D. Degryse:
Mixed transputer/DSP architecture for modular signal processing machines. 1803-1806 - R. Settineri, Gérard Favier:
Adaptive RLS lattice filters for fast nonstationary signals. 1807-1810 - Ta-Kang Ku, C.-C. Jay Kuo:
Design and analysis of Toeplitz preconditioners. 1811-1814 - Robert W. Stewart, Roy Chapman:
Fast stable Kalman filter algorithms utilising the square root. 1815-1818 - Nacer K. M'Sirdi, Odile Macchi, Jean-Luc Zarader:
Behaviour analysis of adaptive ARMA predictors with nonstationary inputs. 1819-1822 - S. Lawrence Marple Jr.:
A fast computational algorithm for the QR-like decomposition of the modified covariance method of linear prediction. 1823 - Steve J. Young:
Competitive training in hidden Markov models [speech recognition]. 681-684 - William E. Higgins, C. Morice, Erik L. Ritman:
New shape-based interpolation technique for three-dimensional images. 1841-1844 - George Yu, William Russell, Richard M. Schwartz, John Makhoul:
Discriminant analysis and supervised vector quantization for continuous speech recognition. 685-688 - Yongjian Bao, Helmut Oswald, Eckart Fleck:
The pyramid structure and its application to binary image reconstruction from two projections. 1845-1848 - C. D. Caquineau, J.-L. Amans:
A processor architecture for the image and volume reconstruction. 1849-1852 - Xuedong Huang, Kai-Fu Lee, Hsiao-Wuen Hon:
On semi-continuous hidden Markov modeling. 689-692 - Makoto Tabei, Mitsuhiro Ueda:
Backprojection with Fourier series expansion and FFT. 1853-1856 - Christian Hamon, Christian Roux, Jean-Louis Coatrieux, René Collorec:
An analytical method for 3D tomographic reconstruction from a small number of projections using a simple interpolation scheme. 1857-1860 - Eng-Fong Huang, Frank K. Soong:
A probabilistic acoustic map based discriminative HMM training. 693-696 - Ken D. Sauer, Bede Liu:
Image reconstruction from a limited number of projections using multiple object detection/estimation. 1861-1864 - Subrata K. Das:
Supervised selection of prototypes for classification [speech recognition]. 697-700 - Berkman Sahiner, Andrew E. Yagle:
Application of time-frequency distributions to magnetic resonance imaging of non-constant flow. 1865-1868 - Neeraj Magotra, Monica M. Puri:
Synthetic aperture radar image formation. 1869-1872 - Mehrdad Soumekh:
A system model and inversion for synthetic aperture radar imaging. 1873-1876 - Yong Liu, Avideh Zakhor:
Optimal binary image design based on the branch and bound algorithm. 1877-1880 - Yen-Lu Chow:
Maximum mutual information estimation of HMM parameters for continuous speech recognition using the N-best algorithm. 701-704 - Tian-Hu Yu, Sanjit K. Mitra:
Detail-preserving impulse noise removal of images using modified dynamic programming. 1881-1884 - Mari Ostendorf, Jan Robin Rohlicek:
Joint quantizer design and parameter estimation for discrete hidden Markov models. 705-708 - Stanley J. Reeves, Russell M. Mersereau:
Regularization parameter estimation for iterative image restoration in a weighted Hilbert space. 1885-1888 - Andrej Ljolje, Yariv Ephraim, Lawrence R. Rabiner:
Estimation of hidden Markov model parameters by minimizing empirical error rate. 709-712 - Reginald L. Lagendijk, Jan Biemond, Dick E. Boekee:
Hierarchical blur identification. 1889-1892 - A. Murat Tekalp, Mehmet K. Özkan, A. Tanju Erdem:
Image modeling using higher-order statistics with application to predictive image coding. 1893-1896 - M. Ibrahim Sezan, H. Joel Trussell:
Prototype image-based constraints for set-theoretic image restoration. 1897-1900 - Allen D. Hillery, Roland T. Chin:
Iterative Wiener filters for image restoration. 1901-1904 - Natalie Fortier, Guy Demoment, Yves Goussard:
Implementation and practical comparison of two estimators of the smoothing parameter in linear image restoration. 1905-1908 - Joon Ki Paik, Aggelos K. Katsaggelos:
Image restoration using the Hopfield network with nonzero autoconnection. 1909-1912 - Gordana Pavlovic, A. Murat Tekalp:
Restoration in the presence of multiplicative noise with application to scanned photographic images. 1913-1916 - Luigi Fortuna, Giuseppe Nunnari:
Signal processing of tilt data for ground deformation modeling in volcanic areas. 1917-1920 - Pietro Burrascano, P. Lucci, Giuseppe Martinelli, Renzo Perfetti:
Shear velocity estimation by the combined use of supervised and unsupervised neural networks. 1921-1924 - Ali Özbek, Bernard C. Levy, M. Nafi Toksoz:
Inversion of conic projections with applications to inverse scattering for elastic media. 1925-1928 - Hermann Ney:
Experiments on mixture-density phoneme-modelling for the speaker-independent 1000-word speech recognition DARPA task. 713-716 - Pietro Burrascano, P. Lucci, Giuseppe Martinelli, Renzo Perfetti:
Predictive semblance estimation in seismic inversion. 1929-1932 - Daniel Burnside, Thomas W. Parks:
Amplitude bound interpolation of sonic well-log data. 1933-1936 - Chih-Ming Chen, Marwan A. Simaan:
Frequency-domain filters for suppression of multiple interferences on array data. 1937-1940 - Francisco Casacuberta, Enrique Vidal, B. Mas, Hector Rulot Segovia:
Learning the structure of HMM's through grammatical inference techniques. 717-720 - Peter Boles, Boualem Boashash:
Practical use of the cross Wigner-Ville distribution to process vibroseis VSP data. 1941-1944 - Vito Roberto, L. Gargiulo, Adriano Peron, Claudio Chiaruttini:
A knowledge-based system for geophysical interpretation. 1945-1948 - Chin-Hui Lee, Lawrence R. Rabiner, Roberto Pieraccini, Jay G. Wilpon:
Acoustic modeling of subword units for speech recognition. 721-724 - Marwan A. Simaan, Zhen Zhang:
Automatic segmentation of seismic data via knowledge-based image processing techniques. 1949-1952 - Hsiao-Wuen Hon, Kai-Fu Lee:
On vocabulary-independent speech modeling. 725-728 - Kuldip K. Paliwal:
Lexicon-building methods for an acoustic sub-word based speech recognizer. 729-732 - Andrea Monti Guarnieri, Claudio Maria Prati:
Autofocusing and residual migration of SAR images. 1953-1956 - Jong-Whan Jang, Sarah A. Rajala:
Segmentation based image coding using fractals and the human visual system. 1957-1960 - Michael K. Brown, Jay G. Wilpon:
Automatic generation of lexical and grammatical constraints for speech recognition. 733-736 - A. Tanju Erdem, A. Murat Tekalp:
Blur identification using bispectrum. 1961-1964 - Egidio P. Giachin, Aaron E. Rosenberg, Chin-Hui Lee:
Word juncture modeling using phonological rules for HMM-based continuous speech recognition. 737-740 - Nadia Baaziz, Claude Labit:
Laplacian pyramid versus wavelet decomposition for image sequence coding. 1965-1968 - Hiroyuki Morikawa, Hiroshi Harashima:
3-D structure extraction coding of image sequences. 1969-1972 - Li Deng, Vishwa Gupta, Matthew Lennig, Patrick Kenny, Paul Mermelstein:
Acoustic recognition component of an 86000-word speech recognizer. 741-744 - Serafim N. Efstratiadis, Aggelos K. Katsaggelos:
A model-based pel-recursive motion estimation algorithm. 1973-1976 - S. A. Euler, Biing-Hwang Juang, Chin-Hui Lee, Frank K. Soong:
Statistical segmentation and word modeling techniques in isolated word recognition. 745-748 - Michael T. Orchard:
Predictive motion field segmentation for image sequence coding. 1977-1980 - Kai-Fu Lee, Satoru Hayamizu, Hsiao-Wuen Hon, Cecil Huang, Jonathan Swartz, Robert Weide:
Allophone clustering for continuous speech recognition. 749-752 - Hidenobu Harasaki, Mitsuharu Yano, Takao Nishitani:
Background separation/filtering for videophone applications. 1981-1984 - Francine R. Chen:
Identification of contextual factors for pronunciation networks. 753-756 - Paul Haskell, David G. Messerschmitt:
Reconstructing lost video data in a lapped orthogonal transform based coder. 1985-1988 - Eduardo Lleida, Climent Nadeu, Enrique Monte, José B. Mariño:
Statistical feature selection for isolated word recognition. 757-760 - Yue Min Zhu, Robert Goutte, Françoise Peyrin:
On the use of 2D analytic signals for Wigner analysis of 2D real signals. 1989-1992 - Moshe Shamash, Izidor Gertner:
A new parallel algorithm for the multidimensional Fourier transform processing. 1993-1996 - Harald Singer, Taizo Umezaki, Fumitada Itakura:
Low bit quantization of the smoothed group delay spectrum for speech recognition. 761-764 - Y. Y. Choi, Irving S. Reed, Ming-Tang Shih:
The new arithmetical approach to Fourier analysis for a 2D signal. 1997-2000 - Jérôme Idier, Yves Goussard:
Markovian modeling for Bayesian multi-channel deconvolution. 2001-2004 - Harry Chang:
Transducer effect on ASR applications over the telephone network. 765-768 - Richard R. Hansen Jr., Rama Chellappa:
Empirical robust estimators for 2D noncausal autoregressive models. 2005-2008 - Tetsunori Kobayashi, Kazuhiro Watanabe:
Dependence of phonemic feature on context. 769-772 - Jerome M. Shapiro, David H. Staelin:
Multidimensional adaptive filtering via McClellan transformations. 2009-2012 - Avideh Zakhor:
Stability of reconstruction of real and complex multidimensional signals from Fourier transform magnitude. 2013-2016 - John W. Butzberger, Mari Ostendorf, Patti J. Price, Stefanie Shattuck-Hufnagel:
Isolated word intonation recognition using hidden Markov models. 773-776 - Wen-Hsien Fang, Andrew E. Yagle:
Discrete fast algorithms for two-dimensional linear prediction on a polar raster. 2017-2020 - Ahmad A. Masoud, Moustafa M. Fahmy:
A new method for designing M-D linear-phase FIR digital filters using the WKS sampling theory. 2021-2024 - Nanette M. Veilleux, Mari Ostendorf, Patti J. Price, Stefanie Shattuck-Hufnagel:
Markov modeling of prosodic phrase structure. 777-780 - Sanjit K. Mitra, Ognjan V. Shentov, Mariane R. Petraglia:
A method for fast approximate computation of discrete time transforms. 2025-2028 - Chang-Keng Lin, Bor-Shenn Jeng:
On-line recognition of handwritten Chinese characters and alphabets. 2029-2032 - Aruna Bayya, Hynek Hermansky:
Towards feature-based speech metric. 781-784 - Brian Zuerndorfer, Gregory H. Wakefield, Anthony W. England:
Recovery of fine resolution information in multispectral processing. 2033-2036 - D. Mitch Wilkes, Lester E. Lynd Jr., Janos Sztipanovits, Gabor Karsai:
The multigraph approach to parallel, distributed, structurally adaptive signal processing. 2037-2040 - K. R. Tampi, C. S. Sridhar:
Shape detection using word like image description. 2041-2043 - Yi-Teh Lee, Daniel Kahn:
Information-theoretic distortion measures for speech recognition: theoretical considerations and experimental results. 785-788 - Kris Jensen, Dimitris Anastassiou:
Spatial resolution enhancement of images using nonlinear interpolation. 2045-2048 - Sadaoki Furui:
On the use of hierarchical spectral dynamics in speech recognition. 789-792 - John Goutsias:
Unilateral approximation of Gibbs random field images. 2049-2052 - Stephen A. Zahorian, Zaki B. Nossair:
Dynamic spectral shape features for speaker-independent automatic recognition of stop consonants. 793-796 - P. Cometti, Gianni Orlandi, Francesco Piazza:
An enhanced image registration algorithm by the Burg filter. 2053-2056 - Joseph M. Scanlan, Douglas M. Chabries, Richard W. Christiansen:
A shadow detection and removal algorithm for 2-D images. 2057-2060 - Jack S. N. Jean:
A new distance measure for binary images. 2061-2064 - David Burshtein:
Joint maximum likelihood estimation of pitch and AR parameters using the EM algorithm. 797-800 - Dan Schonfeld, John Goutsias:
Robust morphological representation of binary images. 2065-2068 - Kuldip K. Paliwal:
A study of LSF representation for speaker-dependent and speaker-independent HMM-based speech recognition systems. 801-804 - Ali Mohammad-Djafari, Guy Demoment:
Estimating priors in maximum entropy image processing. 2069-2072 - Edward C. Bronson, Michael R. Taaffe, Leah H. Jamieson:
A phoneme model of English speech for the design of a parallel speech understanding system. 805-808 - Kari-Pekka Estola:
Interpolated Gaussian scale-space filters. 2073-2076 - Fernando M. Silva, Luís B. Almeida:
Speech separation by means of stationary least-squares harmonic estimation. 809-812 - Xinhua Zhuang, Hyonam Joo, Seho Oh, Yunxin Zhao, Thomas S. Huang:
An analog neural net performing multidimensional maximum entropy spectral estimation. 2077-2080 - Peter Strobach:
A neural network with Boolean output layer. 2081-2084 - M. H. Ahmad Fadzil, Tim J. Dennis:
Sample selection in subband vector quantization. 2085-2088 - Steven L. Eddins, Mark J. T. Smith:
A three-source multirate model for image compression. 2089-2092 - Chaouki Diab, Rémy Prost, Robert Goutte:
Block-adaptive subband coding of images. 2093-2096 - O. Johnsen, Ognjan V. Shentov, Sanjit K. Mitra:
A technique for the efficient coding of the upper bands in subband coding of images. 2097-2100 - Christine Podilchuk, Nikil S. Jayant, Peter Noll:
Sparse codebooks for the quantization of nondominant sub-bands in image coding. 2101-2104 - Naoto Tanabe, Nariman Farvardin:
Subband image coding using entropy-coded quantization over noisy channels. 2105-2108 - Ali N. Akansu, Manjunatha S. Kadur:
Subband coding of video with adaptive vector quantization. 2109-2112 - Jan Biemond, Frank Bosveld, Reginald L. Lagendijk:
Hierarchical subband coding of HDTV in BISDN. 2113-2116 - Marc A. Zissman, Clifford J. Weinstein, Louis D. Braida:
Automatic talker activity labeling for co-channel talker interference suppression. 813-816 - Kazunari Irie, Ryozo Kishimoto:
Adaptive subband coder design for HDTV signal transmission. 2117-2120 - Dimitrios S. Kalivas, Alexander A. Sawchuk:
Motion compensated enhancement of noisy image sequences. 2121-2124 - Thippur V. Sreenivas, Karamjeet Singh, Russell J. Niederjohn:
Spectral resolution and noise robustness in auditory modeling. 817-820 - Bor-Shenn Jeng, San-Wei Sun, Chun-Jen Lee, Tieh-Min Wu, Ming-Wen Chang:
Chinese character recognition with neural nets classifier. 2125-2128 - Thomas F. Quatieri, Robert J. McAulay:
Noise reduction using a soft-decision sine-wave vector quantizer. 821-824 - Andrew Calway, Roland Wilson:
Curve extraction in images using the multiresolution Fourier transform. 2129-2132 - Shinichi Tamura, Masami Nakamura:
Improvements to the noise reduction neural network. 825-828 - Alan C. Bovik:
Properties of multichannel texture analysis filters. 2133-2136 - T. N. Tan, Anthony G. Constantinides:
Texture analysis based on a human visual model. 2137-2140 - Yariv Ephraim:
A minimum mean square error approach for speech enhancement. 829-832 - Charles A. Bouman, Bede Liu:
Unsupervised estimation of image textures using an autoregressive model. 2141-2144 - Joon Ki Paik, Aggelos K. Katsaggelos:
Edge detection using a neural network. 2145-2148 - L. Torres, Antoni Gasull:
Temporal automatic edge detection of echocardiographic images. 2149-2152 - Y. T. Zhou:
A circular smoothing edge detector. 2153-2156 - Mati Wax:
Construction of tree structured classifiers by the MDL principle. 2157-2160 - Dirk Van Compernolle:
Switching adaptive filters for enhancing noisy and reverberant speech from microphone array recordings. 833-836 - Dongming Zhao, David G. Daut:
An efficient approach to automatic shape recognition. 2161-2164 - Victor M. Alvarado, Harvey F. Silverman:
Experimental results showing the effects of optimal spacing between elements of a linear microphone array. 837-840 - Jean-Claude Junqua, Yolande Anglade:
Acoustic and perceptual studies of Lombard speech: application to isolated-words automatic speech recognition. 841-844 - Andrew Varga, Roger K. Moore:
Hidden Markov model decomposition of speech and noise. 845-848 - Alejandro Acero, Richard M. Stern:
Environmental robustness in automatic speech recognition. 849-852 - Adoram Erell, Mitch Weintraub:
Estimation using log-spectral-distance criterion for noise-robust speech recognition. 853-856 - Brian A. Hanson, Ted H. Applebaum:
Robust speaker-independent word recognition using static, dynamic and acceleration features: experiments with Lombard and noisy speech. 857-860 - Alexander C. Loui, Anastasios N. Venetsanopoulos, Kenneth C. Smith:
Two-dimensional shape representation using morphological correlation functions. 2165-2168 - Sun-Yuan Kung, Kostas I. Diamantaras:
A neural network learning algorithm for adaptive principal component extraction (APEX). 861-864 - Pietro Burrascano, P. Lucci:
A learning rule eliminating local minima in multilayer perceptrons. 865-868 - Qiuzhen Xue, Yu Hen Hu, Paul H. Milenkovic:
Analyses of the hidden units of the multi-layer perceptron and its application in acoustic-to-articulatory mapping. 869-872 - Andrew D. Girson, Ronald D. Williams:
Articulator-based synthesis for conversational speech. 873-876 - Andrea Maccato, Rui J. P. de Figueiredo:
Structured neural network topologies with application to acoustic transients. 877-880 - Chia-Jiu Wang, Mark A. Wickert, Chwan-Hwa Wu:
Three-layer neural networks for spectral estimation. 881-884 - Ioannis Pitas:
Morphological signal decomposition. 2169-2172 - Peter J. W. Rayner, Michael R. Lynch:
Complexity reduction in Volterra connectionist modelling by consideration of output mapping. 885-888 - Craig H. Richardson, Ronald W. Schafer:
Symbolic manipulation and analysis of morphological expressions. 2173-2176 - Sanjay K. Jha, Tariq S. Durrani:
Bearing estimation using neural optimisation methods. 889-892 - Subhasis Chaudhuri, Shankar Chatterjee:
A fast method for global surface interpolation. 2177-2180 - Chen-Mi Wu, Robert Michael Owens, Mary Jane Irwin:
Distortion processing in image matching problems. 2181-2184 - Peter J. Black, Teresa H.-Y. Meng:
A hardware efficient parallel Viterbi algorithm. 893-896 - Daechul Park, Don R. Hush:
Statistical analysis of a change detector based on image modeling of difference picture. 2185-2188 - Khaled M. Elleithy, Magdy A. Bayoumi:
A formal high level synthesis approach for DSP architectures. 897-990 - William G. Bliss, Archie W. Julien:
Efficient and reliable VLSI algorithms and architectures for the discrete Fourier transform. 901-904 - Stanislav B. Kesler, Adel S. El-Fishawy:
Adaptive change detection in image sequence. 2189-2192 - Josiane Zerubia, Rama Chellappa:
Mean field approximation using compound Gauss-Markov random field for edge detection and image restoration. 2193-2196 - Teresa M. Pytosh, Alberto M. Magnani:
A new parallel 2-D FFT architecture. 905-908 - Gem-Sun Jason Young, Rama Chellappa:
Inherent ambiguities in recovering 3-D motion of a planar surface from a noisy flow field. 2197-2200 - Alan Gatherer, Teresa H.-Y. Meng:
High sampling rate adaptive decision feedback equalizers. 909-912 - M. Yan, John V. McCanny, Yi Hu:
VLSI architectures for digital image coding. 913-916 - Raymond Chan, Wan-Chi Siu:
Fast detection of ellipses using chord bisectors. 2201-2204 - Ravi Ramaswamy, Gavin Brebner, David Aspinall:
Architecture of a highly reliable systolic correlator array. 917-920 - Janusz Konrad, Eric Dubois:
Use of colour information in Bayesian estimation of 2-D motion. 2205-2208 - Daniel T. L. Lee, Hiromi T. Tanaka, Yukio Kobayashi, Fumio Kishino:
Curved surface characterization in 3-D range images. 2209-2212 - Francis Jutand, Nicolas Demassieux, Alain Artieri:
A new VLSI architecture for large kernel real time convolution. 921-924 - Haluk Derin, Sadiye Guler:
Realizations and parameter estimation for line processes. 2213-2216 - B. Yegnanarayana, R. Ramaseshan, A. Ravichandran:
An algorithm for thinning noisy images. 2217-2220 - Mario Salerno, Gian Carlo Cardarilli, Roberto Lojacono, F. Sargeni:
RNS approach to full parallel linear combinators. 925-928 - V. Ralph Algazi, R. R. R. Estes, Gary E. Ford, Phillip L. Kelly:
Progressive color block coding of bilevel scanned documents. 2221-2224 - William L. Abbott, Philip S. Bednarz, Kevin D. Fisher, John M. Cioffi:
A high-speed adaptive equalizer for magnetic disk drives. 929-932 - Arnaud E. Jacquin:
A novel fractal block-coding technique for digital images. 2225-2228 - Y. B. Yu, M. H. Chan, Anthony G. Constantinides:
Low bit rate video coding using variable block size model. 2229-2232 - Paul J. Hurst, K. Wayne Current, Iskender Agi, Eric Shieh:
A VLSI architecture for two-dimensional Radon transform computations. 933-936 - Lucas Hui:
An adaptive block truncation coding algorithm for image compression. 2233-2236 - U. Totzek, F. Matthiesen, S. Wohlleben, Tobias G. Noll:
CMOS VLSI implementation of the 2D-DCT with linear processor arrays. 937-940 - Faramarz Azadegan:
Discrete cosine transform encoding of two-dimensional processes. 2237-2240 - Basavaraj I. Pawate, George R. Doddington, Shivaling S. Mahant-Shetti, Mark G. Harward, Derek Smith:
Memory based digital signal processing. 941-944 - Michael W. Marcellin:
Transform coding of images using trellis coded quantization. 2241-2244 - Mary Jane Irwin, Robert Michael Owens:
A two-dimensional, distributed logic processor for machine vision. 945-948 - Michael Gilge:
Region-oriented transform coding (ROTC) of images. 2245-2248 - Andrew M. Sutherland, M. Campbell, Yasuo Ariki, Mervyn A. Jack:
OSPREY: a transputer based continuous speech recognition system. 949-952 - Duan-Shin Lee, Kou-Hu Tzou:
Hierarchical DCT coding of HDTV for ATM networks. 2249-2252 - Sudhir S. Dixit, James B. Nardone:
A variable bit rate layered DCT video coder for packet switched (ATM) networks. 2253-2256 - Hon Keung Kwan, Pang-Chung Tsang:
Systolic implementation of counterpropagation networks. 953-956 - Junji Suzuki, Mitsuru Nomura, Sadayasu Ono:
Comparative study of transform coding for super high definition images. 2257-2260 - Nasser M. Nasrabadi, Yushu Feng:
A dynamic finite-state vector quantization scheme. 2261-2264 - Guy R. L. Sohie:
Implementation of a parallel Jacobi algorithm on multiple DSP96002 digital signal processors. 957-960 - Daniele D. Giusto, Gianni Vernazza:
Color-image coding by an advanced vector-quantizer. 2265-2268 - Robert A. Cohen, John W. Woods:
Entropy-constrained SBPVQ for image coding. 2269-2272 - Imran Shah, Olu Akiwumi-Assani, Brian Johnson:
A transform coding chip set for image compression. 961-964 - Marcelino Veiga, José Parera, Juan Santos:
Programming DSP systems on multiprocessor architectures. 965-968 - Jens-Rainer Ohm:
Still image coding using predictive tree-VQ with sub-band decomposition. 2273-2276 - Jerrold L. Boxerman, Ho John Lee:
Variable block-sized vector quantization of grayscale images with unconstrained tiling. 2277-2280 - Joseph B. Evans, Yong Ching Lim, Bede Liu:
A high speed programmable digital FIR filter. 969-971 - Xiaonong Ran, Nariman Farvardin:
Combined VQ-DCT coding of images using interblock noiseless coding. 2281-2284 - Philippe Monet, Claude Labit:
Codebook replenishment in classified pruned tree-structured vector quantization of image sequences. 2285-2288 - Jose Chau Lee, Eric Cheval, Joe Gergen:
The Motorola 16-bit DSP ASIC core. 973-976 - Khoanh K. Truong, Russell M. Mersereau:
Structural image codebooks and the self-organizing feature map algorithm. 2289-2292 - Jean D. Mc Auliffe, Les E. Atlas, Carlos Rivera:
A comparison of the LBG algorithm and Kohonen neural network paradigm for image vector quantization. 2293-2296 - Marc Antonini, Michel Barlaud, Pierre Mathieu, Ingrid Daubechies:
Image coding using vector quantization in the wavelet transform domain. 2297-2300 - S. Alexandres, J. Morán, J. Carazo, Andrés Santos:
Parallel architecture for real-time speech recognition in Spanish. 977-980 - Til Aach, André Kaup, Rudolf Mester:
Combined displacement estimation and segmentation of stereo image pairs based on Gibbs random fields. 2301-2304 - Fabrice Heitz, Patrick Bouthemy:
Motion estimation and segmentation using a global Bayesian approach. 2305-2308 - Laurie M. Linnett, A. J. Richardson:
Texture segmentation using directional operators. 2309-2312 - Thierry M. Bernard, Patrick Garda, Bertrand Y. Zavidovique:
A neural halftoning algorithm suiting VLSI implementation. 981-984 - P. W. Fung, G. Grebbin, Yianni Attikiouzel:
Model-based region growing segmentation of textured images. 2313-2316 - Schuyler Quackenbush:
Hardware implementation of a color image decoder for remote database access. 985-988 - Kurt R. Smith, Michael I. Miller:
A Bayesian approach incorporating Rissanen complexity for learning Markov random field texture models. 2317-2320 - Sangil Park:
A real-time implementation of half-band filters to obtain 18-20 bit resolution from the DSP56ADC16 sigma-delta A/D converter. 989-992 - Zhen Zhang, Marwan A. Simaan:
Knowledge-based texture image segmentation using iterative linked quadtree splitting. 2321-2324 - M. Reha Civanlar, Hayder Radha:
Image processing performance evaluation for DSP based parallel computers with distributed frame buffers. 993-996 - Zhenyu Wu, Richard M. Leahy:
A new unsupervised hierarchical segmentation algorithm for textured images. 2325-2328 - P. W. Fung, G. Grebbin, Yianni Attikiouzel:
Contextual classification and segmentation of textured images. 2329-2332 - Badrinath Roysam, Michael I. Miller:
Stochastic representation of memoryless Boolean functions: application to boundary estimation at low contrast. 2333-2336 - M. N. Chong, John J. Soraghan, Tariq S. Durrani:
Parallel implementation and analysis of adaptive transform coding. 997-1000 - Deng Wenbiao, Zheng Bao:
Tomographic imaging algorithm based on circular arc projections. 2337-2340 - Doug Young Suh, Russell M. Mersereau, Robert L. Eisner, Roderic I. Pettigrew:
A system for knowledge-based boundary detection of cardiac magnetic resonance image sequences. 2341-2344 - Graham C. Freeland, Tariq S. Durrani:
IFS fractals and the wavelet transform. 2345-2348 - J. S. McGarrity, John J. Soraghan, Tariq S. Durrani:
A fast implementation of the ESPRIT algorithm. 1001-1004 - Lixin Wang, Jerry M. Mendel:
Cumulant-based parameter estimation using neural networks. 2369-2372 - David Lee Sherman, Michael D. Zoltowski:
Frequency and biphase estimation of quadratically coupled sinusoids using cumulant projections. 2373-2376 - Nelson Morgan, James Beck, Eric Allman, Joachim Beer:
RAP: a ring array processor for multilayer perceptron applications. 1005-1008 - Zhenya He, Haibo Li:
Nonlinear predictive image coding with a neural network. 1009-1012 - Sohail A. Dianat, Raghuveer M. Rao:
Fast algorithms for bispectral reconstruction of two-dimensional signals. 2377-2379 - Chrysostomos L. Nikias, Fang Liu:
Bicepstrum computation based on second- and third-order statistics with applications. 2381-2385 - Alfons A. J. de Lange, Ed F. Deprettere, Alle-Jan van der Veen, Jichun Bu:
Real time application of the floating point pipeline CORDIC processor in massive-parallel pipelined DSP algorithms. 1013-1016 - Jitendra K. Tugnait:
Identification of noncausal ARMA models of non-Gaussian processes using higher-order statistics. 2387-2390 - Doron Kletter, Hagit Messer:
Optimal detection of a random multitone signal and its relation to bispectral analysis. 2391-2394 - KuoJuey Ray Liu, Kung Yao:
Spectral decomposition via systolic triarray based on QR iteration. 1017-1020 - Gaetano Scarano, Roberto Cusani, Alberto Laurenti:
Spectral analysis by mixtures of higher order moments. 2395-2398 - John E. Whelchel, John O'Malley, William J. Rinard, James F. McArthur:
The systolic phase rotation FFT-a new algorithm and parallel processor architecture. 1021-1024 - Maria Rangoussi, Georgios B. Giannakis:
FIR modeling using log-bispectra: weighted least-squares algorithms and performance analysis. 2399-2402 - Jenq-Neng Hwang, James A. Ritcey:
Systolic architectures for radar CFAR detectors. 1025-1028 - Roger F. Dwyer:
The extraction of range and Doppler from fourth-order spectra. 2403-2406 - S. W. Nam, Sung Bae Kim, Edward J. Powers:
On the identification of a third-order Volterra nonlinear system using a frequency-domain block RLS adaptive algorithm. 2407-2410 - Y. T. Hwang, Y. H. Hu:
Parameterized dependence graph and its applications to multi-stage systolic mapping procedure. 1029-1032 - Yingbo Hua, Tapan K. Sarkar:
Estimation of spatial and temporal parameters of multiple transient signals. 2411-2414 - Petar M. Djuric, Steven M. Kay:
Predictive probability as a criterion for model selection. 2415-2418 - Souheil F. Odeh, John R. Deller Jr.:
A systolic algorithm for adaptive set membership identification. 2419-2422 - Nasser Khalili, Joe K. Hammond:
Cepstral analysis of systems with dispersion. 2423-2426 - Jean-Pierre Le Cadre, Patrice Ravazzola:
Model reduction and wideband analysis [underwater acoustic application]. 2427-2430 - Rajinder Jit Singh, John V. McCanny, Roger F. Woods:
Pipelined two-port adaptor for wave digital filtering. 1033-1036 - Kiyohito Tokuda, Yumi Takizawa, Satoru Shimizu, Atsushi Fukasawa:
Model order selection method for multiple structured time sequence signals. 2431-2434 - Michael J. Paradie, S. Hamid Nawab:
The classification of ringing sounds. 2435-2438 - James L. Lansford, Philip R. Kennedy, James E. Schroeder:
Robust models for neural signal processing. 2439-2441 - Tongxin Lu, Mahmood R. Azimi-Sadjadi, A. R. Rostampour:
Skew-pipeline and interleaved pipeline structures for 2-D recursive filtering. 1037-1040 - James D. George, Phillip L. Ainsleigh:
A constrained Newton algorithm for maximum likelihood modeling of sums of exponentials, in noise. 2443-2446 - Keshab K. Parhi:
High-speed architectures for dynamic programming problems. 1041-1044 - Bock-Sim Chia, Delores M. Etter:
Adaptive modelling of sparse systems. 2447-2450 - Leon Cohen, Chongmoon Lee:
Instantaneous bandwidth for signals and spectrogram. 2451-2454 - Robert B. Feinberg:
Vectorizing IIR filters. 1045-1048 - Patrick Flandrin, Olivier Rioul:
Affine smoothing of the Wigner-Ville distribution. 2455-2458 - Thomas W. Parks, Ram G. Shenoy:
Time-frequency concentrated basis functions. 2459-2462 - Haiguang Chen, Gloria Faye Boudreaux-Bartels:
On application of two-dimensional AR modeling of Wigner distribution for efficient coding and effective filtering. 2463-2466 - Aihua Li, Sayfe Kiaei:
VLSI design of multi-rate arrays for DSP algorithm. 1049-1052 - Graeme Jones, Boualem Boashash:
Instantaneous frequency, instantaneous bandwidth and the analysis of multicomponent signals. 2467-2470 - Byung Moo Kim, Larry P. Heck:
Automatic design of parallel implementations of DSP algorithms. 1053-1056 - Moeness G. Amin:
High temporal resolution estimators through reduced rank periodograms. 2471-2474 - David I. Feinstein, Fred B. Holt:
Estimating the parameters for an exponential process via a Hough transform. 2475-2478 - Y. S. Cho, Sung Bae Kim, Edward J. Powers:
Time-frequency analysis using AR models with variable forgetting factors. 2479-2482 - Dominique Genin, Paul N. Hilfinger, Jan M. Rabaey, Chris J. Scheers, Hugo De Man:
DSP specification using the Silage language. 1056-1060 - Arun Kumar, Daniel R. Fuhrmann:
The Frazier-Jawerth transform. 2483-2486 - Stephen M. Kafka:
An assembly source level global compacter for digital signal processors. 1061-1064 - Thomas E. Koczwara, Douglas L. Jones:
On mask selection for time-varying filtering using the Wigner distribution. 2487-2490 - R. Kumar:
A novel multi-stage estimation of signal parameters. 2491-2494 - Kevin W. Leary, William Waddington:
DSP/C: a standard high level language for DSP and numeric processing. 1065-1068 - Dominique Garreau:
Multiscale inverse filtering. 2495-2498 - Sze Fong Yau, Yoram Bresler:
Parameter estimation of superimposed signals by dynamic programming. 2499-2502 - Aníbal R. Figueiras-Vidal, Domingo Docampo-Amoedo, José R. Casar Corredera, Antonio Artés-Rodríguez:
On using the adaptive delay filter in sparse identification and extensions to sparse deconvolution and sinusoid detection. 2503-2506 - Amir Hekmatpour, Scott R. Powell, Paul M. Chau:
KINFIDA: an object-oriented model-based digital filter design assistant. 1069-1072 - Izidor Gertner:
Optimal detection and separation of chirp signals. 2507-2510 - Kurt Baudendistel, James H. McClellan:
Code generation for the AT&T DSP32. 1073-1076 - Olivier Besson, Francis Castanie:
Doppler frequency estimator performance analysis. 2511-2514 - Pamela A. Nielsen, John B. Thomas:
Detection of a constant signal using the bi-covariance function. 2515-2518 - Alois C. Knoll, Rupert C. Nieberle:
CADiSP-a graphical compiler for the programming of DSP in a completely symbolic way. 1077-1080 - L. S. Biradar, V. Umapathi Reddy:
SVD based information theoretic criteria for detection of the number of sinusoids. 2519-2522 - Taikang Ning, Susan M. Whiting:
Power spectrum estimation via orthogonal transformation. 2523-2526 - Stewart G. Smith, Ralph W. Morgan, Julian G. Payne:
A high-level ASIC design tool for digital signal processing. 1081-1084 - Leland B. Jackson, Haiguang Chen:
Two dimensional AR modeling by autocorrelation fitting. 2527-2530 - Patrick L. Combettes, H. Joel Trussell:
New methods for the synthesis of set theoretic estimates [digital signal processing]. 2531-2534 - Toshiyuki Takezawa, Katsuhiko Shirai:
Interactive design environment of VLSI architecture for digital signal processing. 1085-1088 - Mostafa Kaveh, Antonios Bassias:
Threshold extension based on a new paradigm for MUSIC-type estimation. 2535-2538 - Carol L. DiMonte, K. S. Arun:
Tracking the frequencies of superimposed time-varying harmonics. 2539-2542 - Vijay K. Jain:
Statistical error analysis of VLSI architectures: methodology and a case study. 1089-1092 - Yu Liu, John J. Soraghan, Tariq S. Durrani:
Detection of number of harmonics by maximum eigenvalue varied rate criteria. 2543-2546 - Akihiko Sugiyama, Fumie Hazu, Masahiro Iwadare, Takao Nishitani:
Adaptive transform coding with an adaptive block size (ATC-ABS). 1093-1096 - Thierry Pitarque, Gérard Alengrin, André Ferrari:
Spectral estimation methods avoiding eigenvector decomposition. 2547-2550 - Alireza Khotanzad, Jinn-Her Lu:
Non-parametric prediction of AR processes using neural networks. 2551-2554 - W. R. Th. ten Kate, Leon Maria van de Kerkhof, F. F. M. Zijderveld:
Digital audio carrying extra information. 1097-1100 - Sharad Singhal:
High quality audio coding using multipulse LPC. 1101-1104 - Jim E. Schroeder, John E. Hershey:
Suboptimal robust sinusoidal frequency estimation. 2555-2558 - Ernst Eberlein, Heinz Gerhäuser, Stefan Krägeloh:
Audio codec for 64 kbit/sec (ISDN channel)-requirements and results. 1105-1108 - Detlev K. Seidel, Patricia Davies:
Modeling nonlinear systems by using nonlinear autoregressive moving average models. 2559-2562 - Wai-Yip Chan, Allen Gersho:
High fidelity audio transform coding with vector quantization. 1109-1112 - Akira Sano, Hiroyuki Tsuji:
Optimal determination of number of sinusoids in SVD-based spectral emission. 2563-2566 - Hiroshi Kanai, Noriyoshi Chubachi, Ken'iti Kido, Yoshiro Koiwa, Takehiko Takagi, Junichi Kikuchi, Tamotsu Takishima:
A new approach to time dependent AR modeling of multi-frame signals and its application to diagnosis of myocardial infarction. 2567-2570 - Shigenobu Minami, Osamu Okada:
Stereophonic ADPCM voice coding method. 1113-1116 - Shih-Fu Hsieh, K. J. Ray Liu, Kung Yao:
Applications of truncated QR methods to sinusoidal frequency estimation. 2571-2574 - Fu Li, Richard J. Vaccaro, Donald W. Tufts:
Unified performance analysis of subspace-based estimation algorithms. 2575-2578 - Grant A. Davidson, Louis D. Fielder, Mike Antill:
High-quality audio transform coding at 128 kbits/s. 1117-1120 - Lin Yin, Jaakko Astola, Yrjö Neuvo:
Application of median operation in robust estimation of AR model. 2579-2582 - Samuel Itzikowitz, Amir Averbuch:
General smoothing techniques for estimating deterministic sinusoidal frequencies from noisy data. 2583-2586 - Stephen A. Zahorian, Subhashri Venkat:
Vowel articulation training aid for the deaf. 1121-1124 - Ronald D. DeGroat, L. R. Hunt, Darel A. Linebarger:
Discrete-time nonlinear systems modeling. 2587-2590 - James M. Kates:
Feedback cancellation in hearing aids. 1125-1128 - Huili Wang, Gregory H. Wakefield:
Non-asymptotic performance analysis of eigenstructure spectral methods. 2591-2594 - Zoran B. Banjanin, J. R. Cruz, Dusan S. Zrnic:
Eigendecomposition methods for frequency estimation: a unified approach. 2595-2598 - C. Adnet, N. Martin:
Unified and intelligent spectral analysis. 2599-2602 - Theagenis J. Abatzoglou:
Frequency superresolution performance of the constrained total least squares method. 2603-2606 - Lonnie C. Ludeman:
Nonlinear formulas for interpolation, extrapolation, and estimation of single and multiple sinusoids. 2607-2610 - Qiansheng Cheng:
Parameter estimation of exponential models for spectra. 2611-2614 - Patrick M. Zurek, Julie E. Greenberg, Patrick Peterson:
Sensitivity to design parameters in an adaptive-beamforming hearing aid. 1129-1132 - Shoji Makino, Yutaka Kaneda:
Acoustic echo canceller algorithm based on the variation characteristics of a room impulse response. 1133-1136 - W. Ferzali, John G. Proakis:
Adaptive SVD algorithm for covariance matrix eigenstructure computation. 2615-2618 - S. Unnikrishna Pillai:
Generalization of maximum entropy spectrum extension method. 2619-2622 - Hong Fan, Ram Vemuri:
Robust adaptive algorithms for active noise and vibration control. 1137-1140 - Rafid A. Sukkar, Joseph L. LoCicero:
Iterative pole-zero estimation with a robust error criterion. 2623-2626 - Abdelhak M. Zoubir, Johann F. Böhme:
Optimization of sensor positions in knock detection using relevancy tests of sensor output signals. 2627-2630 - Braham Himed, Donald D. Weiner:
Compensation for mutual coupling effects in direction finding. 2631-2634 - Björn E. Ottersten, Mats Viberg:
Asymptotic robustness of sensor array processing methods. 2635-2638 - Steven L. Gay, Richard J. Mammone:
Fast converging subband acoustic echo cancellation using RAP on the WE DSP16A. 1141-1144 - Hamid Krim, John H. Cozzens, John G. Proakis:
On spatial smoothing and linear prediction. 2639-2642 - Birger Hätty:
Recursive least squares algorithms using multirate systems for cancellation of acoustical echoes. 1145-1148 - Xiao-Liang Xu, Kevin M. Buckley:
A comparison of element and beam space spatial-spectrum estimation for multiple source clusters. 2643-2646 - A. Lee Swindlehurst, Thomas Kailath:
An analysis of subspace fitting algorithms in the presence of sensor errors. 2647-2650 - James Parry:
Microphone arrays for desktop computers and speech recognition. 1149-1151 - David R. Farrier:
Direction of arrival estimation by subspace methods. 2651-2654 - Kenneth Steiglitz:
A filter compiler for digital sound synthesis. 1153-1156 - Jean-François Cardoso:
Eigen-structure of the fourth-order cumulant tensor with application to the blind source separation problem. 2655-2658 - Chris Chafe:
Pulsed noise in self-sustained oscillations of musical instruments. 1157-1160 - Sing H. So, Allan O. Steinhardt:
Stabilized complex downdating in adaptive beamforming. 2659-2662 - Jar-Ferr Yang, Hui-Ju Lin, Jyh-Chern Ho:
A robust and high resolution adaptive eigensubspace algorithm for nonstationary environments. 2663-2666 - Robert T. Schumacher, Chris Chafe:
Characterization of aperiodicity in nearly periodic signals. 1161-1164 - Stephan V. Schell, William A. Gardner:
Signal-selective high-resolution direction finding in multipath. 2667-2670 - Michael W. Hoffman, Xiao-Liang Xu, Kevin M. Buckley:
Eigenspace based spatial-spectrum estimation for multiple beam antennas. 2671-2674 - Yong Soo Cho, Sung Bae Kim, Elmer L. Hixson, Edward J. Powers:
Nonlinear distortion analysis using digital higher-order coherence spectra. 1165-1168 - Boaz Porat, Benjamin Friedlander:
Direction finding algorithms based on high-order statistics. 2675-2678 - John Lane, Sangil Park:
An adaptive IIR phase measurement structure for estimation of multiple sinusoids. 1169-1172 - Juha Backman, Matti Karjalainen:
Audio and ultrasonic transducers based on electrothermomechanical film (ETMF). 1173-1776 - Mark A. Randolph:
A data-driven method for discovering and predicting allophonic variation. 1177-1180 - K. R. Raghavan, Sathyanarayan S. Rao, Gregory H. Wakefield, Purusottam Mookerjee:
An eigenstructure based nonparametric algorithm to estimate the frequencies of two sinusoids in white noise. 2679-2682 - Sophocles J. Orfanidis:
Adaptive structured eigenvector methods. 2683-2686 - Bhaskar D. Rao, K. V. S. Hari:
Effect of spatial smoothing on the performance of noise subspace methods. 2687-2690 - Tsung-Ching Liu, Barry D. Van Veen:
Multiple window based minimum variance broadband spatial spectrum estimation. 2691-2694 - Joseph M. Francos, A. Zvi Meiri, Boaz Porat:
On a wold-like decomposition of 2-D discrete random fields. 2695-2698