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ORCIDICASSP 1994: Adelaide, South Australia, Australia
- Proceedings of ICASSP '94: IEEE International Conference on Acoustics, Speech and Signal Processing, Adelaide, South Australia, Australia, April 19-22, 1994. IEEE Computer Society 1994, ISBN 0-7803-1775-0
Volume 1
Speech Enhancement
- Srinivas Nandkumar, John H. L. Hansen:
Speech enhancement based on a new set of auditory constrained parameters. 1-4 - Gary H. Whipple:
Low residual noise speech enhancement utilizing time-frequency filtering. 5-8 - Athina P. Petropulu, Suresh Subramaniam:
Cepstrum based deconvolution for speech dereverberation. 9-12 - Hamid Sheikhzadeh, Hossein Sameti, Li Deng, Robert L. Brennan:
Comparative performance of spectral subtraction and HMM-based speech enhancement strategies with application to hearing and design. 13-16 - Anthony Teolis, John J. Benedetto:
Noise suppression using a wavelet model. 17-20 - Subrata K. Das, Arthur Nádas, David Nahamoo, Michael Picheny:
Adaptation techniques for ambience and microphone compensation in the IBM Tangora speech recognition system. 21-24 - Michael I. Savic, Huiqin Gao, Jeffrey S. Sorensen:
Co-channel speaker separation based on maximum-likelihood deconvolution. 25-28 - Michael A. Ramalho, Richard J. Mammone:
A new speech enhancement technique with application to speaker identification. 29-32
Acoustic/Phonetic Modelling
- Les T. Niles:
Acoustic modeling for speech recognition based on spotting of phonetic units. 33-36 - Tony Robinson, Mike Hochberg, Steve Renals:
IPA: improved phone modelling with recurrent neural networks. 37-40 - R. N. V. Sitaram, Thippur V. Sreenivas:
Phoneme recognition in continuous speech using large inhomogeneous hidden Markov models. 41-44 - Li Deng, Don X. Sun:
Phonetic classification and recognition using HMM representation of overlapping articulatory features for all classes of English sounds. 45-48 - S. Krishnan, P. V. S. Rao:
Segmental phoneme recognition using piecewise linear regression. 49-52 - Basavaraj I. Pawate, Eric M. Dowling:
A new method for segmenting continuous speech. 53-56 - Yifan Gong, Jean Paul Haton:
Stochastic trajectory modeling for speech recognition. 57-60 - Antonio M. Peinado
, José C. Segura, Antonio J. Rubio, M. Carmen Benítez:
Using multiple vector quantization and semicontinuous hidden Markov models for speech recognition. 61-64 - Nelly Suaudeau, Régine André-Obrecht:
An efficient combination of acoustic and supra-segmental informations in a speech recognition system. 65-68 - Takashi Yoshimura, Satoru Hayamizu, Kazuyo Tanaka:
Word accent patterns modelling by concatenation of mora hidden Markov models. 69-72 - David B. Grayden, Michael S. Scordilis:
Phonemic segmentation of fluent speech. 73-76 - Ara Samouelian:
Knowledge based approach to consonant recognition. 77-80
Telephone Based Speech Recognition and Databases
- Jared Bernstein, Kelsey Taussig, John J. Godfrey:
Macrophone: an American English telephone speech corpus for the Polyphone project. 81-84 - Mitchel Weintraub, Leonardo Neumeyer:
Constructing telephone acoustic models from a high-quality speech corpus. 85-88 - Thomas Staples, Joseph Picone, Nozomi Arai:
The voice across Japan database-the Japanese language contribution to Polyphone. 89-92 - Ronald A. Cole, David G. Novick, Daniel C. Burnett, Brian Hansen, Stephen Sutton, Mark Fant:
Towards automatic collection of the US census. 93-96 - J. Bruce Millar, Julie Vonwiller, Jonathan Harrington, Phillip Dermody:
The Australian National Database of Spoken Language. 97-100 - Kazuhiro Kondo, Joseph Picone, Barbara Wheatley:
A comparative analysis of Japanese and English digit recognition. 101-104 - Eric R. Buhrke, Régis Cardin, Yves Normandin, Mazin G. Rahim, Jay G. Wilpon:
Application of vector quantized hidden Markov modeling to telephone network based connected digit recognition. 105-108 - Pedro J. Moreno, Richard M. Stern:
Sources of degradation of speech recognition in the telephone network. 109-112 - John S. Garofolo, Tony Robinson, Jonathan G. Fiscus:
The development of file formats for very large speech corpora: SPHERE and SHORTEN. 113-116 - Yu-Hung Kao, Charles T. Hemphill, Barbara Wheatley, Raja Rajasekaran:
Toward vocabulary independent telephone speech recognition. 117-120 - Ove Andersen, Paul Dalsgaard, William J. Barry:
On the use of data-driven clustering technique for identification of poly- and mono-phonemes for four European languages. 121-124
Speaker Recognition
- Tomoko Matsui, Sadaoki Furui:
Speaker adaptation of tied-mixture-based phoneme models for text-prompted speaker recognition. 125-128 - Khaled T. Assaleh, Richard J. Mammone:
Robust cepstral features for speaker identification. 129-132 - Julian P. Eatock, John S. D. Mason:
A quantitative assessment of the relative speaker discriminating properties of phonemes. 133-136 - Shoji Hayakawa, Fumitada Itakura:
Text-dependent speaker recognition using the information in the higher frequency band. 137-140 - Michael A. Lund, Chung-Tshuy Lee, Chung-Chieh Lee, Robert W. Bossemeyer:
A distributed decision approach to speaker verification. 141-144 - Herbert Gish, Michael Schmidt, Angela Mielke:
A robust, segmental method for text independent speaker identification. 145-148 - Jean-Luc Le Floch, Claude Montacié, Marie-José Caraty:
Investigations on speaker characterization from Orphee system techniques. 149-152 - Jayant M. Naik, David M. Lubensky:
A hybrid HMM-MLP speaker verification algorithm for telephone speech. 153-156 - Jeffrey Sorensen, Michael I. Savic:
Hierarchical pattern classification for high performance text-independent speaker verification systems. 157-160 - Lynn Wilcox, Francine Chen, Don Kimber, Vijay Balasubramanian:
Segmentation of speech using speaker identification. 161-164 - Kevin R. Farrell, Richard J. Mammone:
Speaker identification using neural tree networks. 165-168 - Johan Schalkwyk, Etienne Barnard, Jeffrey R. Sachs:
Detecting an imposter in telephone speech. 169-172
Speech Coding Techniques
- Yoshinori Tanaka, Hisanari Kimura:
Low-bit-rate speech coding using a two-dimensional transform of residual signals and waveform interpolation. 173-176 - Victoria E. Sánchez, José L. Pérez-Córdoba, Juan M. López-Soler, Antonio J. Rubio:
Transform trellis coded quantization of speech using small frame sizes. 177-180 - Gao Yang, Henri Leich:
High-quality harmonic coding at very low bit rates. 181-184 - Jes Thyssen, Henrik Nielsen, Steffen Duus Hansen:
Non-linear short-term prediction in speech coding. 185-188 - Craig R. Watkins, Sam Crisafulli, Robert R. Bitmead, Robert Orsi:
Variable bit rate ADPCM via arithmetic coding. 189-192 - Roch Lefebvre, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul:
High quality coding of wideband audio signals using transform coded excitation (TCX). 193-196 - Keiichi Tokuda, Hidetoshi Matsumura, Takao Kobayashi, Satoshi Imai:
Speech coding based on adaptive mel-cepstral analysis. 197-200 - Stan A. McClellan, Jerry D. Gibson:
Spectral entropy: an alternative indicator for rate allocation? 201-204 - William R. Gardner, Bhaskar D. Rao:
Mixed-phase AR models for voiced speech and perceptual cost functions. 205-208 - Sun-Won Park:
Speech compression using ARMA model and wavelet transform. 209-212
Adaptation and Training Techniques
- Jesús Esteban Díaz Verdejo, José C. Segura, Pedro García-Teodoro, Antonio J. Rubio:
SLHMM: a continuous speech recognition system based on Alphanet-HMM. 213-216 - Ivica Rogina, Alex Waibel:
Learning state-dependent stream weights for multi-codebook HMM speech recognition systems. 217-220 - Qiang Huo, Chorkin Chan, Chin-Hui Lee:
Bayesian learning of the SCHMM parameters for speech recognition. 221-224 - Finn Tore Johansen, Magne Hallstein Johnsen:
Non-linear input transformations for discriminative HMMs. 225-228 - Yoshihiko Gotoh, Michael M. Hochberg, Harvey F. Silverman:
Using MAP estimated parameters to improve HMM speech recognition performance. 229-232 - Abdelhamid Mellouk, Patrick Gallinari:
Discriminative training for improved neural prediction systems. 233-236 - Barbara Wheatley, Kazuhiro Kondo, Wallace W. Anderson, Yeshwant K. Muthusamy:
An evaluation of cross-language adaptation for rapid HMM development in a new language. 237-240 - Shigeki Okawa, Tetsunori Kobayashi, Katsuhiko Shirai:
Automatic training of phoneme dictionary based on mutual information criterion. 241-244 - Tetsuo Kosaka, Shigeki Sagayama:
Tree-structured speaker clustering for fast speaker adaptation. 245-248 - Yasunaga Miyazawa, Jun-ichi Takami, Shigeki Sagayama, Shoichi Matsunaga:
All-phoneme ergodic hidden Markov network for unsupervised speaker adaptation. 249-252 - Yasuo Ariki, Keisuke Doi:
Phoneme recognition improvement by restricting training section in concatenated HMM training. 253-256
CELP Coding I
- Jörg-Martin Müller, Bertram Wächter:
A codec candidate for the GSM half rate speech channel. 257-260 - Michel Mauc, Geneviève Baudoin, Milan Jelinek:
Complexity reduction for FS-1016 with multistage search. 261-264 - Kumar Swaminathan, Kalyan Ganesan, Yi-Sheng Wang, Prabhat K. Gupta:
Speech and channel codec candidate for the half rate digital cellular channel. 265-268 - Kazunori Ozawa, Masahiro Serizawa, Toshiki Miyano, Toshiyuki Nomura:
M-LCELP speech coding at 4 kbps. 269-272 - Susan Yim, Dipanjan Sen, W. Harvey Holmes:
Comparison of ARMA modelling methods for low bit rate speech coding. 273-276 - Yoshiaki Asakawa, Hidetoshi Sekine, Makoto Takashima, Nobuyoshi Ishikawa, Toshiyuki Matsuda, Tom Okamoto, Ryujiro Muramatsu:
A 5.6 kb/s speech codec using a pulse codebook and improved Viterbi decoding. 277-280 - Luca Cellario, Daniele Sereno, Mario Giani, Peter Blöcher, Karl Hellwig:
A VR-CELP codec implementation for CDMA mobile communications. 281-284 - David E. Ray, Douglas J. Rahikka:
Reed-Solomon coding for CELP EDAC in land mobile radio. 285-288
Speaker and Language Recognition
- Kay M. Berkling, Takayuki Arai, Etienne Barnard:
Analysis of phoneme-based features for language identification. 289-292 - Lori Lamel, Jean-Luc Gauvain:
Language identification using phone-based acoustic likelihoods. 293-296 - Kung-Pu Li:
Automatic language identification using syllabic spectral features. 297-300 - Roger C. F. Tucker, Michael J. Carey, Eluned S. Parris:
Automatic language identification using sub-word models. 301-304 - Marc A. Zissman, Elliot Singer:
Automatic language identification of telephone speech messages using phoneme recognition and N-gram modeling. 305-308 - Chintana Griffin, Tomoko Matsui, Sakaoki Furui:
Distance measures for text-independent speaker recognition based on MAR model. 309-312 - Mark E. Forsyth, Mervyn A. Jack:
Discriminating semi-continuous HMM for speaker verification. 313-316 - Jonathan Foote, Harvey F. Silverman:
A model distance measure for talker clustering and identification. 317-320 - Lawrence G. Bahler, Jack E. Porter, Alan L. Higgins:
Improved voice identification using a nearest-neighbor distance measure. 321-324 - Chi-Shi Liu, Chin-Hui Lee, Biing-Hwang Juang, Aaron E. Rosenberg:
Speaker recognition based on minimum error discriminative training. 325-328 - Richard Ricart, Jim Cupples, Laurie Fenstermacher:
Speaker recognition in tactical communications. 329-332 - Yeshwant K. Muthusamy, Neena Jain, Ronald A. Cole:
Perceptual benchmarks for automatic language identification. 333-336
Speech Understanding and Language Modelling I
- Clifford J. Weinstein:
Demonstrations and applications of spoken language technology: highlights and perspectives from the 1993 ARPA Spoken Language Technology and Applications Day. 337-340 - Sunil Issar, Wayne H. Ward:
Unanswerable queries in a spontaneous speech task. 341-344 - Monika Woszczyna, Naomi Aoki-Waibel, Finn Dag Buø, Noah Coccaro, Keiko Horiguchi, Thomas Kemp, Alon Lavie, Arthur E. McNair, Thomas Polzin, Ivica Rogina, Carolyn P. Rosé, Tanja Schultz, Bernhard Suhm, Masaru Tomita, Alex Waibel:
JANUS 93: towards spontaneous speech translation. 345-348 - Douglas D. O'Shaughnessy:
Correcting complex false starts in spontaneous speech. 349-352 - Helmut Lucke:
Reducing the computational complexity for inferring stochastic context-free grammar rules from example text. 353-356 - Thomas Kuhn, Heinrich Niemann, Ernst Günter Schukat-Talamazzini:
Ergodic hidden Markov models and polygrams for language modeling. 357-360 - Jerry H. Wright, Gareth J. F. Jones, Harvey Lloyd-Thomas:
A robust language model incorporating a substring parser and extended n-grams. 361-364 - Finn Dag Buø, Thomas Polzin, Alex Waibel:
Learning complex output representations in connectionist parsing of spoken language. 365-368
Word Spotting and Other Topics
- Yoshiaki Itoh, Jiro Kiyama, Ryuichi Oka:
Sentence spotting applied to partial sentences and unknown words. 369-372 - Hervé Bourlard, Bart D'hoore, Jean-Marc Boite:
Optimizing recognition and rejection performance in wordspotting systems. 373-376 - David A. James, Steve J. Young:
A fast lattice-based approach to vocabulary independent wordspotting. 377-380 - Philippe Jeanrenaud, Man-Hung Siu, Jan Robin Rohlicek, Marie Meteer, Herbert Gish:
Spotting events in continuous speech. 381-384 - John W. McDonough, Kenney Ng, Philippe Jeanrenaud, Herbert Gish, Jan Robin Rohlicek:
Approaches to topic identification on the switchboard corpus. 385-388 - Richard Lippmann, Eric I. Chang, Charles R. Jankowski Jr.:
Wordspotter training using figure-of-merit back propagation. 389-392 - Rafid A. Sukkar:
Rejection for connected digit recognition based on GPD segmental discrimination. 393-396 - Takashi Otsuki, Akinori Ito, Shozo Makino, Teruhiko Otomo:
The performance prediction method on sentence recognition system using a finite state automaton. 397-400 - Kari Torkkola:
New ways to use LVQ-codebooks together with hidden Markov models. 401-404 - Jianmin Li, Ditang Fang:
Parallel distributed binary mapping models for speech recognition. 405-408
Robust Speech Recognition
- Juan Arturo Nolazco-Flores, Steve J. Young:
Continuous speech recognition in noise using spectral subtraction and HMM adaptation. 409-412 - Sahar E. Bou-Ghazale, John H. L. Hansen:
Duration and spectral based stress token generation for HMM speech recognition under stress. 413-416 - Leonardo Neumeyer, Mitchel Weintraub:
Probabilistic optimum filtering for robust speech recognition. 417-420 - Joachim Koehler, Nelson Morgan, Hynek Hermansky, Hans-Günter Hirsch, Grace Tong:
Integrating RASTA-PLP into speech recognition. 421-424 - Hidefumi Kobatake, Yousuke Matsunoo:
Degraded word recognition based on segmental signal-to-noise ratio weighting. 425-428 - Johan Smolders, Tom Claes, Gert Sablon, Dirk Van Compernolle:
On the importance of the microphone position for speech recognition in the car. 429-432 - Tasos Anastasakos, Francis Kubala, John Makhoul, Richard M. Schwartz:
Adaptation to new microphones using tied-mixture normalization. 433-436 - William C. Treurniet, Yifan Gong:
Noise independent speech recognition for a variety of noise types. 437-440 - Philip Lockwood, Patrice Alexandre:
Root adaptive homomorphic deconvolution schemes for speech recognition in noise. 441-444 - Mazin G. Rahim, Biing-Hwang Juang:
Signal bias removal for robust telephone based speech recognition in adverse environments. 445-448 - Yoichi Takebayashi, Hiroshi Kanazawa:
Adaptive noise immunity learning for word spotting. 449-452
Speech Analysis
- Gernot Kubin, W. Bastiaan Kleijn:
Time-scale modification of speech based on a nonlinear oscillator model. 453-456 - Hong Sub Choi, Seung Chan Bang, Souguil Ann:
A robust sequential parameter estimation for time-varying speech signal analysis. 457-460 - Naoto Iwahashi, Yoshinori Sagisaka:
Speech spectrum transformation by speaker interpolation. 461-464 - Suhuai Luo, Robin W. King:
A novel approach for classifying continuous speech into visible mouth-shape related classes. 465-468 - Hideyuki Mizuno, Masanobu Abe:
Voice conversion based on piecewise linear conversion rules of formant frequency and spectrum tilt. 469-472 - C. S. Ramalingam, Ramdas Kumaresan:
Voiced-speech analysis based on the residual interfering signal canceler (RISC) algorithm. 473-476 - Hani Yehia, Fumitada Itakura:
Determination of human vocal-tract dynamic geometry from formant trajectories using spatial and temporal Fourier analysis. 477-480 - Jan S. Erkelens, Piet M. T. Broersen:
Analysis of spectral interpolation with weighting dependent on frame energy. 481-484 - Alain Biem, Shigeru Katagiri:
Filter bank design based on discriminative feature extraction. 485-488
Spectrum Quantization for Speech Coding
- Minoru Kohata, Tasuku Takagi:
Vector quantization with hyper-columnar clusters. 489-492 - Kwok-Wah Law, Cheung-Fat Chan:
A novel split residual vector quantization scheme for low bit rate speech coding. 493-496 - Harald Skinnemoen, Andrew Perkis:
Efficient vector quantisation of LPC parameters for noisy channels. 497-500 - Stefan Bruhn:
Efficient interblock noiseless coding of speech LPC parameters. 501-504 - Philip A. Chou, Tom D. Lookabaugh:
Variable dimension vector quantization of linear predictive coefficients of speech. 505-508 - Roar Hagen:
Spectral quantization of cepstral coefficients. 509-512 - Jianping Pan, Thomas R. Fischer:
Vector quantization-lattice vector quantization of speech LPC coefficients. 513-516 - Torbjørn Svendsen:
Segmental quantization of speech spectral information. 517-520 - Wai-Yip Chan, David Chemla:
Low-complexity encoding of speech LSF parameters using constrained-storage TSVQ. 521-524 - Per Hedelin:
Single stage spectral quantization at 20 bits. 525-528 - Terrence G. Champion, Robert J. McAulay, Thomas F. Quatieri:
High-order allpole modelling of the spectral envelope. 529-532
Large Vocabulary Speech Recognition I
- Lalit R. Bahl, Peter V. de Souza, Ponani S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:
Robust methods for using context-dependent features and models in a continuous speech recognizer. 533-536 - Vassilios Digalakis, Hy Murveit:
Genones: optimizing the degree of mixture tying in a large vocabulary hidden Markov model based speech recognizer. 537-540 - Roger K. Moore, Martin J. Russell, Peter Nowell, Simon Downey, Sue Browning:
A comparison of phoneme decision tree (PDT) and context adaptive phone (CAP) based approaches to vocabulary-independent speech recognition. 541-544 - Hsiao-Wuen Hon, Baosheng Yuan, Yen-Lu Chow, Shankar Narayan, Kai-Fu Lee:
Towards large vocabulary Mandarin Chinese speech recognition. 545-548 - Mei-Yuh Hwang, Ronald Rosenfeld, Eric H. Thayer, Mosur Ravishankar, Lin Lawrence Chase, Robert Weide, Xuedong Huang, Fil Alleva:
Improving speech recognition performance via phone-dependent VQ codebooks and adaptive language models in SPHINX-II. 549-552 - Patrick Kenny, Paul Labute, Zhishun Li, Douglas D. O'Shaughnessy:
New graph search techniques for speech recognition. 553-556 - Jean-Luc Gauvain, Lori Faith Lamel, Gilles Adda, Martine Adda-Decker:
The LIMSI continuous speech dictation system: evaluation on the ARPA Wall Street Journal task. 557-560 - Francis Kubala, Anastasios Anastasakos, John Makhoul, Long Nguyen, Richard M. Schwartz, George Zavaliagkos:
Comparative experiments on large vocabulary speech recognition. 561-564
Speech Synthesis
- Thierry Dutoit:
High quality text-to-speech synthesis: a comparison of four candidate algorithms. 565-568 - Hisashi Kawai, Norio Higuchi, Tohru Shimizu, Seiichi Yamamoto:
Development of a text-to-speech system for Japanese based on waveform splicing. 569-572 - Francisco M. Gimenez de los Galanes, Mohammad Hasan Savoji, José Manuel Pardo:
New algorithm for spectral smoothing and envelope modification for LP-PSOLA synthesis. 573-576 - Kenzo Itoh, Shin'ya Nakajima, Tomohisa Hirokawa:
A new waveform speech synthesis approach based on the COC speech spectrum. 577-584 - Vesa Välimäki, Matti Karjalainen, Timo Kuisma:
Articulatory speech synthesis based on fractional delay waveguide filters. 585-588 - Eduardo López Gonzalo, Luis A. Hernández Gómez:
Data-driven joint f0 and duration modeling in text to speech conversion for Spanish. 589-592 - Yoichi Yamashita, Riichiro Mizoguchi:
Automatic generation of prosodic rules for speech synthesis. 593-596
Isolated Word Recognition
- Hideki Noda, Mehdi N. Shirazi:
A MRF-based parallel processing algorithm for speech recognition using linear predictive HMM. 597-600 - Ben P. Milner, Saeed Vaseghi:
Speech modelling using cepstral-time feature matrices and hidden Markov models. 601-604 - Weon-Goo Kim, Jeung-Yoon Choi, Dae Hee Youn:
HMM with global path constraint in Viterbi decoding for isolated word recognition. 605-608 - Philippe Le Cerf, Kris Demuynck, Jacques Duchateau, Dirk Van Compernolle:
Pseudo-segment based speech recognition using neural recurrent whole-word recognizers. 609-612 - Yaxin Zhang, Michael D. Alder, Roberto Togneri:
Using Gaussian mixture modeling in speech recognition. 613-616 - Jean-Marc Boite, Hervé Bourlard, Bart D'hoore, Sari Accaino, Johan Vantieghem:
Task independent and dependent training: performance comparison of HMM and hybrid HMM/MLP approaches. 617-620 - Stephan Euler, Joachim Zinke:
The influence of speech coding algorithms on automatic speech recognition. 621-624 - Jung-Kuei Chen, Frank K. Soong:
Discriminative training of high performance speech recognizer using N best candidates. 625-628 - Pietro Laface, Luciano Fissore:
Model topology selection for isolated word recognition. 629-632 - Joseph Di Martino, Jean-François Mari, Bruno Mathieu, Karine Perot, Kamel Smaïli:
Which model for future speech recognition systems: hidden Markov models or finite-state automata? 633-636 - Eiichi Tsuboka, Jun'ichi Nakahashi:
On the fuzzy vector quantization based hidden Markov model. 637-640
Volume 2
Speech Understanding and Language Modelling II
- Helen M. Meng, Stephanie Seneff, Victor W. Zue:
Phonological parsing for reversible letter-to-sound/sound-to-letter generation. 1-4 - Ryosuke Isotani, Shoichi Matsunaga:
A stochastic language model for speech recognition integrating local and global constraints. 5-8 - Ana L. N. Fred, José M. N. Leitão:
Improving sentence recognition in stochastic context-free grammars. 9-12 - Toshihisa Tashiro, Toshiyuki Takezawa, Tsuyoshi Morimoto, Masaaki Nagata:
Efficient chart parsing of speech recognition candidates. 13-16 - Wayne H. Ward, Sunil Issar:
Integrating semantic constraints into the Sphinx-II recognition search. 17-20 - Sheryl R. Young:
Detecting misrecognitions and out-of-vocabulary words. 21-24 - Tatsuya Kawahara, Masahiro Araki, Shuji Doshita:
Heuristic search integrating syntactic, semantic and dialog-level constraints. 25-28 - Tung-Hui Chiang, Yi-Chung Lin, Keh-Yih Su:
On jointly learning the parameters in a character-synchronous integrated speech and language model. 29-32 - Hiroshima Matsu'ura, Yasuyuki Masai, Jun'ichi Iwasaki, Shin'ichi Tanaka, Hiroyuki Kamio, Tsuneo Nitta:
A multimodal, keyword-based spoken dialogue system-MultiksDial. 33-36 - Roland Kuhn, Renato De Mori, Evelyne Millien:
Learning consistent semantics from training data. 37-40 - Ludwig A. Schmid:
Parsing word graphs using a linguistic grammar and a statistical language model. 41-44
Speech Enhancement and Noise Reduction
- Levent M. Arslan, John H. L. Hansen:
Minimum cost based phoneme class detection for improved iterative speech enhancement. 45-48 - John P. Openshaw, John S. Mason:
On the limitations of cepstral features in noise. 49-52 - Fei Xie, Dirk Van Compernolle:
A family of MLP based nonlinear spectral estimators for noise reduction. 53-56 - Tetsunori Kobayashi, Ryuji Mine, Katsuhiko Shirai:
Markov model based noise modeling and its application to noisy speech recognition using dynamical features of speech. 57-60 - Fu-Hua Liu, Richard M. Stern, Alejandro Acero, Pedro J. Moreno:
Environment normalization for robust speech recognition using direct cepstral comparison. 61-64 - Saeed Vaseghi, Ben P. Milner, Jason J. Humphries:
Noisy speech recognition using cepstral-time features and spectral-time filters. 65-68 - Javier Hernando, Climent Nadeu:
Speech recognition in noisy car environment based on OSALPC representation and robust similarity measuring techniques. 69-72 - Michael E. Knappe, Rafik A. Goubran:
Steady-state performance limitations of full-band acoustic echo cancellers. 73-76 - Malcolm Slaney, Daniel Naar, Richard F. Lyon:
Auditory model inversion for sound separation. 77-80 - Tsuyoshi Usagawa, Makoto Iwata, Masanao Ebata:
Speech parameter extraction in noisy environment using a masking model. 81-84 - Frank Seide, Alfred Mertins:
Non-linear regression based feature extraction for connected-word recognition in noise. 85-88 - Yuqing Gao, Jean-Paul Haton:
A hierarchical LPNN network for noise reduction and noise degraded speech recognition. 89-92
CELP Coding II
- Akitoshi Kataoka, Takehiro Moriya, Shinji Hayashi:
Implementation and performance of an 8-kbit/s conjugate structure CELP speech coder. 93-96 - Redwan Salami, Claude Laflamme, Jean-Pierre Adoul:
8 kbit/s ACELP coding of speech with 10 ms speech-frame: a candidate for CCITT standardization. 97-100 - James M. Ooi, Vishu Viswanathan:
A computationally efficient wavelet transform CELP coder. 101-104 - Dipanjan Sen, W. Harvey Holmes:
Perceptual enhancement of CELP speech coders. 105-108 - Christian G. Gerlach:
CELP speech coding with almost no codebook search. 109-112 - Satoshi Miki, Kazunori Mano, Takehiro Moriya, Kumiko Oguchi, Hitoshi Ohmuro:
A pitch synchronous innovation CELP (PSI-CELP) coder for 2-4 kbit/s. 113-116 - Paul Mermelstein, Ping Zheng, M. Saikaly:
Multi-band residual coding of CELP codecs at 8 kb/s. 117-120 - Erik Harborg, Jan E. Knudsen, Arild Fuldseth, Finn Tore Johansen:
A real-time wideband CELP coder for a videophone application. 121-124
Large Vocabulary Speech Recognition II
- Philip C. Woodland, J. J. Odell, V. Valtchev, Steve J. Young:
Large vocabulary continuous speech recognition using HTK. 125-128 - Xavier L. Aubert, Christian Dugast, Hermann Ney, Volker Steinbiss:
Large vocabulary continuous speech recognition of Wall Street Journal data. 129-132 - Jia-Lin Shen, Hsin-Min Wang, Bo-Ren Bai, Lin-Shan Lee:
An initial study on a segmental probability model approach to large-vocabulary continuous Mandarin speech recognition. 133-136 - Jung-Kuei Chen, Frank K. Soong, Lin-Shan Lee:
Large vocabulary word recognition based on tree-trellis search. 137-140 - Yasuhiro Minami, Kiyohiro Shikano, Satoshi Takahashi, Tomokazu Yamada:
Search algorithm that merges candidates in meaning level for very large vocabulary spontaneous speech recognition. 141-144 - Michael K. Brown, Stephen C. Glinski:
Context-free large-vocabulary connected speech recognition with evolutional grammars. 145-148 - Harald Singer, Jun-ichi Takami, Shoichi Matsunaga:
Non-uniform unit parsing for SSS-LR continuous speech recognition. 149-152 - Wu Chou, Tatsuo Matsuoka, Biing-Hwang Juang, Chin-Hui Lee:
An algorithm of high resolution and efficient multiple string hypothesization for continuous speech recognition using inter-word models. 153-156 - Harinath Garudadri, Paul Labute, Gilles Boulianne, Patrick Kenny:
Fast match acoustic models in large vocabulary continuous speech recognition. 157-160 - Ponani S. Gopalakrishnan, David Nahamoo, Mukund Padmanabhan, Michael A. Picheny:
A channel-bank-based phone detection strategy. 161-164 - Laurence Devillers, Christian Dugast:
Hybrid system combining expert-TDNNs and HMMs for continuous speech recognition. 165-168
Prosody/Analysis
- Andrew Hunt:
A generalised model for utilising prosodic information in continuous speech recognition. 169-172 - Ralf Kompe, Anton Batliner, Andreas Kießling, Ute Kilian, Heinrich Niemann, Elmar Nöth, Peter Regel-Brietzmann:
Automatic classification of prosodically marked phrase boundaries in German. 173-176 - Pierre Dumouchel:
Suprasegmental features and continuous speech recognition. 177-180 - Adrian Grigoriu, Julie Vonwiller, Robin W. King:
An automatic intonation tone contour labelling and classification algorithm. 181-184 - Hiroshi Shimodaira, Mitsuru Nakai:
Prosodic phrase segmentation by pitch pattern clustering. 185-188 - Hiroshi Ohmura:
Fine pitch contour extraction by voice fundamental wave filtering method. 189-192 - Shoji Kajita, Fumitada Itakura:
Subband-Autocorrelation analysis and its application for speech recognition. 193-196 - Michiel Bacchiani, Kiyoaki Aikawa:
Optimization of time-frequency masking filters using the minimum classification error criterion. 197-200 - Yves Laprie, Marie-Odile Berger:
A new paradigm for reliable automatic formant tracking. 201-204 - Mahesan Niranjan, Ingemar J. Cox, Sunita L. Hingorani:
Recursive tracking of formants in speech signals. 205-208
Audio and Electroacoustics I
- Peter A. Monta, Shiufun Cheung:
Low rate audio coder with hierarchical filterbanks and lattice vector quantization. 209-212 - Christopher M. Hicks, Simon J. Godsill:
A two-channel approach to the removal of impulsive noise from archived recordings. 213-216 - Erkan Dorken, S. Hamid Nawab:
Improved musical pitch tracking using principal decomposition analysis. 217-220 - Francisco Javier Casajús-Quirós, Pablo Fernandez-Cid:
Real-time, loose-harmonic matching fundamental frequency estimation for musical signals. 221-224 - Akira Nakamura, Nobumasa Seiyama, Ryou Ikezawa, Tohru Takagi, Eiichi Miyasaka:
Real time speech rate converting system for elderly people. 225-228 - Hector R. Javkin, Elizabeth Keate, Norma Antonanzas-Barroso, Yoshinori Yamada, Karen Youdelman:
Automatic model parameter generation for the speech training of deaf children. 229-232 - Simon J. Godsill:
Recursive restoration of pitch variation defects in musical recordings. 233-236 - John D. Hoyt, Harry Wechsler:
Detection of human speech in structured noise. 237-240 - Scott D. Snyder:
Active control using IIR filters-a second look. 241-244 - Akihiro Hirano, Akihiko Sugiyama, Yasuhiro Arasawa, Noboru Kawayachi:
DSP implementation and performance evaluation of a compact stereo echo canceller. 245-248
Audio and Electroacoustics II
- Héctor M. Pérez Meana, Luís Nino de Rivera, Mariko Nakano-Miyatake, Fausto Casco-Sanchez, Juan Carlos Sánchez-García:
A time varying step size normalized LMS echo canceler algorithm. 249-252 - Larry J. Eriksson, Mark C. Allie, Douglas E. Melton, Steven R. Popovich, Trevor A. Laak:
Fully adaptive generalized recursive control system for active acoustic attenuation. 253-256 - Paul C. Meuse, Harvey F. Silverman:
Characterization of talker radiation pattern using a microphone array. 257-260 - Stéphane Bédard, Benoît Champagne, Alex Stephenne:
Effects of room reverberation on time-delay estimation performance. 261-264 - Marcio G. Siqueira, Abeer Alwan:
New adaptive-filtering techniques applied to speech echo cancellation. 265-268 - Mark A. Poletti:
Colouration in assisted reverberation systems. 269-272 - Maurizio Omologo, Piergiorgio Svaizer:
Acoustic event localization using a crosspower-spectrum phase based technique. 273-276
Ocean Signal Processing - Radar, Geophysical and Biomedical
- Abderrahman Essebbar:
Parametric separation applied to underwater acoustic multipath propagation. 277-280 - Brigitte Colnet, Jean-Paul Haton:
Far field array processing with neural networks. 281-284 - David J. Kershaw, Robin J. Evans:
Adaptive waveform selection for active sonar. 285-288 - Carlo S. Regazzoni, Alessandra Tesei, Giorgio Tacconi:
A comparison between spectral and bispectral analysis for ship detection from acoustical time series. 289-292 - Jean-Claude Di Martino, Brigitte Colnet, Marc Di Martino:
The use of non-supervised neural networks to detect lines in lofargram. 293-296 - Alex B. Gershman, Vitaly A. Zverev:
Adaptive detection of moving signal using shallow sea hydroacoustic data. 297-300 - Yann Stéphan, François Regis Martin-Lauzer:
GASTOM 90 tomography experiment data inversion. 301-304 - François Regis Martin-Lauzer, Didier Mauuary, Yann Stéphan:
Probabilistic ray identification: a new tool for ocean acoustic tomography. 305-308 - Steven Ivandich:
Quantisation error modelling of narrowband adaptive arrays using projected perturbation sequences. 309-312 - R. Rajagopal, K. Anoop Kumar, P. Ramakrishna Rao:
An integrated approach to passive target classification. 313-316 - Olivier Trémois, Jean-Pierre Le Cadre:
Maneuvering target motion analysis using hidden Markov model. 317-320
Detection, Classification and Localisation I
- Sanjay K. Mehta, Edward L. Titlebaum:
Optimum Costas-like decompositions of Costas arrays for channel characterization and communications. 321-324 - Amlan Kundu, George C. Chen, Charles E. Persons:
Transient sonar signal classification using hidden Markov model and neural net. 325-328 - Justin J. Smith, Yee Hong Leung, Antonio Cantoni:
Statistics of the phase delays between array receivers estimated from the eigendecomposition of the signal correlation matrix. 329-332 - Michel Bouvet, Laurent Kerleguer, Philippe Mennecier, Michel Cresp:
Extraction of pertinent parameters in oceanographic data. 333-336 - Christoph F. Mecklenbräuker, Johann F. Böhme:
Matched field processing in shallow ocean: signal arrival identification using EM algorithm. 337-340 - Didier Mauuary, Geneviève Jourdain:
Bayesian time delay estimation for ocean acoustic tomography. 341-344 - Donald W. Tufts, Hongya Ge, Ramdas Kumaresan:
Resolving ambiguities in estimating spatial frequencies in sparse linear array. 345-348 - Herve Chuberre, Jean-Jacques Fuchs:
A deconvolution approach to moving sources localization. 349-352
Detection, Classification and Localisation II
- Dimitris Pantzartzis, Dimitri Alexandrou, Vincent Premus:
High-resolution bathymetric simulations based on Kirchhoff scattering theory and anisotropic seafloor modeling. 353-356 - George Henry Niezgoda, K. C. Ho:
Geolocalization by combined range difference and range rate difference measurements. 357-360 - Partha Pratim Kanjilal, Sarbani Palit:
Extraction of multiple periodic waveforms from noisy data. 361-364 - Tayfun Akgül, Amro El-Jaroudi, Marwan A. Simaan:
Deconvolution of sensor array signals using time-scaling. 365-368 - David A. Noon, Dennis Longstaff, Glen F. Stickley:
Correction of I/Q errors in homodyne step frequency radar refocuses range profiles. 369-372 - Anthony Zyweck, Robert E. Bogner:
Radar target recognition using range profiles. 373-376 - Naoki Ehara, Iwao Sasase, Shinsaku Mori:
Weak radar signal detection based on wavelet transform. 377-380 - Zouak Mouhcine, Joseph Saillard:
Detection, identification and tracking of active scatterers. 381-384 - Philippe Delachartre, Didier Vray, Zhigang Sun, Gérard Gimenez, Albin Dziedzic:
Estimation of the causal impulse response of underwater target. 385-388 - Zoran Zvonar, David Brady, Josko Catipovic:
An adaptive linear multiuser receiver for deep water acoustic local area networks. 389-392 - Yiu Tong Chan, K. C. Ho:
An efficient closed-form localization solution from time difference of arrival measurements. 393-396 - Yisong Ye, Jitendra K. Tugnait:
Time delay estimation using integrated polyspectrum. 397-400
VLSI Architectures for Video and Speech Processing
- Mohan Vishwanath, Chaitali Chakrabarti:
A VLSI architecture for real-time hierarchical encoding/decoding of video using the wavelet transform. 401-404 - Chetana Nagendra, Mary Jane Irwin, Robert Michael Owens:
Digit pipelined discrete wavelet transform. 405-408 - Benjamin M. Gordon, Teresa H.-Y. Meng:
A low power subband video decoder architecture. 409-412 - Winfried Gehrke, Richard Hoffer, Peter Pirsch:
A hierarchical multiprocessor architecture based on heterogeneous processors for video coding applications. 413-416 - Toshiyuki Araki, Masaki Toyokura, Toshihide Akiyama, Hiroshi Takeno, Brent Wilson, Kunitoshi Aono:
Video DSP architecture for MPEG2 codec. 417-420 - Junji Suzuki, Florent Colin, Sadayasu Ono:
Arithmetic codec from behavioral description based LSI-CAD for fully programmable image coding system. 421-424 - Luís Díez del Río, Sofía Moreno-Pérez, Rafael Sarmiento de Sotomayor, José Parera, Marcelino Veiga-Pérez, Ramón García-Gómez:
Secure speech and data communication over the public switching telephone network. 425-428 - Jari Nurmi, Ville Eerola, Erwin Ofner, Andreas Gierlinger, Jürgen Jernej, Teppo Karema, Tommi Raita-aho:
A DSP core for speech coding applications. 429-432
CAD for VLSI Signal Processing
- Lisa M. Guerra, Miodrag Potkonjak, Jan M. Rabaey:
Concurrency characteristics in DSP programs. 433-436 - Mazen A. R. Saghir, Paul Chow, Corinna G. Lee:
Application-driven design of DSP architectures and compilers. 437-440 - Sujit Dey, Miodrag Potkonjak, Rabindra K. Roy:
Behavioral synthesis of low-cost partial scan designs for DSP applications. 441-444 - José Luis Pino, Thomas M. Parks, Edward A. Lee:
Automatic code generation for heterogeneous multiprocessors. 445-448 - Matthias Pankert, Oliver Mauss, Sebastian Ritz, Heinrich Meyr:
Dynamic data flow and control flow in high level DSP code synthesis. 449-452 - Praveen K. Murthy, Shuvra S. Bhattacharyya, Edward A. Lee:
Minimizing memory requirements for chain-structured synchronous dataflow programs. 453-456 - Wonyong Sung, Ki-Il Kum:
Word-length determination and scaling software for a signal flow block diagram. 457-460 - Minjoong Rim, Rajiv Jain:
RECALS II: a new list scheduling algorithm. 461-464 - Vojin Zivojnovic, Sebastian Ritz, Heinrich Meyr:
Retiming of DSP programs for optimum vectorization. 465-468 - H. John Reekie, John M. Potter:
Generating efficient loop code for programmable DSPs. 469-472 - Ingrid Verbauwhede, Chris J. Scheers, Jan M. Rabaey:
Specification and support for multidimensional DSP in the SILAGE language. 473-476
Algorithms, Architectures, and Circuits for VLSI Signal Processing
- Marc Moonen, Ian K. Proudler, John G. McWhirter, Gerben Hekstra:
On the formal derivation of a systolic array for recursive least squares estimation. 477-480 - Knut Hüper, Steffen Paul, Rainer Pauli:
Computation of the real Schur decomposition of nonsymmetric matrices and its hardware implementation. 481-484 - Amitabha Das:
A novel efficient parallel algorithm for RNS to binary conversion for arbitrary moduli set. 485-488 - Konstantinos Konstantinides, Balas K. Natarajan:
Algorithm and architecture: for non-linear noise filtering via piecewise linear compression. 489-492 - Ed F. Deprettere, Hylke W. van Dijk, Gerben J. Hekstra:
A 'Jacobi' signal processing unit for time-adaptive SVD. 493-496 - Miodrag Potkonjak, Mani B. Srivastava:
Design of high throughput, low latency and low cost structures for linear systems. 497-500 - Jiun-In Guo, Chi-Min Liu, Chein-Wei Jen:
A novel VLSI array design for the discrete Hartley transform using cyclic convolution. 501-504 - Phillip M. S. Burt:
Using preliminary decisions to reduce complexity in adaptive equalizers. 505-508 - Kamal Nourji, Nicolas Demassieux:
Optimal VLSI architecture for distributed arithmetic-based algorithms. 509-512 - Neil W. Bergmann, J. Craig Mudge:
An analysis of FPGA-based custom computers for DSP applications. 513-516 - Sal Bernadas, Charles Thompson:
A multi-channel decimator for a tri-level delta-sigma modulator. 517-520 - Vijay K. Jain, Lei Lin:
Square-root, reciprocal, sine/cosine, arctangent cell for signal and image processing. 521-524
Neural Network: Theory and Implementation
- A. David M. Garvin:
A self-structuring algorithm for artificial neural networks. 525-528 - Lizhong Wu, Mahesan Niranjan:
On the design of nonlinear speech predictors with recurrent nets. 529-532 - José Carlos Príncipe, Jyh-Ming Kuo:
Noise reduction in state space using the focused gamma neural network. 533-536 - Pascale Hirschauer, Pascal Larzabal, Henri Clergeot:
Design of neural estimators for multisensors: second order backpropagation, initialization and generalization. 537-540 - Herbert Gish, Man-Hung Siu:
An invariance property of neural networks. 541-544 - Mohamed Ibnkahla, Stéphane Puechmorel, Francis Castanie:
A constrained neural network with complex activation function: application to time-frequency analysis. 545-548 - Mang Zhu, James A. Cadzow:
The conditional expectation via a general class of nonlinear networks. 549-552 - Piero Cosi, Emanuela Magno Caldognetto, Kyriaki Vagges, Gian Antonio Mian, Matteo Contolini:
Bimodal recognition experiments with recurrent neural networks. 553-556
Neural Modelling and Image Processing
- Ce Zhu, Lihua Li, Zhenya He, Jun Wang:
A new competitive learning algorithm for vector quantization. 557-560 - Bin Qiu, Paul Im, Anne Pleasants:
The design of neural network configuration for object recognition. 561-564 - Alex Sherstinsky, Rosalind W. Picard:
M-lattice: a novel non-linear dynamical system and its application to halftoning. 565-568 - Stefanos D. Kollias, Dimitrios Kalogeras:
A multiresolution probabilistic neural network for image segmentation. 569-572 - Tom Brotherton, Tom Pollard, Pat Simpson, Anthony DeMaria:
Echocardiogram structure and tissue classification using hierarchical fuzzy neural networks. 573-576 - Andre Yakovleff, Mario Cavaiuolo:
A simulation environment for very large neural networks. 577-580 - Clifford Sze-Tsan Choy, Wan-Chi Siu:
Approach of using a density equalizing function to self-organizing learning for solving travelling salesman problem. 581-584 - Shih-Shien You, Jenq-Neng Hwang, I-Chang Jou, Shyh-Rong Lay:
A new cascaded projection pursuit network for nonlinear regression. 585-588 - Patrizia Bollini, Guido Castellini, Enrico Del Re, Giacomo Mangani, Laura Pierucci:
Neural receiver for CPM signals. 589-592 - William B. Ribbens, Jaehong Park, DaeEun Kim:
Application of neural networks to detecting misfire in automotive engines. 593-596
Neural Network Pattern Recognition
- Andrew Luk, Wai-Fung Leung:
Optimal, matching-score network for pattern classification. 597-600 - Jongwan Kim, Jesung Ahn, Chong-Sang Kim, Heeyeung Hwang, Seongwon Cho:
Multiple neural networks using the reduced input dimension. 601-604 - Eveline J. Bellegarda, Jerome R. Bellegarda, Jin H. Kim:
On-line handwritten character recognition using parallel neural networks. 605-608 - V. Chandrasekaran, Marimuthu Palaniswami, Terry M. Caelli:
Invariant property of spatio-temporal feature maps using gated neuronal architecture. 609-612 - Ahmed M. Darwish, Gasser Auda:
A new composite feature vector for Arabic handwritten signature recognition. 613-616 - Luan Ling Lee:
On two-pattern classification and feature selection using neural networks. 617-620 - Victor E. DeBrunner, Tod Bussert:
Using artificial neural networks to improve the mechanical signature analysis test. 621-624 - Hong Yan:
Transformation of optimized prototypes for handwritten digit recognition. 625-628 - Minh Tue Vo:
Incremental learning using the time delay neural network. 629-632 - Stefan Manke, Ulrich Bodenhausen:
A connectionist recognizer for on-line cursive handwriting recognition. 633-636 - Markus Schenkel, Isabelle Guyon, Don Henderson:
On-line cursive script recognition using time delay neural networks and hidden Markov models. 637-640 - Nanda Nandagopal, N. M. Martin, R. P. Johnson, Peter Lozo, Marimuthu Palaniswami:
Performance of radar target recognition schemes using neural networks-a comparative study. 641-644
Applications of Neural Networks to Speech Processing
- Gerhard Rigoll:
Mutual information neural networks: a new connectionist approach for dynamic speech recognition tasks. 645-648 - Vahidi Tabarabaee, Babak Azimi-Sadjadi, Seyed Bahram Zahir Azami, Caro Lucas:
Isolated word recognition using a hybrid neural network. 649-652 - Sin-Horng Chen, Yuan-Fu Liao, Wen-Yuan Chen:
Application of a generalized probabilistic descent method to recurrent neural network based speech recognition. 653-656 - Helge B. D. Sørensen, Uwe Hartmann:
Hybrid model decomposition of speech and noise in a radial basis function neural model framework. 657-660 - Abdesselam Bouzerdoum, Michael L. Southcott, Jihan Zhu, Robert E. Bogner:
A neural architecture for hierarchical clustering. 661-664 - Khaled Hassanein, Li Deng, Mohamed I. Elmasry:
Vowel classification using a neural predictive HMM: a discriminative training approach. 665-668 - Christoph Bregler, Yochai Konig:
"Eigenlips" for robust speech recognition. 669-672 - Victor Abrash, Michael Cohen, Horacio Franco, Isao Arima:
Incorporating linguistic features in a hybrid HMM/MLP speech recognizer. 673-676 - Wolfgang Reichl, Peter Caspary, Günther Ruske:
A new model-discriminant training algorithm for hybrid NN-HMM systems. 677-680 - Jun He, Henri Leich:
Combining stochastic trajectory model and discriminative feature in speech recognizer. 681-684 - Ravi Sankar, Shrenik Patravali:
Noise immunization using neural net for speech recognition. 685-688
Volume 3
Linear Time-Frequency Representations: Gabor and Wavelet Transforms
- Richard S. Orr:
A Gabor sampling theorem and some time-bandwidth implications. 1-4 - Nicholas J. Redding, Garry N. Newsam:
Computation and conditioning of the finite-discrete Gabor transform. 5-8 - Levent Onural, Mefharet Kocatepe, Haldun M. Özaktas:
A class of wavelet kernels associated with wave propagation. 9-12 - Feng Bao, Nurgun Erdol:
The optimal wavelet transform and translation invariance. 13-16 - Peter Rieder, Jürgen Götze, Josef A. Nossek:
Algebraic design of discrete multiwavelet transforms. 17-20 - Cheng-Youn Lu:
Second-order statistics of the wavelet transform of multiplicative white stochastic process. 21-24 - Thomas Frederick, Nurgun Erdol:
Arbitrary tilings of phase space. 25-28 - Shie Qian, Dapang Chen:
Time-frequency distribution series. 29-32
Adaptive Methods and Analysis
- Seung Chan Bang, Hong Sub Choi, Souguil Ann:
Performance analysis of the dual sign algorithm with contaminated-Gaussian noise. 33-36 - Michael R. Frater, C. Richard Johnson Jr.:
Local minima escape transients of CMA. 37-40 - Shigeji Ikeda, Akihiko Sugiyama:
A fast convergence algorithm for sparse-tap adaptive FIR filters for an unknown number of multiple echoes. 41-44 - Michael O. Kolawole:
Stabilization of tracks with multiple model filters. 45-48 - Eric Moreau, Odile Macchi:
A one stage self-adaptive algorithm for source separation. 49-52 - Yoshihiro Ono, Hitoshi Kiya:
Performance analysis of subband adaptive systems using an equivalent model. 53-56 - Stefaan Van Gerven, Dirk Van Compernolle:
On the use of decorrelation in scalar signal separation. 57-60 - Kutluyil Dogancay, Rodney A. Kennedy:
A globally admissible off-line modulus restoral algorithm for low-order adaptive channel equalisers. 61-64 - José Manuel Páez-Borrallo, Iván A. Pérez-Álvarez, Santiago Zazo-Bello:
Adaptive filtering in data communications with self improved error reference. 65-68 - Miwa Sakai, Kiyoharu Aizawa, Mitsutoshi Hatori:
Fractionally spaced equalizers with adaptive sampling. 69-72
Non-Linear and Adaptive Filters
- S. Douglas Peters, Andreas Antoniou:
Parallel adaptation for enhanced RLS tracking. 73-76 - Constantin Papaodysseus, Cristos C. Halkias, Costas N. Triantafyllou, V. Asimakopoulos:
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 schemes. 77-80 - Kalavai J. Raghunath, Keshab K. Parhi:
Fixed and floating point error analysis of QRD-RLS and STAR-RLS adaptive filters. 81-84 - Shigenori Kinjo:
New interference robust block adaptive filter with correlated signal input. 85-88 - Jian Zhan, Fu Li:
A new algorithm of realizing arbitrary nonlinear filters-adaptive neural filters. 89-92 - Constantine Kotropoulos, Ioannis Pitas:
Multichannel L-filter design based on marginal data ordering. 93-96 - Ki Dong Lee, Yong Hoon Lee:
Minimum mean square error filtering over the class of extended threshold Boolean filters. 97-100 - Jinhui Chao, Teruyuki Sato, Shigeo Tsujii:
A stable and globally convergent OEM IIR ADF. 101-104 - Sin Chun Ng, Chi Yin Chung, Shu Hung Leung, Andrew Luk:
Fast convergent genetic search for adaptive IIR filtering. 105-108
Multirate Processing and Wavelets
- Ramesh A. Gopinath:
Factorization approach to time-varying filter banks and wavelets. 109-112 - Alfred Mertins:
Statistical optimization of PR-QMF banks and wavelets. 113-116 - Zoran Cvetkovic, Martin Vetterli:
Wavelet extrema and zero-crossings representations: properties and consistent reconstruction. 117-120 - Ram G. Shenoy:
Analysis of multirate components and application to multirate filter design. 121-124 - Joël Mau:
Regular M-band modulated orthogonal transforms. 125-128 - Tsuhan Chen, P. P. Vaidyanathan:
Phase linearization of filters in analysis/synthesis filter banks. 129-132 - François Déprez, Olivier Rioul, Pierre Duhamel:
Border recovery for subband processing of finite-length signals. Application to time-varying filter banks. 133-136 - Masaaki Ikehara:
Cosine-modulated 2 dimensional FIR filter banks satisfying perfect reconstruction. 137-140 - Shing-Chow Chan:
Two dimensional nonseparable modulated filter banks. 141-144 - Kyusik Park, Richard A. Haddad:
Modeling and optimal compensation of quantization in multidimensional M-band filter bank. 145-148
Subband Processing and Filter Banks
- Norbert J. Fliege:
Modified DFT polyphase SBC filter banks with almost perfect reconstruction. 149-152 - Bruce W. Suter, Mark E. Oxley:
A generalized lapped orthonormal transform for asymmetrically overlapped windows. 153-156 - Iraj Sodagar, Kambiz Nayebi, Thomas P. Barnwell III, Mark J. T. Smith:
A novel structure for time-varying FIR filter banks. 157-160 - Cormac Herley, Nguyen T. Thao:
Implementable orthogonal signal projections based on multirate filters. 161-164 - Andrew Reilly, Gordon Frazer:
An efficient algorithm for analytic signal generation for time-frequency distributions. 165-168 - Lu Lin, Tyseer Aboulnasr:
Adaptive signal processing in subbands using sigma-delta modulation technique. 169-172 - Ute Petersohn, Norbert J. Fliege, Horst Unger:
Exact analysis of aliasing effects and non-stationary quantization noise in multirate systems. 173-176 - P. P. Vaidyanathan:
Causal FIR matrices with anticausal FIR inverses, and application in characterization of biorthonormal filter banks. 177-180 - Imran A. Shah, Ton Kalker:
On ladder structures and linear phase conditions for bi-orthogonal filter banks. 181-184 - Greg Smart, Alan B. Bradley:
Filter bank design based on time domain aliasing cancellation with non-identical windows. 185-188 - Hervé Le Bihan, Pierre Siohan:
Identification techniques for the design of cascade forms perfect-reconstruction two-channel filter banks. 189-192
DSP Applications
- Chi-Min Liu, Kuo-Guan Wu, Jer-Heh Sheu:
A single Kalman filtering algorithm for maneuvering target tracking. 193-196 - Jane E. Perkins, Ian Coat:
Pulse train deinterleaving via the Hough transform. 197-200 - Letizia Lo Presti, Giuseppe Cardamone:
A direct digital frequency synthesizer using an IIR filter implemented with a DSP microprocessor. 201-204 - Russell D. Priebe, Gary R. Wilson:
Wavelet applications to structural analysis. 205-208 - Soo-Chang Pei, Chien-Cheng Tseng:
A technique for transient suppression of IIR notch filter. 209-212 - Hsiang-Feng Chi, Ja-Ling Wu:
Echo cancellation with reference signal generator and reliable receiving schemes for intersymbol interference. 213-216 - Takashi Kohama, Shogo Nakamura, Hiroshi Hoshino:
A noise cancelling filter for the digital Holter monitoring system. 217-220 - Mohammed A. Khasawneh, Tariq Haddad:
Real-time echo cancellation using a new fast LMS-based algorithm. 221-224 - Jeng-Kuang Hwang, Shun-Tai Wei:
Adaptive DOA tracking by rank-revealing QR updating and exponential sliding windows techniques. 225-228 - B. Madhukar, Kasi Rajgopal, Lalit M. Patnaik:
Parametric technique for P wave delineation. 229-232
DSP Applications in Data Communications
- Uma Gummadavelli, Jitendra K. Tugnait:
Blind channel estimation and equalization with partial-response input signals. 233-236 - Stephen Oh, Domingo Garcia:
Implementation of a parallel DFE using residue number system. 237-240 - Sally L. Wood, John R. Treichler:
Computational and performance analysis of Radon transform based constellation identification. 241-244 - Julien J. Nicolas, Jae S. Lim:
On the performance of multicarrier modulation in a broadcast multipath environment. 245-248 - Pervez M. Aziz, Henrik V. Sorensen, Jan Van der Spiegel:
Multi band sigma delta analog to digital conversion. 249-252 - Albert M. Gottlieb:
A DSP-based research prototype reverse channel transmitter/receiver for ADSL. 253-256 - Gin-Kou Ma, Junghsi Lee, V. John Mathews:
A RLS bilinear filter for channel equalization. 257-260 - Selim U. Zaman, K. Warren Yates:
Use of the DFT for synchronization in packetized data communications. 261-264
Signal Reconstruction
- Luc Vandendorpe, Paul Delogne, Benoît Maison, Laurent Cuvelier:
MMSE design of polyphase components for generalized interpolators. 265-268 - Farook Sattar, Göran Salomonsson:
A method for non-parametric waveform estimation based on filter banks. 269-272 - Alan J. Coulson, Rodney G. Vaughan:
A new sensor array signal processing technique. 273-276 - Li-Chien Lin, C.-C. Jay Kuo:
Robust signal extrapolation using wavelets. 277-280 - Xiang-Gen Xia, C.-C. Jay Kuo, Zhen Zhang:
The generalized Backus-Gilbert inversion method for signal recovery in multiresolution spaces. 281-284 - Andrzej Tarczynski, Wojciech Kozinski, Gerald D. Cain:
Sampling rate conversion using fractional-sample delay. 285-288 - Orhan Arikan, Bülent Baygün:
Least squares signal reconstruction under normalized autocorrelation constraints. 289-292 - Jong Wook Park, Dong Sik Kim, Sang Uk Lee:
On the error concealment technique for DCT based image coding. 293-296
Adaptive Equalization
- Tülay Adali, M. Kemal Sönmez:
Channel equalization with perceptrons: an information-theoretic approach. 297-300 - Christian Bourget, Tyseer Aboulnasr:
Inverse filtering of room impulse response for binaural recording playback through loudspeakers. 301-304 - Iain B. Collings, John B. Moore:
Adaptive HMM filters for signals in noisy fading channels. 305-308 - Woon-Seng Gan, John J. Soraghan, Tariq S. Durrani:
Functional-link models for adaptive channel equaliser. 309-312 - James P. LeBlanc, Kutluyil Dogancay, Rodney A. Kennedy, C. Richard Johnson Jr.:
Effects of input data correlation on the convergence of blind adaptive equalizers. 313-316 - Sylvie Mayrargue:
A blind spatio-temporal equalizer for a radio-mobile channel using the constant modulus algorithm (CMA). 317-320 - Constantinos B. Papadias, Dirk T. M. Slock:
New adaptive blind equalization algorithms for constant modulus constellations. 321-324 - Ken Yamazaki, Rodney A. Kennedy:
On globally convergent blind equalization for QAM systems. 325-328
Time-Frequency Analysis and Applications
- Patrick Flandrin, Richard G. Baraniuk, Olivier J. J. Michel:
Time-frequency complexity and information. 329-332 - John R. O'Hair, Bruce W. Suter:
Kernel design techniques for alias-free time-frequency distributions. 333-336 - Sergio Barbarossa, Giuseppe Schiappa:
Analysis of multicomponent signals by multilinear time-frequency representations. 337-340 - Srivathsan Krishnamachari, William J. Williams:
Adaptive kernel design in the generalized marginals domain for time-frequency analysis. 341-344 - Morgan J. Arnold, Mark Roessgen, Boualem Boashash:
Filtering real signals through frequency modulation and peak detection in the time-frequency plane. 345-348 - Unto K. Laine, Matti Karjalainen, Toomas Altosaar:
Warped linear prediction (WLP) in speech and audio processing. 349-352 - Ashruf S. El-Dinary, Timothy D. Cole:
Non-recursive FM demodulation of laser radar backscatter using time-frequency distributions. 353-356 - Richard G. Baraniuk:
Beyond time-frequency analysis: energy densities in one and many dimensions. 357-360
Adaptive Filtering Applications and LS Implementation
- Shin'ichi Koike:
Analysis of an echo canceller with an adaptive FIR filter using the Sign algorithm for Gaussian transmit and receive signals. 361-364 - Ramin A. Nobakht:
An interactive solution to adaptive phase jitter cancellation. 365-368 - Mahlar Palkar, José C. Príncipe:
Echo cancellation with the gamma filter. 369-372 - Shoji Makino, Yutaka Kaneda:
A new RLS algorithm based on the variation characteristics of a room impulse response. 373-376 - Kyung Y. Yoo, Nancy Hubing:
The WRLS algorithm for speech processing. 377-380 - Leonardo S. Resende, João Marcos Travassos Romano, Maurice G. Bellanger:
A robust FLS algorithm for linearly-constrained adaptive filtering. 381-384 - Constantin Papaodysseus, D. Gorgoyannis, Elias Koukoutsis, Panayiotis Rousopoulos:
A very robust, fast, parallelizable adaptive least squares algorithm with excellent tracking abilities. 385-388 - Hideaki Sakai, Manabu Kuroda:
A vectorized systolic array for block constrained RLS. 389-392 - Bernard A. Schnaufer, W. Kenneth Jenkins:
A fault tolerant FIR adaptive filter based on the FFT. 393-396 - Jenq-Tay Yuan:
Least-squares lattice interpolation filters. 397-400
Adaptive Filters: LMS, 2D and Transform Methods
- Sang-Sik Ahn, Peter J. Voltz:
Almost-sure convergence of the non-homogeneous DNLMS algorithm with decreasing step size. 401-404 - John Homer:
The curse of dimension on the learning rate of the LMS adaptive FIR filter. 405-408 - Srinath Hosur, Ahmed H. Tewfik, Daniel Boley:
Generalized URV subspace tracking LMS algorithm. 409-412 - Kiyoshi Nishikawa, Hitoshi Kiya:
An LS based new gradient type adaptive algorithm-least squares gradient. 413-416 - Chien-Cheng Tseng, Soo-Chang Pei:
Two dimensional adaptive filter using Laguerre function. 417-420 - K. F. Wan, P. C. Ching:
On the optimality of convergence behaviour for transform-domain split-path adaptive filter. 421-424 - Terence Wang, Chin-Liang Wang:
A new two-dimensional block adaptive FIR filtering algorithm. 425-428 - Scott C. Douglas:
Mean-square analysis of the multiple-error and block LMS adaptive algorithms. 429-432 - Umashankar Iyer, Majid Nayeri, Hiroshi Ochi:
A poly-phase structure for system identification and adaptive filtering. 433-436
Fast Algorithms
- Lap-Pui Chau, Wan-Chi Siu:
Efficient formulation for the realization of discrete cosine transform using recursive structure. 437-440 - Jürgen Götze:
Monitoring the stage of diagonalization in Jacobi-type methods. 441-444 - Haitao Guo, Gary A. Sitton, C. Sidney Burrus:
The quick discrete Fourier transform. 445-448 - Erkan Dorken, S. Hamid Nawab:
Frame-adaptive techniques for quality versus efficiency tradeoffs in STFT analysis. 449-452 - Ali Saidi:
Decimation-in-time-frequency FFT algorithm. 453-456 - Zhongkan Liu, Mingyong Zhou, Hiromitsu Hama:
Maximum Ch-entropy estimation of p-adic stationary process and its fast algorithm. 457-460 - Anissa Zergaïnoh-Mokraoui, Pierre Duhamel, Jean Pierre Vidal:
Efficient implementation of composite length fast FIR filtering on the "ADSP-2100". 461-464 - Daniel Pak-Kong Lun:
On efficient software realization of the prime factor discrete cosine transform. 465-468 - Zhongde Wang, Graham A. Jullien, William C. Miller:
A regular recursive algorithm for the discrete sine transform. 469-472 - Ryszard Stasinski, Tor A. Ramstad:
Fast algorithms for interpolation and decimation filter banks. 473-476
Adaptive Methods for Non-FIR and Nonlinear Structures
- Mu-Huo Cheng, Virginia L. Stonick:
Convergence, convergence point and convergence rate for Steiglitz-McBride method; a unified approach. 477-480 - Carlos Aurélio Faria da Rocha, Odile Macchi:
A novel self-learning adaptive recursive equalizer with unique optimum for QAM. 481-484 - Robert H. T. Jongwe:
Fuzzy based system identification. 485-488 - Junghsi Lee, V. John Mathews:
Adaptive bilinear predictors. 489-492 - Tokunbo Ogunfunmi, Zhuobin Chen:
Neural network algorithms based on the QR decomposition method of least squares. 493-496 - Junibakti Sanubari, Keiichi Tokuda, Mahoki Onoda:
Robust recursive spectral estimation based on AR model excited by a t-distribution process. 497-500 - Virginia L. Stonick, Mu-Huo Cheng:
Adaptive IIR filtering: composite prefiltered regressor method. 501-504 - Corey T. Wangler, Colin H. Hansen:
Genetic algorithm adaptation of non-linear filter structures for active sound and vibration control. 505-508
Nonlinear Systems
- Bernard Mulgrew:
Orthonormal functions for nonlinear signal processing and adaptive filtering. 509-512 - John Tsimbinos, Kenneth V. Lever:
Sampling frequency requirements for identification and compensation of nonlinear systems. 513-516 - Walter A. Frank:
MMD-an efficient approximation to the 2nd order Volterra filter. 517-520 - Maciej Ogorzalek, Hervé Dedieu:
Chaotic signal processing via unstable cycle extraction. 521-524 - Barbara F. La Scala, Robert R. Bitmead:
Design of an extended Kalman filter frequency tracker. 525-528 - Engsiong Chng, Sheng Chen, Bernard Mulgrew:
Reducing the computational requirement of the orthogonal least squares algorithm. 529-532 - Xiao Zhao, Vasilis Z. Marmarelis:
Kernel invariance method for relating continuous-time with discrete-time nonlinear parametric models. 533-536 - Murtaza Ali, Ahmed H. Tewfik:
Multiscale difference equation signal modeling and analysis techniques. 537-540 - John T. Stonick, Jim L. Rulla, Sasan H. Ardalan:
Look-ahead decision-feedback ΣΔ modulation. 541-544
Filter Design
- Jose Antonio Barreto, C. Sidney Burrus:
Lp-complex approximation using iterative reweighted least squares for FIR digital filters. 545-548 - Ashraf Alkhairy:
A complex Chebyshev approximation algorithm for FIR filter design. 549-552 - Ching-Yih Tseng:
A generalized Remez multiple exchange algorithm for complex FIR filter design. 553-556 - Tarek Tutunji, Victor E. DeBrunner:
Pareto optimal designs of low sensitivity digital filters: parallel and cascade form structures. 557-560 - Tolga Çiloglu, Zafer Ünver:
Design of FIR filters with powers of two coefficients based on a new quantization quality criterion. 561-564 - Beth A. Weisburn, Thomas W. Parks, Ram G. Shenoy:
Error criteria for filter design. 565-568 - Ashraf Alkhairy:
An efficient method for IIR filter design. 569-572 - Alan N. Willson Jr., Henry John Orchard:
An improvement to the Powell and Chau linear phase IIR filters. 573-576 - Max Gerken:
Allpass transfer functions with prescribed group delay. 577-580 - Gang Li:
Minimization of pole/zero sensitivity in digital filter design with sparse structure consideration. 581-584 - S. Radhakrishnan Pillai, Gregory H. Allen:
Generalized magnitude and power complementary filters. 585-588
Filter Theory and Analysis
- Gregory H. Allen:
Perturbation effects on filters having multiple poles. 589-592 - Bartlomiej Beliczynski, Jeremi Gryka, Izzet Kale:
Critical comparison of Hankel-norm optimal approximation and balanced model truncation algorithms as vehicles for FIR-to-IIR filter order reduction. 593-596 - Johnson Ihyeh Agnomua:
Generalised running DHT and real-time (DHT) analysers. 597-600 - Chimin Tsai, Meng-Liang Lin, Adly T. Farn:
On the sensitivity of realizations of real coefficient digital filters using complex arithmetic. 601-604 - Wei Xing Zheng, Antonio Cantoni, Kok Lay Teo:
The sensitivity of envelope-constrained filters with uncertain input. 605-608 - Peter H. Bauer:
Asymptotic behavior of digital filters with block floating point arithmetic. 609-612 - Tohru Kiryu, Hidekazu Kaneko, Yoshiaki Saitoh:
Artifact elimination using fuzzy rule based adaptive nonlinear filter. 613-616 - Magdy T. Hanna:
Windows with rapidly decaying sidelobes and steerable sidelobe dips. 617-620 - Gerald D. Cain, N. Paul Murphy, Andrzej Tarczynski:
Evaluation of several variable FIR fractional-sample delay filters. 621-624 - Risto Suoranta, Kari-Pekka Estola, Seppo Rantala, Heli Väätäjä:
PDF estimation using order statistic filter bank. 625-628
Volume 4
Computation and Algorithms
- Michael D. Zoltowski, Gregory M. Kautz:
Novel multirate processing of beamspace noise eigenvectors. 1-4 - Mohammad Usman, Alfred O. Hero III:
Recursive CR bounds: algebraic and statistical acceleration. 5-8 - Ana I. Pérez-Neira, Miguel Angel Lagunas:
Multiuser array beamforming based on a neural network mapping. 9-12 - Aydin Akan, Luis F. Chaparro:
Adaptive time-varying parametric modeling. 13-16 - Peter O'Shea:
Fast parameter estimation algorithms for linear FM signals. 17-20 - Donald W. Tufts, Abhijit A. Shah:
Rank determination in time-series analysis. 21-24 - Haesun Park, Sabine Van Huffel, Lars Eldén:
Fast algorithms for exponential data modeling. 25-28 - Alain Marsal, Sylvie Marcos:
A reduced complexity ESPRIT method and its generalization to an antenna of partially unknown shape. 29-32 - Bin Yang, Frank Gersemsky:
An adaptive algorithm of linear computational complexity for both rank and subspace tracking. 33-36 - Xing Li, Weiru Fang, Qi Tian:
Error criteria analysis and robust data fusion. 37-40
Stochastic Filtering and Estimation
- Nathalie Delfosse, Philippe Loubaton:
Adaptive separation of independent sources: a deflation approach. 41-44 - Hiroshi Kanai, Kazuhiro Ikikame, Noriyoshi Chubachi:
A tapered SVD without rank determination for estimation of multipulse input time series from noisy output. 45-48 - Piet M. T. Broersen:
On bias in transfer functions estimated with stochastic excitation. 49-52 - William A. Gardner, Jeffrey L. Schenck, Stephan V. Schell:
Programmable blind adaptive spatial filtering. 53-56 - Chong-Yung Chi, Jian-Lin Hwang:
A new cumulant based parameter estimation method for noncausal autoregressive systems. 57-60 - Kie-Bum Eom, Rama Chellappa:
Hierarchical stochastic modelling for speech compression. 61-64 - Montse Nájar, Miguel Angel Lagunas, Ignasi Bonet:
Blind wideband source separation. 65-68 - Mohamed A. Deriche:
AR parameter estimation from noisy data using the EM algorithm. 69-72
Arrays: Applications and Processing with Uncertainty
- Nikolaos Fistas, Athanassios Manikas:
A new general global array calibration method. 73-76 - Bill M. Radich, Kevin M. Buckley:
EEG dipole localization bounds for head models with parameter uncertainties. 77-80 - Jean-Jacques Fuchs:
Array shape reconstruction for a nominally linear array. 81-84 - Mats Viberg, A. Lee Swindlehurst:
A Bayesian approach to direction finding with parametric array uncertainty. 85-88 - Jian Li, Petre Stoica:
Efficient parameter estimation of partially polarized electromagnetic waves. 89-92 - Doug A. Gray, Benjamin J. Slocumb, Stephen D. Elton:
Parameter estimation for periodic discrete event processes. 93-96 - Derek Gerlach, Arogyaswami Paulraj:
Spectrum reuse using transmitting antenna arrays with feedback. 97-100 - Ayman F. Naguib, Babak Hossein Khalaj, Arogyaswami Paulraj, Thomas Kailath:
Adaptive channel equalization for TDMA digital cellular communications using antenna arrays. 101-104
Wavelets and Nonlinear Systems
- Russell D. Priebe, Kevin W. Baugh:
Wavelet based detectors. 105-108 - Ahmed H. Tewfik:
Wavelet domain bearing estimation in unknown correlated noise. 109-112 - Hiromichi Yasuoka, Masaaki Ikehara:
2-dimensional recursive orthogonal wavelet transformation. 113-116 - John E. Gilbert, Joseph D. Lakey:
Wavelets of composite type. 117-120 - Nurgun Erdol, Feng Bao, Filiz Basbug:
Optimal receiver design with wavelet bases. 121-124 - Lance M. Kaplan, C.-C. Jay Kuo:
Signal modeling using increments of extended self-similar processes. 125-128 - Benjamin J. Slocumb, John Kitchen:
A polynomial phase parameter estimation-phase unwrapping algorithm. 129-132 - Alba Pagès-Zamora, Miguel Angel Lagunas:
A novel architecture to model non-linear systems. 133-136 - Robert D. Nowak, Barry D. Van Veen:
Volterra filtering with spectral constraints. 137-140
Cyclostationary Signal Processing
- Amod V. Dandawate, Georgios B. Giannakis:
Extraction of almost periodic signals using cyclostationarity. 141-144 - Josep Sala-Alvarez, Gregori Vázquez-Grau:
Separation of digital communication signals through joint space-time decorrelation. 145-148 - Detlef König, Johann F. Böhme:
Application of cyclostationary and time-frequency signal analysis to car engine diagnosis. 149-152 - Guotong Zhou, Georgios B. Giannakis:
Self coupled harmonics: stationary and cyclostationary approaches. 153-156 - Qiang Wu, Kon Max Wong:
Adaptive beamforming of cyclic signal and fast implementation. 157-160 - Randy S. Roberts, Herschel H. Loomis Jr.:
Computational balance in real-time cyclic spectral analysis. 161-164 - Bart F. Rice, Scott R. Smith, Richard A. Threlkeld:
A neural network classifier for cyclostationary signals. 165-168
DSP Applications II
- Saman S. Abeysekera:
Optimum stuff threshold modulation schemes for digital data transmission. 169-172 - Sachin Ambike, Dimitrios Hatzinakos:
Three receiver structures and their performance analyses for binary signalling in a mixture of Gaussian and α-stable impulsive noises. 173-176 - Otmar Loffeld:
Demodulation of noisy phase or frequency modulated signals with Kalman filters. 177-180 - Garry A. Einicke, Langford B. White:
The extended H∞ filter-a robust EKF. 181-184 - José A. R. Fonollosa, Josep Vidal:
Application of hidden Markov models to blind channel characterization and data detection. 185-188 - Dragan Klimovski, Alex A. Sergejew, Antonio L. Cricenti, Greg K. Egan:
Estimation of the position of electrocortical generators via subspace techniques. 189-192 - Robert Prandolini, Miles Moody:
Dynamic time-warp compensation for correlation of long sequences. 193-196 - Jonggil Lee:
Robust estimation of mean Doppler frequency for the measurement of average wind velocity in a weather radar. 197-200
Statistical Array Processing
- Dean McArthur, James P. Reilly:
A computationally efficient self-calibrating direction-of-arrival estimator. 201-204 - Antony P.-C. Ng, Gim Pew Quek:
Unknown signal wavelength and array processing. 205-208 - Wenyuan Xu, Mostafa Kaveh:
Alternatives for the definition and evaluation of resolution thresholds of signal-subspace parameter estimators. 209-212 - Chi Cheng, Yingbo Hua:
Performance analysis of MUSIC and Pencil-MUSIC algorithms for diversely-polarized array. 213-216 - Petre Stoica, Mats Viberg, Björn E. Ottersten, Thomas Kailath:
Optimal localization of partially known signals in unknown noise fields. 217-220 - Debasis Sengupta, Sarbani Palit:
Arrival angle estimation in non-Gaussian noise. 221-224 - James B. Evans, Yingbo Hua:
Further study of the SDMP method for parameter estimation of multiple transients. 225-228 - Martin Haardt, Markus E. Ali-Hackl:
Unitary ESPRIT: how to exploit additional information inherent in the relational invariance structure. 229-232 - Arnab K. Shaw, Wei Xia:
High-resolution direction of arrival estimation using minimum-norm method without eigendecomposition. 233-236 - Sylvie Marcos, Alain Marsal, Messaoud Benidir:
Performances analysis of the propagator method for source bearing estimation. 237-240 - Geoff Poulton:
Array pattern estimation from amplitude measurements on arbitrary near-field surfaces. 241-244
Adaptive Arrays
- Peter Strobach:
Low rank detection of multichannel Gaussian signals using a constrained inverse. 245-248 - Luis Castedo, Ching-Yih Tseng, Aníbal R. Figueiras-Vidal, Lloyd J. Griffiths:
Linearly-constrained adaptive beamforming using cyclostationary signal properties. 249-252 - Keith A. Burgess, Barry D. Van Veen:
A subspace GLRT for vector-sensor array detection. 253-256 - Qiang Wu, Kon Max Wong:
Array signal number detection for coherent and incoherent signals in unknown noise environments. 257-260 - Hsien-Tsai Wu, Jar-Ferr Yang, Fwu-Kuen Chen:
Source number estimator using Gerschgorin disks. 261-264 - Jiankan Yang, Soumendra Daas, A. Lee Swindlehurst:
Improved signal copy with partially known or unknown array response. 265-268 - Sofiène Affes, Saeed Gazor, Yves Grenier:
Robust adaptive beamforming via LMS-like target tracking. 269-272 - Jean-François Cardoso, Adel Belouchrani, Beate H. Laheld:
A new composite criterion for adaptive and iterative blind source separation. 273-276 - Karim Abed-Meraim, Adel Belouchrani, Jean-François Cardoso, Eric Moulines:
Asymptotic performance of second order blind separation. 277-280 - J. Scott Goldstein, Mary Ann Ingram, E. Jeff Holder, Richard N. Smith:
Adaptive subspace selection using subband decompositions for sensor array processing. 281-284
Nonstationary and Time-Frequency
- Syed Ismail Shah, Luis F. Chaparro, A. Salim Kayhan:
Evolutionary maximum entropy spectral analysis. 285-288 - C. S. Detka, Amro El-Jaroudi:
The transitory evolutionary spectrum. 289-292 - Gordon Frazer, Boualem Boashash:
Multiple window spectrogram and time-frequency distributions. 293-296 - Akbar M. Sayeed, Douglas L. Jones:
Optimal kernels for Wigner-Ville spectral estimation. 297-300 - Javier Rodríguez Fonollosa, Chrysostomos L. Nikias:
A new positive time-frequency distribution. 301-304 - Mark L. Brown, William J. Williams, Alfred O. Hero III:
Non-orthogonal Gabor representation of biological signals. 305-308 - Moeness G. Amin, James F. Carroll:
Time-frequency kernel design via point and derivative constraints. 309-312 - David C. Reid, Jonathon C. Ralston:
An optimal window length for the PWVD with application to passive acoustic parameter estimation. 313-316 - François Auger, Patrick Flandrin:
Generalization of the reassignment method to all bilinear time-frequency and time-scale representations. 317-320 - Langford B. White, Peter J. Sherman:
Periodic uncertainty in periodic spectral analysis of processes associated with periodic phenomena. 321-324 - Boualem Boashash, Branko Ristic:
Time-varying higher-order cumulant spectra: application to the analysis of composite FM signals in multiplicative and additive noise. 325-328 - Yumi Takizawa, Keisuke Oda, Atsushi Fukasawa:
Instantaneous spectral estimation of nonstationary signals. 329-332
Detection and Estimation I
- Mehmet Karan, Brian D. O. Anderson, Robert C. Williamson:
A simple calculation of the joint moments of hidden Markov models. 333-336 - Vikram Krishnamurthy:
Adaptive estimation of hidden nearly completely decomposable Markov chains. 337-340 - Barry G. Quinn, Ross F. Barrett, Stephen J. Searle:
The estimation and HMM tracking of weak narrowband signals. 341-344 - Douglas E. Johnston, Petar M. Djuric:
An efficient Bayes solution to AR signal modelling for short sequences. 345-348 - Ki Yong Lee, Byung-Gook Lee, Souguil Ann, Iickho Song:
Robust recursive estimation for linear systems with non-Gaussian state and measurement noises. 349-352 - Wayne T. Padgett, Douglas B. Williams:
Detection of nonstationary random signals in colored noise. 353-356 - Langford B. White:
Robust approximate likelihood ratio tests for nonlinear dynamic systems. 357-360 - Peter M. Schultheiss, Lal C. Godara:
Detection of weak stochastic signals in non-Gaussian noise: a general result. 361-364
Stochastic Filtering and System Identification
- Fernando D. Nunes, José M. N. Leitão:
Statistical signal processing in broadband reflectometry. 365-368 - Weimin Zhang:
A frequency domain filtering method for generation of long complex Gaussian sequences with required spectra. 369-372 - Jyh-Ming Kuo, Samel Çelebi:
Adaptation of memory depth in the gamma filter. 373-376 - S. Unnikrishna Pillai, Won Cheol Lee:
Parametrization of stable systems from impulse response data. 377-380 - Heli Väätäjä, Risto Suoranta, Seppo Rantala:
Coherence analysis of multichannel time series applying conditioned multivariate autoregressive spectra. 381-384 - Tomás Oliveira e Silva:
On the equivalence between Gamma and Laguerre filters. 385-388 - Michel Krob, Messaoud Benidir:
A criterion for the separation of a linear-quadratic mixture of independent components. 389-392 - Jebu J. Rajan, Peter J. W. Rayner:
Bayesian model order selection for the Karhunen-Loeve transform and the singular value decomposition. 393-396 - Seth D. Silverstein:
Pseudo roots and the effects of model order overestimation on the ESPRIT algorithm. 397-400 - David J. Reader, Jason B. Scholz:
Blind maximum likelihood sequence estimation for fading channels. 401-404
Higher Order Statistics
- Md. Anisur Rahman, Kai-Bor Yu:
An efficient implementation of a nonlinear predictor using a zero-memory nonlinearity followed by a second order Volterra filter. 405-408 - Jaume Riba-Sagarra, Gregori Vázquez-Grau:
Recursive Bayes risk parameter estimation from the cyclic autocorrelation matrix. 409-412 - Amro El-Jaroudi, Tayfun Akgül, Marwan A. Simaan:
Application of higher order spectra to multi-scale deconvolution of sensor array signals. 413-416 - Shankar Prakriya, Dimitrios Hatzinakos:
Identification of parametric linear models with cyclostationary inputs. 417-420 - Mariano García Otero, Jose A. Moral-Beneitez:
A lattice structure for the detection of a non-Gaussian signal in Gaussian AR noise. 421-424 - Tariq S. Durrani, Abdul Rahim Leyman, John J. Soraghan:
"Whiter than white" noise. 425-428 - John M. M. Anderson:
Nonlinear system identification using a Hammerstein model and a cumulant-based Steiglitz-McBride algorithm. 429-432 - Ananthram Swami:
Performance analysis of some cumulant-based estimators: harmonics in noise. 433-436 - Edward J. Powers, Sungbin Im:
The utilization of higher-order spectra to determine nonlinear radar cross sections. 437-440 - Jitendra K. Tugnait:
Parameter identifiability of multichannel ARMA models of linear non-Gaussian signals via cumulant matching. 441-444 - Vinod Chandran:
On the computation and interpretation of auto- and cross-trispectra. 445-448
Power Spectrum Estimation
- Jean-Pierre Le Cadre, Olivier Trémois:
Phase-only multidimensional spatio-temporal analysis for moving sources. 449-452 - Roman Ugrinovsky:
Maximum entropy algorithm for spectral estimation problem with gaps. 453-456 - Kenneth J. Pope, Peter J. W. Rayner:
Non-linear system identification using Bayesian inference. 457-460 - Filiep Vanpoucke, Marc Moonen, Yannick Berthoumieu:
An efficient subspace algorithm for 2-D harmonic retrieval. 461-464 - Anthony G. Constantinides, Tania Stathaki:
Complex interpolation for rational orthogonal signal approximation with applications. 465-468 - Yasemin Yardimci, James A. Cadzow, A. Enis Çetin:
Robust signal modeling through nonlinear least squares. 469-472 - Hiroshi Kanai, Noriyoshi Chubachi:
Accurate estimation of AR model by tapered SVD without rank determination. 473-476 - Jiun-Horng Deng, Jeng-Kuang Hwang:
Enhanced state-space method for high resolution estimation of multiple 2-D coherent sinusoids. 477-480 - Jan-Olof Gustavsson, Per Ola Börjesson:
A simultaneous maximum likelihood estimator based on a generalized matched filter. 481-484 - Aleksandar Kavcic, Bin Yang:
A new efficient subspace tracking algorithm based on singular value decomposition. 485-488
Detection and Estimation II
- Hui Liu, Fu Li:
Maximum likelihood velocity estimation of multiple seismic wavefronts. 489-492 - Anthony Quinn:
New lower bounds to the variance of signal parameter estimators using Bayesian inference. 493-496 - Stella N. Batalama, Demetrios Kazakos:
Generalized Cramer-Rao bound and the location parameter case. 497-500 - Chao-Ming Cho, Petar M. Djuric:
Detection and estimation of multiple cisoids in colored noise by Bayesian predictive densities. 501-504 - Petar M. Djuric:
A MAP solution to off-line segmentation of signals. 505-508 - Hongyi Wang, K. J. Ray Liu, Henry Anderson:
Spatial smoothing for arrays with arbitrary geometry. 509-512 - Joseph Ó Ruanaidh, William J. Fitzgerald, Kenneth J. Pope:
Recursive Bayesian location of a discontinuity in time series. 513-516 - A. Satish, Rangasami L. Kashyap:
Wideband multiple target tracking. 517-520 - José M. N. Leitão, José M. F. Moura:
Nonlinear phase estimators based on the Kullback distance. 521-524 - Sanjeev Tavathia, John F. Doherty:
Robust signal estimation using H∞ criteria. 525-528