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ORCIDICASSP 1999: Phoenix, Arizona, USA
- Proceedings of the 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '99, Phoenix, Arizona, USA, March 15-19, 1999. IEEE Computer Society 1999, ISBN 0-7803-5041-3
Volume 1
CELP Coding
- Paul Mermelstein, Yasheng Qian:
Analysis by synthesis speech coding with generalized pitch prediction. 1-4 - Pierre Combescure, Jürgen Schnitzler, Kyrill A. Fischer, Ralf Kirchherr, Claude Lamblin, Alain Le Guyader, Dominique Massaloux, Catherine Quinquis, Joachim Stegmann, Peter Vary:
A 16, 24, 32 kbit/s wideband speech codec based on ATCELP. 5-8 - Stefan Heinen, Marc Adrat, Oliver Steil, Peter Vary, Wen Xu:
A 6.1 to 13.3-kb/s variable rate CELP codec (VR-CELP) for AMR speech coding. 9-12 - Tadashi Amada, Kimio Miseki, Masami Akamine:
CELP speech coding based on an adaptive pulse position codebook. 13-16 - Miguel Arjona Ramírez
, Max Gerken:
A multistage search of algebraic CELP codebooks. 17-20 - Nam Kyu Ha:
A fast search method of algebraic codebook by reordering search sequence. 21-24 - Roar Hagen, Erik Ekudden:
An 8 kbit/s ACELP coder with improved background noise performance. 25-28 - Harald Pobloth, W. Bastiaan Kleijn:
On phase perception in speech. 29-32
Large Vocabulary Recognition
- Steven Wegmann, Puming Zhan, Larry Gillick:
Progress in Broadcast News transcription at Dragon Systems. 33-36 - Scott Saobing Chen, Ellen Eide, Mark J. F. Gales, Ramesh A. Gopinath, Dimitri Kanevsky, Peder A. Olsen:
Recent improvements to IBM's speech recognition system for automatic transcription of broadcast news. 37-40 - Jayadev Billa, Thomas Colthurst, Amro El-Jaroudi, Rukmini Iyer, Kristine W. Ma, Spyridon Matsoukas, Carl Quillen, Fred Richardson, Man-Hung Siu, George Zavaliagkos, Herbert Gish:
Recent experiments in large vocabulary conversational speech recognition. 41-44 - Martine Adda-Decker, Gilles Adda, Jean-Luc Gauvain, Lori Lamel:
Large vocabulary speech recognition in French. 45-48 - Sue E. Johnson, Pierre Jourlin, Gareth L. Moore, Karen Spärck Jones, Philip C. Woodland:
The Cambridge University spoken document retrieval system. 49-52 - Barbara Peskin, Michael Newman, Don McAllaster, Venkatesh Nagesha, Hywel B. Richards, Steven Wegmann, Melvyn J. Hunt, Larry Gillick:
Improvements in recognition of conversational telephone speech. 53-56 - Thomas Hain, Philip C. Woodland, Thomas Niesler, Edward W. D. Whittaker:
The 1998 HTK system for transcription of conversational telephone speech. 57-60 - James R. Glass, Timothy J. Hazen, I. Lee Hetherington:
Real-time telephone-based speech recognition in the Jupiter domain. 61-64
Speech Analysis and Enhancement
- Chung-Hsien Hu, Jau-Hung Chen:
Template-driven generation of prosodic information for Chinese concatenative synthesis. 65-68 - Hiroshi Saruwatari, Shoji Kajita, Kazuya Takeda, Fumitada Itakura:
Speech enhancement using nonlinear microphone array with complementary beamforming. 69-72 - David C. Smith, Jeffrey Townsend, Douglas J. Nelson, Dan Richman:
A multivariate speech activity detector based on the syllable rate. 73-76 - Jeff Kuo, Eva B. Holmberg, Robert E. Hillman:
Discriminating speakers with vocal nodules using aerodynamic and acoustic features. 77-80 - Kenji Matsui, Noriyo Hara:
Enhancement of esophageal speech using formant synthesis. 81-84 - Helen M. Hanson, Richard S. McGowan, Kenneth N. Stevens, Robert E. Beaudoin:
Development of rules for controlling the HLsyn speech synthesizer. 85-88 - Daniel Tapias, Carlos García, Christophe Cazassus:
On the characteristics and effects of loudness during utterance production in continuous speech recognition. 89-92 - Francesco Beritelli, Salvatore Casale, Alfredo Cavallaro:
A multichannel speech/silence detector based on time delay estimation and fuzzy classification. 93-96 - Toshio Irino:
Noise suppression using a time-varying, analysis/synthesis gamma chirp filterbank. 97-100 - Peter Søren Kirk Hansen, Per Christian Hansen, Steffen Duus Hansen, John Aasted Sørensen:
Experimental comparison of signal subspace based noise reduction methods. 101-104
Acoustic Modeling I
- Man-Hung Siu, Michael Jonas, Herbert Gish:
Using a large vocabulary continuous speech recognizer for a constrained domain with limited training. 105-108 - Joseph Picone, Sandi Pike, Roland Reagan, Terri Kamm, John S. Bridle, Li Deng, Z. Ma, Hywel B. Richards, Mike Schuster:
Initial evaluation of hidden dynamic models on conversational speech. 109-112 - Spyros Matsoukas, George Zavaliagkos:
Convolutional density estimation in hidden Markov models for speech recognition. 113-116 - Rita Singh, Bhiksha Raj, Richard M. Stern:
Automatic clustering and generation of contextual questions for tied states in hidden Markov models. 117-120 - Tetsunori Kobayashi, Junko Furuyama, Ken Masumitsu:
Partly hidden Markov model and its application to speech recognition. 121-124 - Jae H. Kim, Raziel Haimi-Cohen, Frank K. Soong:
Hidden Markov models with divergence based vector quantized variances. 125-128 - Yuqing Gao, Ea-Ee Jan, Mukund Padmanabhan, Michael Picheny:
HMM training based on quality measurement. 129-132 - Koji Iwano, Keikichi Hirose:
Prosodic word boundary detection using statistical modeling of moraic fundamental frequency contours and its use for continuous speech recognition. 133-136
ASR Systems and Applications
- Yukikuni Nishida, Yoshio Nakadai, Yoshitake Suzuki, Tetsuma Sakurai, Toshihide Kurokawa, Hirokazu Sato:
Voice recognition focusing on vowel strings on a fixed-point 20-MIPS DSP board. 137-140 - Makoto Shozakai:
Speech interface VLSI for car applications. 141-144 - Stephen W. Anderson, Natalie Liberman, Erica G. Bernstein, Stephen Foster, Erin Cate, Brenda Levin, Randy Hudson:
Recognition of elderly speech and voice-driven document retrieval. 145-148 - Michael J. Carey, Eluned S. Parris, Harvey Lloyd-Thomas:
A comparison of features for speech, music discrimination. 149-152 - Mazin G. Rahim:
Recognizing connected digits in a natural spoken dialog. 153-156 - Dong-Suk Yuk, James L. Flanagan:
Telephone speech recognition using neural networks and hidden Markov models. 157-160 - Ji Ming, Philip Hanna, Darryl Stewart, Marie Owens, Francis Jack Smith:
Improving speech recognition performance by using multi-model approaches. 161-164 - Coimbatore S. Ramalingam, Yifan Gong, Lorin Netsch, Wallace W. Anderson, John J. Godfrey, Yu-Hung Kao:
Speaker-dependent name dialing in a car environment with out-of-vocabulary rejection. 165-168 - Qing Guo, Fang Zheng, Jian Wu, Wenhu Wu:
An new method used in HMM for modeling frame correlation. 169-172 - Patrick Nguyen, Philippe Gelin, Jean-Claude Junqua, Jen-Tzung Chien:
N-best based supervised and unsupervised adaptation for native and non-native speakers in cars. 173-176
Topics in Speech Coding
- Simão Ferraz de Campos Neto, Franklin L. Corcoran, Ara Karahisar:
Performance assessment of tandem connection of enhanced cellular coders. 177-180 - Ki-Seung Lee, Richard V. Cox:
TTS based very low bit rate speech coder. 181-184 - Ling Kok Ng, Gang Li, Xiao Lin, Guoan Bi:
Wideband speech coding with toll quality based on IA-model. 185-188 - Kazunori Ozawa:
4 kb/s multi-pulse based CELP speech coding using excitation switching. 189-192 - Erdal Paksoy, Juan Carlos De Martin, Alan McCree, Christian G. Gerlach, Anand K. Anandakumar, Wai-Ming Lai, Vishu Viswanathan:
An adaptive multi-rate speech coder for digital cellular telephony. 193-196 - Azhar Mustapha, Suat Yeldener:
An adaptive post-filtering technique based on the modified Yule-Walker filter. 197-200 - Anthony J. Accardi, Richard V. Cox:
A modular approach to speech enhancement with an application to speech coding. 201-204 - Gernot Kubin, W. Bastiaan Kleijn:
On speech coding in a perceptual domain. 205-208
Speech Analysis
- Panuthat Boonpramuk, Tetsuo Funada, Noboru Kanedera:
Speech analysis/synthesis/conversion by using sequential processing. 209-212 - Mike Brookes, Han Pin Loke:
Modelling energy flow in the vocal tract with applications to glottal closure and opening detection. 213-216 - Srinivasan Umesh, Leon Cohen, Douglas J. Nelson:
Fitting the Mel scale. 217-220 - Wai Kat Liu, Pascale Fung:
Fast accent identification and accented speech recognition. 221-224 - Howard Hua Yang, Sarel van Vuuren, Hynek Hermansky:
Relevancy of time-frequency features for phonetic classification measured by mutual information. 225-228 - Keiichi Tokuda, Takashi Masuko, Noboru Miyazaki, Takao Kobayashi:
Hidden Markov models based on multi-space probability distribution for pitch pattern modeling. 229-232 - Ashraf Alkhairy:
An algorithm for glottal volume velocity estimation. 233-236 - Khaled El-Maleh, Ara Samouelian, Peter Kabal:
Frame level noise classification in mobile environments. 237-240
Low Bit Rate Speech Coding I
- Nicola R. Chong, Ian S. Burnett, Joe F. Chicharo:
Low delay multi-level decomposition and quantisation techniques for WI coding. 241-244 - Takahiro Unno, Thomas P. Barnwell III, Kwan K. Truong:
An improved mixed excitation linear prediction (MELP) coder. 245-248 - Stephane Villette, Milos Stefanovic, Ahmet M. Kondoz:
Split band LPC based adaptive multi-rate GSM candidate. 249-252 - Milan Jelinek, Jean-Pierre Adoul:
Frequency-domain spectral envelope estimation for low rate coding of speech. 253-256 - Chunyan Li, Vladimir Cuperman, Allen Gersho:
Robust closed-loop pitch estimation for harmonic coders by time scale modification. 257-260 - Hong-Goo Kang, Dipanjan Sen:
Phase adjustment in waveform interpolation. 261-264 - Jongseo Sohn, Wonyong Sung:
A low resolution pulse position coding method for improved excitation modeling of speech transition. 265-268 - Oded Gottesman:
Dispersion phase vector quantization for enhancement of waveform interpolative coder. 269-272
Robust Speech Recognition in Noisy Environments
- Firas Jabloun, A. Enis Çetin:
The Teager energy based feature parameters for robust speech recognition in car noise. 273-276 - Ascensión Gallardo-Antolín, Fernando Díaz-de-María, Francisco J. Valverde-Albacete:
Avoiding distortions due to speech coding and transmission errors in GSM ASR tasks. 277-280 - Mike Peters:
Binaural bark subband pre-processing of nonstationary signals for noise robust speech feature extraction. 281-284 - Satoru Tsuge, Toshiaki Fukuda, Harald Singer:
Speaker normalized spectral subband parameters for noise robust speech recognition. 285-288 - Hynek Hermansky, Sangita Sharma:
Temporal patterns (TRAPs) in ASR of noisy speech. 289-292 - Montri Karnjanadecha, Stephen A. Zahorian:
Signal modeling for isolated word recognition. 293-296 - Yifan Gong, John J. Godfrey:
Transforming HMMs for speaker-independent hands-free speech recognition in the car. 297-300 - Shuen Kong Wong, Bertram E. Shi:
Channel and noise adaptation via HMM mixture mean transform and stochastic matching. 301-304
Speaker Recognition
- Jialong He, Li Liu:
Speaker verification performance and the length of test sentence. 305-308 - Rivarol Vergin, Douglas D. O'Shaughnessy:
On the use of some divergence measures in speaker recognition. 309-312 - Roland Auckenthaler, Eluned S. Parris, Michael J. Carey:
Improving a GMM speaker verification system by phonetic weighting. 313-316 - Yong Gu, Trevor Thomas:
A hybrid score measurement for HMM-based speaker verification. 317-320 - William M. Campbell, Khaled T. Assaleh:
Polynomial classifier techniques for speaker verification. 321-324 - Alvin Garcia, Richard J. Mammone:
Channel-robust speaker identification using modified-mean cepstral mean normalization with frequency warping. 325-328 - Mübeccel Demirekler, Ali Haydar:
Feature selection using genetics-based algorithm and its application to speaker identification. 329-332
Acoustic Modeling II
- Daniel Povey, Philip C. Woodland:
Frame discrimination training for HMMs for large vocabulary speech recognition. 333-336 - Françoise Beaufays, Mitchel Weintraub, Yochai Konig:
Discriminative mixture weight estimation for large Gaussian mixture models. 337-340 - Richard C. Rose, Giuseppe Riccardi:
Modeling disfluency and background events in ASR for a natural language understanding task. 341-344 - Wu Chou, Wolfgang Reichl:
Decision tree state tying based on penalized Bayesian information criterion. 345-348 - Jiayu Li, Alejandro Murua:
A 2D extended HMM for speech recognition. 349-352 - Xiaoqiang Luo, Frederick Jelinek:
Probabilistic classification of HMM states for large vocabulary continuous speech recognition. 353-356 - Hywel B. Richards, John S. Bridle:
The HDM: a segmental hidden dynamic model of coarticulation. 357-360 - Sankar Basu, Charles A. Micchelli, Peder A. Olsen:
Maximum likelihood estimates for exponential type density families. 361-364
Speech Production and Synthesis
- Stephen D. Peters, Peter Stubley, Jean-Marc Valin:
On the limits of speech recognition in noise. 365-368 - Qian-Jie Fu, Robert V. Shannon:
Recognition of spectrally degraded speech in noise with nonlinear amplitude mapping. 369-372 - Robert E. Donovan, Martin Franz, Jeffrey S. Sorensen, Salim Roukos:
Phrase splicing and variable substitution using the IBM trainable speech synthesis system. 373-376 - Yannis Stylianou:
Assessment and correction of voice quality variabilities in large speech databases for concatenative speech synthesis. 377-380 - Darragh O'Brien, Alex I. C. Monaghan:
Shape invariant time-scale modification of speech using a harmonic model. 381-384 - Kim E. A. Silverman, Jerome R. Bellegarda:
Using a sigmoid transformation for improved modeling of phoneme duration. 385-388 - Karthik Narasimhan, José C. Príncipe, Donald G. Childers:
Nonlinear dynamic modeling of the voiced excitation for improved speech synthesis. 389-392 - Arnaud Robert:
Results on perceptual invariants to transformations on speech. 393-396
Feature Extraction
- Reinhold Haeb-Umbach:
Investigations on inter-speaker variability in the feature space. 397-400 - Seung Ho Choi, Hong Kook Kim, Hwang Soo Lee:
LSP weighting functions based on spectral sensitivity and mel-frequency warping for speech recognition in digital communication. 401-404 - Chun-Ping Chan, Yiu Wing Wong, Tan Lee, Pak-Chung Ching:
Two-dimensional multi-resolution analysis of speech signals and its application to speech recognition. 405-408 - Rathinavelu Chengalvarayan:
Hierarchical subband linear predictive cepstral (HSLPC) features for HMM-based speech recognition. 409-412 - Douglas D. O'Shaughnessy, Hesham Tolba:
Towards a robust/fast continuous speech recognition system using a voiced-unvoiced decision. 413-416 - Jhing-Fa Wang, Shi-Huang Chen:
A C/V segmentation algorithm for Mandarin speech signal based on wavelet transforms. 417-420 - Tsuneo Nitta:
Feature extraction for speech recognition based on orthogonal acoustic-feature planes and LDA. 421-424 - Partha Niyogi, Chris Burges, Padma Ramesh:
Distinctive feature detection using support vector machines. 425-428
Robust Speech Recognition and Adaption
- Nam Soo Kim:
Time-varying noise compensation using multiple Kalman filters. 429-432 - Wei-Tyng Hong, Sin-Horng Chen:
A segment-based C0 adaptation scheme for PMC-based noisy Mandarin speech recognition. 433-436 - Jeih-weih Hung, Jia-Lin Shen, Lin-Shan Lee:
Improved parallel model combination techniques with split Gaussian mixtures for speech recognition under noisy conditions. 437-440 - Günther Ruske, Ki Yong Lee:
Speech recognition and enhancement by a nonstationary AR HMM with gain adaptation under unknown noise. 441-444 - Alexander Fischer, Volker Stahl:
Database and online adaptation for improved speech recognition in car environments. 445-448 - Diego Giuliani, Marco Matassoni, Maurizio Omologo, Piergiorgio Svaizer:
Training of HMM with filtered speech material for hands-free recognition. 449-452 - Chafic Mokbel, Olivier Collin:
Incremental enrolment of speech recognizers. 453-456 - Bishnu S. Atal:
Automatic speech recognition: a communication perspective. 457-460
Low Bit Rate Speech Coding II
- Mohammad Reza Nakhai, Farokh A. Marvasti:
Split band CELP (SB-CELP) speech coder. 461-464 - Najam Malik, W. Harvey Holmes:
Log amplitude modeling of sinusoids in voiced speech. 465-468 - Minoru Kohata:
1.2 kbit/s harmonic coder using auditory filters. 469-472 - Jesper Jensen, Søren Holdt Jensen, Egon Hansen:
Exponential sinusoidal modeling of transitional speech segments. 473-476 - Eric W. M. Yu, Cheung-Fat Chan:
Harmonic+noise coding using improved V/UV mixing and efficient spectral quantization. 477-480 - Suat Yeldener:
A 4 kb/s toll quality harmonic excitation linear predictive speech coder. 481-484 - Jacek Stachurski, Alan McCree, Vishu Viswanathan:
High quality MELP coding at bit-rates around 4 kb/s. 485-488 - Thomas Eriksson, Hong-Goo Kang:
Pitch quantization in low bit-rate speech coding. 489-492
Speech Understanding
- Gies Bouwman, Janienke Sturm, Louis Boves:
Incorporating confidence measures in the Dutch train timetable information system developed in the ARISE project. 493-496 - Klaus Ries:
HMM and neural network based speech act detection. 497-500 - Lori Faith Lamel, Sophie Rosset, Jean-Luc Gauvain, Samir Bennacef:
The LIMSI ARISE system for train travel information. 501-504 - Matthew Siegler, Michael Witbrock:
Improving the suitability of imperfect transcriptions for information retrieval from spoken documents. 505-508 - John Golden, Owen Kimball, Man-Hung Siu, Herbert Gish:
Automatic topic identification for two-level call routing. 509-512 - Yoshihiko Gotoh, Steve Renals, Gethin Williams:
Named entity tagged language models. 513-516 - Hermann Ney:
Speech translation: coupling of recognition and translation. 517-520 - Frederick Walls, Hubert Jin, Sreenivasa Sista, Richard M. Schwartz:
Probabilistic models for topic detection and tracking. 521-524
Language Modeling I
- Volker Warnke, Stefan Harbeck, Elmar Nöth, Heinrich Niemann, Michael Levit:
Discriminative estimation of interpolation parameters for language model classifiers. 525-528 - Reinhard Blasig:
Combination of words and word categories in varigram histories. 529-532 - Hirofumi Yamamoto, Yoshinori Sagisaka:
Multi-class composite N-gram based on connection direction. 533-536 - Christer Samuelsson, Wolfgang Reichl:
A class-based language model for large-vocabulary speech recognition extracted from part-of-speech statistics. 537-540 - Milind Mahajan, Doug Beeferman, Xuedong Huang:
Improved topic-dependent language modeling using information retrieval techniques. 541-544 - Sven C. Martin, Hermann Ney, Jörg Zaplo:
Smoothing methods in maximum entropy language modeling. 545-548 - Stanley F. Chen, Ronald Rosenfeld:
Efficient sampling and feature selection in whole sentence maximum entropy language models. 549-552 - Sanjeev Khudanpur, Jun Wu:
A maximum entropy language model integrating N-grams and topic dependencies for conversational speech recognition. 553-556
Volume 2
Acoustic Modeling III
- Cristina Chesta, Pietro Laface, Franco Ravera:
Connected digit recognition using short and long duration models. 557-560 - Kishore Papineni:
Discriminative training via linear programming. 561-564 - Daniel Willett, Christoph Neukirchen, Jörg Rottland, Gerhard Rigoll:
Refining tree-based state clustering by means of formal concept analysis, balanced decision trees and automatically generated model-sets. 565-568 - Stavros Tsakalidis, Vassilios Digalakis, Leonardo Neumeyer:
Efficient speech recognition using subvector quantization and discrete-mixture HMMs. 569-572 - Wolfgang Reichl, Wu Chou:
A unified approach of incorporating general features in decision tree based acoustic modeling. 573-576 - Qiang Huo, Bin Ma:
Irrelevant variability normalization in learning HMM state tying from data based on phonetic decision-tree. 577-580 - Philip McMahon, Naomi Harte, Saeed Vaseghi, Paul M. McCourt:
Discriminative spectral-temporal multiresolution features for speech recognition. 581-584 - Philip Clarkson, Pedro J. Moreno:
On the use of support vector machines for phonetic classification. 585-588
Lexical Issues/Search
- Jian-Xiong Wu, Vishwa Gupta:
Application of simultaneous decoding algorithms to automatic transcription of known and unknown words. 589-592 - Achim Sixtus, Stefan Ortmanns:
High quality word graphs using forward-backward pruning. 593-596 - Carl D. Mitchell, Anand R. Setlur:
Improved spelling recognition using a tree-based fast lexical match. 597-600 - Fang Zheng:
A syllable-synchronous network search algorithm for word decoding in Chinese speech recognition. 601-604 - Qi Li:
A fast, sequential decoding algorithm with application to speaker verification. 605-608 - Klaus Beulen, Stefan Ortmanns, Christian Elting:
Dynamic programming search techniques for across-word modelling in speech recognition. 609-612 - Long Nguyen, Richard M. Schwartz:
Single-tree method for grammar-directed search. 613-616 - Petra Geutner, Michael Finke, Alex Waibel:
Selection criteria for hypothesis driven lexical adaptation. 617-620
Speech Understanding and System
- Mikel Peñagarikano, Germán Bordel, Amparo Varona, Karmele López de Ipiña:
Using non-word lexical units in automatic speech understanding. 621-624 - Katsutoshi Ohtsuki, Sadaoki Furui, Atsushi Iwasaki, Naoyuki Sakurai:
Message-driven speech recognition and topic-word extraction. 625-628 - Edward C. Kaiser, Michael Johnston, Peter A. Heeman:
PROFER: predictive, robust finite-state parsing for spoken language. 629-632 - Marcello Federico, Fabio Brugnara, Roberto Gretter:
Usability field-test of a spoken data-entry system. 633-636 - Bor-Shen Lin, Lin-Shan Lee:
A framework of performance evaluation and error analysis methodology for speech understanding systems. 637-640 - David Llorens, Francisco Casacuberta, Encarna Segarra, Joan-Andreu Sánchez, Pablo Aibar, María José Castro:
Acoustic and syntactical modeling in the ATROS system. 641-644 - Jason Davenport, Richard M. Schwartz, Long Nguyen:
Towards a robust real-time decoder. 645-648 - William M. Fisher:
A statistical text-to-phone function using ngrams and rules. 649-652
Speech Analysis and Quantization
- John J. Parry, Ian S. Burnett, Joe F. Chicharo:
Linguistic mapping in LSF space for low-bit rate coding. 653-656 - Adriana Vasilache, Marcel Vasilache, Ioan Tabus:
Predictive multiple-scale lattice VQ for LSF quantization. 657-660 - Joseph Rothweiler:
A rootfinding algorithm for line spectral frequencies. 661-664 - Bob Novorita:
Incorporation of temporal masking effects into bark spectral distortion measure. 665-668 - Manohar N. Murthi, Bhaskar D. Rao:
MVDR based all-pole models for spectral coding of speech. 669-672 - Wonho Yang, Robert E. Yantorno:
Improvement of MBSD by scaling noise masking threshold and correlation analysis with MOS difference instead of MOS. 673-676 - Per Hedelin, Jan Skoglund, Jonas Samuelsson:
Performance bounds for LPC spectrum quantization. 677-680 - Jan Linden:
Channel optimized predictive VQ. 681-684
Utterance Verification/Acoustic Modeling
- Tatsuya Kawahara, Shuji Doshita:
Topic independent language model for key-phrase detection and verification. 685-688 - Kwok Leung Lam, Pascale Fung:
A more efficient and optimal LLR for decoding and verification. 689-692 - Katrin Kirchhoff, Jeff A. Bilmes:
Dynamic classifier combination in hybrid speech recognition systems using utterance-level confidence values. 693-696 - Yeou-Jiunn Chen, Chung-Hsien Wu, Gwo-Lang Yan:
Utterance verification using prosodic information for Mandarin telephone speech keyword spotting. 697-700 - Hiroshi Matsuo, Masaaki Ishigame:
Error correction for speaker-independent isolated word recognition through likelihood compensation using phonetic bigram. 701-704 - Andreas Wendemuth, Georg Rose, Hans J. G. A. Dolfing:
Advances in confidence measures for large vocabulary. 705-708 - Denis Jouvet, Katarina Bartkova, Guy Mercier:
Hypothesis dependent threshold setting for improved out-of-vocabulary data rejection. 709-712 - Jeff A. Bilmes:
Buried Markov models for speech recognition. 713-716
Language Modeling II
- Jerome R. Bellegarda:
Speech recognition experiments using multi-span statistical language models. 717-720 - Allen L. Gorin, Giuseppe Riccardi:
Spoken language variation over time and state in a natural spoken dialog system. 721-724 - Demetrio Aiello, Cristina Delogu, Renato De Mori, Andrea Di Carlo, Marina Nisi, Silvia Tummeacciu:
Comparative evaluation of spoken corpora acquired by presentation of visual scenarios and textual descriptions. 725-728 - Amparo Varona, Inés Torres:
Using smoothed K-TSS language models in continuous speech recognition. 729-732 - Mary P. Harper, Michael T. Johnson, Leah H. Jamieson, Stephen Hockema, Christopher M. White:
Interfacing a CDG parser with an HMM word recognizer using word graphs. 733-736 - Motoyuki Suzuki, Shozo Makino, Hirotomo Aso:
An automatic acquisition method of statistic finite-state automaton for sentences. 737-740 - Frank Wessel, Andrea Baader:
Robust dialogue-state dependent language modeling using leaving-one-out. 741-744 - Adam Kalai, Stanley F. Chen, Avrim Blum, Ronald Rosenfeld:
On-line algorithms for combining language models. 745-748
Adaptation/Normalization
- Roland Kuhn, Patrick Nguyen, Jean-Claude Junqua, Robert Boman, Nancy Niedzielski, Steven Fincke, Kenneth L. Field, Matteo Contolini:
Fast speaker adaptation using a priori knowledge. 749-752 - Zuoying Wang, Feng Liu:
Speaker adaptation using maximum likelihood model interpolation. 753-756 - John W. McDonough, William J. Byrne:
Speaker adaptation with all-pass transforms. 757-760 - Lutz Welling, Stephan Kanthak, Hermann Ney:
Improved methods for vocal tract normalization. 761-764 - Vassilios Digalakis, Heather Collier, Sid Berkowitz, Adrian Corduneanu, Enrico Bocchieri, Ashvin Kannan, Constantinos Boulis, Sanjeev Khudanpur, William Byrne, Ananth Sankar:
Rapid speech recognizer adaptation to new speakers. 765-768 - Ashvin Kannan, Sanjeev Khudanpur:
Tree-structured models of parameter dependence for rapid adaptation in large vocabulary conversational speech recognition. 769-772 - Enrico Bocchieri, Vassilios Digalakis, Adrian Corduneanu, Constantinos Boulis:
Correlation modeling of MLLR transform biases for rapid HMM adaptation to new speakers. 773-776 - Prabhu Raghavan, Richard J. Renomeron, ChiWei Che, Dong-Suk Yuk, James L. Flanagan:
Speech recognition in a reverberant environment using matched filter array (MFA) processing and linguistic-tree maximum likelihood linear regression (LT-MLLR) adaptation. 777-780
Speech Enhancement
- Éric Grivel, Marcel Gabrea, Mohamed Najim:
Subspace state space model identification for speech enhancement. 781-784 - Driss Matrouf, Jean-Luc Gauvain:
Using AR HMM state-dependent filtering for speech enhancement. 785-788 - David Malah, Richard V. Cox, Anthony J. Accardi:
Tracking speech-presence uncertainty to improve speech enhancement in non-stationary noise environments. 789-792 - Chuang He, George Zweig:
Adaptive two-band spectral subtraction with multi-window spectral estimation. 793-796 - Anisa Yasmin, Paul W. Fieguth, Li Deng:
Speech enhancement using voice source models. 797-800 - Kuan-Chieh Yen, Yunxin Zhao:
Adaptive decorrelation filtering for separation of co-channel speech signals from m>2 sources. 801-804 - David V. Anderson, Mark A. Clements:
Audio signal noise reduction using multi-resolution sinusoidal modeling. 805-808 - Chang D. Yoo:
Utilizing interband acoustical information for modeling stationary time-frequency regions of noisy speech. 809-812
Topics in Speaker and Language Recognition
- Lisa Yanguas, Thomas F. Quatieri:
Implications of glottal source for speaker and dialect identification. 813-816 - Jack McLaughlin, Douglas A. Reynolds, Elliot Singer, Gerald C. O'Leary:
Automatic speaker clustering from multi-speaker utterances. 817-820 - Ivan Magrin-Chagnolleau, Aaron E. Rosenberg, Sarangarajan Parthasarathy:
Detection of target speakers in audio databases. 821-824 - Olivier Siohan, Chin-Hui Lee, Arun C. Surendran, Qi Li:
Background model design for flexible and portable speaker verification systems. 825-828 - Joseph P. Campbell, Douglas A. Reynolds:
Corpora for the evaluation of speaker recognition systems. 829-832 - François Pellegrino, Régine André-Obrecht:
An unsupervised approach to language identification. 833-836 - Bryan L. Pellom, John H. L. Hansen:
An experimental study of speaker verification sensitivity to computer voice-altered imposters. 837-840 - Toshihiro Isobe, Jun-ichi Takahashi:
A new cohort normalization using local acoustic information for speaker verification. 841-844 - Li Liu, Jialong He:
On the use of orthogonal GMM in speaker recognition. 845-848 - Stephen A. Zahorian:
Reusable binary-paired partitioned neural networks for text-independent speaker identification. 849-852
Echo Cancellation and Noise Control
- Jacob Benesty, Dennis R. Morgan, Joseph L. Hall, M. Mohan Sondhi:
Synthesized stereo combined with acoustic echo cancellation for desktop conferencing. 853-856 - Suehiro Shimauchi, Shoji Makino, Yoichi Haneda, Akira Nakagawa, Sumitaka Sakauchi:
A stereo echo canceller implemented using a stereo shaker and a duo-filter control system. 857-860 - Akihiro Hirano, Kenji Nakayama, Kazunobu Watanabe:
Convergence analysis of stereophonic echo canceller with pre-processing-relation between pre-processing and convergence. 861-864 - Walter A. Frank, Imre Varga:
Implicit decimation for FIR systems and its application to acoustic echo cancellation. 865-868 - Eric A. Woudenberg, Frank K. Soong, Biing-Hwang Juang:
A block least squares approach to acoustic echo cancellation. 869-872 - Stefan Gustafsson, Peter Jax, Axel Kamphausen, Peter Vary:
A postfilter for echo and noise reduction avoiding the problem of musical tones. 873-876 - Alexander Stenger, Lutz Trautmann, Rudolf Rabenstein:
Nonlinear acoustic echo cancellation with 2nd order adaptive Volterra filters. 877-880 - Biljana D. Radlovic, Robert C. Williamson, Rodney A. Kennedy:
On the poor robustness of sound equalization in reverberant environments. 881-884 - Maria de Diego, Alberto González, Clemente Garcia:
On the performance of a local active noise control system. 885-888 - Wai Kuen Lai, Ting Wai Siu, Sze-Fong Yau:
High quality signal reception in the presence of stationary interference-a blind signal separation approach. 889-892
Coding
- Aki Härmä, Unto K. Laine, Matti Karjalainen:
On the utilization of overshoot effects in low-delay audio coding. 893-896 - Akio Jin, Takehiro Moriya, Takeshi Norimatsu, Mineo Tsushima, Tomokazu Ishikawa:
Scalable audio coder based on quantizer units of MDCT coefficients. 897-900 - Trevor R. Trinkaus, Mark A. Clements:
An algorithm for compression of wideband diverse speech and audio signals. 901-904 - Yuan-Hao Huang, Tzi-Dar Chiueh:
A new forward masking model and its application to perceptual audio coding. 905-908 - Markus Erne, George S. Moschytz, Christof Faller:
Best wavelet-packet bases for audio coding using perceptual and rate-distortion criteria. 909-912 - Hossein Najafzadeh-Azghandi, Peter Kabal:
Improving perceptual coding of narrowband audio signals at low rates. 913-916 - Christopher A. Lanciani, Ronald W. Schafer:
Subband-domain filtering of MPEG audio signals. 917-920
Auditory Modeling, Hearing Aids and Applications of Signal Processing to Audio and Acoustics
- Sigisbert Wyrsch, August Kaelin:
Adaptive feedback cancelling in subbands for hearing aids. 921-924 - Marcio G. Siqueira, Abeer Alwan:
Bias analysis in continuous adaptation systems for hearing aids. 925-928 - Matti Karjalainen, Tero Tolonen:
Multi-pitch and periodicity analysis model for sound separation and auditory scene analysis. 929-932 - Lisa C. Gresham, Leslie M. Collins:
A comparison using signal detection theory of the ability of two computational auditory models to predict experimental data. 933-936 - Yiteng Huang, Jacob Benesty, Gary W. Elko:
Adaptive eigenvalue decomposition algorithm for real time acoustic source localization system. 937-940 - Michiaki Omura, Motohiko Yada, Hiroshi Saruwatari, Shoji Kajita, Kazuya Takeda, Fumitada Itakura:
Compensating of room acoustic transfer functions affected by change of room temperature. 941-944 - Thomas F. Quatieri, Thomas E. Hanna:
'Perfect reconstruction' time-scaling filterbanks. 945-948 - Osamu Hoshuyama, Akihiko Sugiyama:
An adaptive microphone array with good sound quality using auxiliary fixed beamformers and its DSP implementation. 949-952 - Michael S. Brandstein:
An event-based method for microphone array speech enhancement. 953-956
Spatial Audio
- Rudolf Rabenstein, Ahlem Zayati:
A direct method to computational acoustics. 957-960 - Corey I. Cheng, Gregory H. Wakefield:
Spatial frequency response surfaces: an alternative visualization tool for head-related transfer functions (HRTFs). 961-964 - Richard O. Duda, Carlos Avendaño, V. Ralph Algazi:
An adaptable ellipsoidal head model for the interaural time difference. 965-968 - Harvey F. Silverman, William R. Patterson III:
Visualizing the performance of large-aperture microphone arrays. 969-972
Music Applications
- Lauri Savioja, Vesa Välimäki:
Reduction of the dispersion error in the interpolated digital waveguide mesh using frequency warping. 973-976 - Vesa Välimäki, Tero Tolonen, Matti Karjalainen:
Plucked-string synthesis algorithms with tension modulation nonlinearity. 977-980 - Tony S. Verma, Teresa H.-Y. Meng:
Sinusoidal modeling using frame-based perceptually weighted matching pursuits. 981-984 - Scott N. Levine, Julius O. Smith III:
A switched parametric and transform audio coder. 985-988 - Yin H. Lam, Robert W. Stewart:
Perception-based residual analysis-synthesis system. 989-992 - Todd D. Hodes, John R. Hauser, John Wawrzynek, Adrian Freed, David Wessel:
A fixed-point recursive digital oscillator for additive synthesis of audio. 993-996
Application - Pattern Recognition and Speech Processing
- Aurelio Uncini, Frencesco Gobbi, Francesco Piazza:
Frequency recovery of narrow-band speech using adaptive spline neural networks. 997-1000 - Klaus Reinhard, Mahesan Niranjan:
Diphone multi-trajectory subspace models. 1001-1004 - Daniel Rodriguez-Porcheron, Marcos Faúndez-Zanuy:
Speaker recognition with a MLP classifier and LPCC codebook. 1005-1008 - Holger Schwenk:
Using boosting to improve a hybrid HMM/neural network speech recognizer. 1009-1012 - Dan Ellis, Nelson Morgan:
Size matters: an empirical study of neural network training for large vocabulary continuous speech recognition. 1013-1016 - Thiagarajan Balachander, Ravi Kothari:
Oriented soft localized subspace classification. 1017-1020 - Mathini Sellathurai, Simon Haykin:
The separability theory of hyperbolic tangent kernels and support vector machines for pattern classification. 1021-1024 - Eisaku Maeda, Hiroshi Murase:
Multi-category classification by kernel based nonlinear subspace method. 1025-1028 - David J. Miller, Lian Yan:
Ensemble classification by critic-driven combining. 1029-1032 - Madiha Sabry-Rizk, Walid A. Zgallai, Sahar El-Khafif, Ewart R. Carson, Kenneth T. V. Grattan, Peter Thompson:
Highly accurate higher order statistics based neural network classifier of specific abnormality in electrocardiogram signals. 1033-1036
Theory & Neural Architecture
- Danilo P. Mandic, Jonathon A. Chambers:
Global asymptotic convergence of nonlinear relaxation equations realised through a recurrent perceptron. 1037-1040 - Michel Winter, Gérard Favier:
A neural network for data association. 1041-1044 - Dongxin Xu, José C. Príncipe:
Training MLPs layer-by-layer with the information potential. 1045-1048 - Lian Yan, David J. Miller:
Time series prediction via neural network inversion. 1049-1052 - Tülay Adali, Hongmei Ni, Bo Wang:
Partial likelihood for estimation of multi-class posterior probabilities. 1053-1056 - João F. G. de Freitas, Mahesan Niranjan, Andrew H. Gee:
Hybrid sequential Monte Carlo/Kalman methods to train neural networks in non-stationary environments. 1057-1060 - Takashi Matsumoto, Motoki Saito, Yoshinori Nakajima, Junjiro Sugi, Hiroaki Hamagishi:
Reconstruction and prediction of nonlinear dynamical systems: a hierarchical Bayes approach with neural nets. 1061-1064 - Mahesan Niranjan:
Sequential Bayesian computation of logistic regression models. 1065-1068
Signal Separation
- Yinchao Guo, Farook Sattar, Christopher K. H. Koh:
Blind separation of temporomandibular joint sound signals. 1069-1072 - Hsiao-Chun Wu, José C. Príncipe:
Generalized anti-Hebbian learning for source separation. 1073-1076 - Eraldo Pomponi, Simone G. O. Fiori, Francesco Piazza:
Complex independent component analysis by nonlinear generalized Hebbian learning with Rayleigh nonlinearity. 1077-1080 - Kenneth Kreutz-Delgado, Bhaskar D. Rao:
Sparse basis selection, ICA, and majorization: towards a unified perspective. 1081-1084 - Seungjin Choi, Andrzej Cichocki, Shun-ichi Amari:
Two spatio-temporal decorrelation learning algorithms and their application to multichannel blind deconvolution. 1085-1088 - Scott C. Douglas, Shun-ichi Amari, Sun-Yuan Kung:
Adaptive paraunitary filter banks for principal and minor subspace analysis. 1089-1092
Application - Image & Nonlinear Signal Processing
- Christoph Neukirchen, Daniel Willett, Gerhard Rigoll:
Experiments in topic indexing of broadcast news using neural networks. 1093-1096 - Avni H. Rambhia, Robb W. Glenny, Jenq-Neng Hwang:
Critical input data channels selection for progressive work exercise test by neural network sensitivity analysis. 1097-1100 - Songyang Yu, Ling Guan:
Feature selection using general regression neural networks for the automatic detection of clustered microcalcifications. 1101-1104 - Jyh-Charn Liu, Gouchol Pok:
Texture edge detection by feature encoding and predictive model. 1105-1108 - Hau-San Wong, Terry Caelli, Ling Guan:
Edge characterization using a model-based neural network. 1109-1112 - Stefano Rovetta, Rodolfo Zunino:
License-plate localization by using vector quantization. 1113-1116 - Marios Poulos, Maria Rangoussi, Nikolaos Alexandris:
Neural network based person identification using EEG features. 1117-1120 - Lars Nonboe Andersen, Whitlow W. L. Au, Jan Larsen, Lars Kai Hansen:
Sonar discrimination of cylinders from different angles using neural networks. 1121-1124 - Hsin-Chia Fu, Z. H. Chen, Yeong-Yuh Xu, C. H. Wang:
A neural network based transcoder for MPEG2 video compression. 1125-1128
Volume 3
Filter Design and Structures
- Ricardo A. Vargas, C. Sidney Burrus:
Adaptive iterative reweighted least squares design of Lp FIR filters. 1129-1132 - Johnny Holmberg, Lennart Harnefors, Svante Signell:
Quantization noise analysis of wave digital and lossless digital integrator allpass/lattice filters. 1133-1136 - Chris W. Schwarz, Soura Dasgupta:
A new normalized relatively stable lattice structure. 1137-1140 - Karl E. Nelson, Michael A. Soderstrand:
Full tunable digital heterodyne IIR filters. 1141-1144 - Niranjan Damera-Venkata, Brian L. Evans:
Optimal design of real and complex minimum phase digital FIR filters. 1145-1148 - Mathias C. Lang:
A multiple exchange algorithm for constrained design of FIR filters in the complex domain. 1149-1152 - Chien-Hsun Tseng, Zhuquan Zang, Kok Lay Teo, Antonio Cantoni:
Robust envelope-constrained filter design with Laguerre bases. 1153-1156 - Anthony G. Constantinides, Tania Stathaki:
A contribution to the stability test for one-dimensional discrete time linear systems. 1157-1160
Detection
- Nicholas B. Pulsone, Michael A. Zatman:
A computationally-efficient two-step implementation of the GLRT detector. 1161-1164 - George Mamic, Nathan Stitt, D. Robert Iskander:
Coherent detection of radar signals in G-distributed clutter. 1165-1168 - Chris W. Reed, Ralph E. Hudson, Kung Yao:
Direct joint source localization and propagation speed estimation. 1169-1172 - Douglas Cochran, Dana Sinno, Axel Clausen:
Source detection and localization using a multi-mode detector: a Bayesian approach. 1173-1176 - Axel Clausen, Douglas Cochran:
Asymptotic non-null distribution of the generalized coherence estimate. 1177-1180 - Jean Philippe Ovarlez, Emmanuelle Jay:
New methods of radar detection performances analysis. 1181-1184 - Thomas T. Liu, Antony C. Fraser-Smith:
An undecimated wavelet transform based detector for transients in 1/f noise. 1185-1188 - John L. Spiesberger:
Detecting multipath signals with the matched-lag filter. 1189-1192 - Marcelo G. S. Bruno, José M. F. Moura:
Performance of the optimal nonlinear detector/tracker in clutter. 1193-1196 - Hwa-Tung Ong, Abdelhak M. Zoubir:
Robust signal detection using the bootstrap. 1197-1200
Wavelets
- Wei Zhao, Raghuveer M. Rao:
A discrete-time wavelet transform based on a continuous dilation framework. 1201-1204 - Felix C. A. Fernandes, C. Sidney Burrus:
Multiwavelet systems with disjoint multiscaling functions. 1205-1208 - Ivan W. Selesnick:
Cardinal multiwavelets and the sampling theorem. 1209-1212 - Faramarz Fekri, Russell M. Mersereau, Ronald W. Schafer:
Theory of wavelet transform over finite fields. 1213-1216 - Carl Taswell:
Least and most disjoint root sets for Daubechies wavelets. 1217-1220 - Nick G. Kingsbury:
Shift invariant properties of the dual-tree complex wavelet transform. 1221-1224 - Lixin Shen, Jo Yew Tham, Seng Luan Lee, Hwee Huat Tan:
A special class of orthonormal wavelets: theory, implementations, and applications. 1225-1228 - Jo Yew Tham, Lixin Shen, Seng Luan Lee, Hwee Huat Tan:
A new multifilter design property for multiwavelet image compression. 1229-1232
Adaptive Filtering: Applications and Implementation
- Mark Dzwonczyk, Teresa H.-Y. Meng:
How good is your predictor? Expanding confidence intervals to define probability densities on adaptive parameters. 1233-1236 - Mounir Bhouri:
Fast QR based IIR adaptive filtering algorithm. 1237-1240 - Milos I. Doroslovacki, Branimir R. Vojcic:
Cone constrained adaptive algorithms and multiple access interference cancellation. 1241-1244 - Robby Gupta, Alfred O. Hero III:
Theoretical aspects of power reduction for adaptive filters. 1245-1248 - Khaled A. Mayyas, Tyseer Aboulnasr:
A fast weighted subband adaptive algorithm. 1249-1252 - Lan-Da Van, Shing Tenqchen, Chia-Hsun Chang, Wu-Shiung Feng:
A tree-systolic array of DLMS adaptive filter. 1253-1256 - Rajarshi Gupta, Kiran, Edward A. Lee:
Computationally efficient version of the decision feedback equalizer. 1257-1260 - Dinko Begusic, Darel A. Linebarger, Eric M. Dowling, Balaji Raghothaman:
Spectral line RLS adaptive filtering algorithm. 1261-1264 - Corneliu Rusu, Colin F. N. Cowan:
Recursive cost function adaptation for echo cancellation. 1265-1268 - Monia Turki-Hadj Alouane, Meriem Jaïdane-Saïdane:
A non-stationary RLS algorithm for adaptive tracking of Markov time varying channel. 1269-1272
Nonlinear Signals and Systems
- Ranveig Nygaard, John Håkon Husøy, Dag Haugland, Sven Ole Aase:
Signal compression by piecewise linear non-interpolating approximation. 1273-1276 - Jakob Ängeby, Mats Viberg, Tony Gustafsson:
Non-linear instantaneous least squares and its high SNR analysis. 1277-1280 - Kutluyil Dogançay, Vikram Krishnamurthy:
Level estimation in nonlinearly distorted hidden Markov models using statistical extremes. 1281-1284 - Naresh Sharma, Edward Ott:
Combating channel distortions for chaotic signals. 1285-1288 - Zhiwen Zhu, Henry Leung:
Adaptive identification of bilinear systems. 1289-1292 - Patrick Celka, Neil J. Bershad, Jean-Marc Vesin:
Analysis of stochastic gradient identification of polynomial nonlinear systems with memory. 1293-1296 - Hans-Peter Bernhard, Georges A. Darbellay:
Performance analysis of the mutual information function for nonlinear and linear signal processing. 1297-1300 - Mahmut Ciftci, Douglas B. Williams:
A novel channel equalizer for chaotic digital communications systems. 1301-1304 - Anders E. Nordsjö, Lars-Henning Zetterberg:
A recursive prediction error algorithm for identification of certain time-varying nonlinear systems. 1305-1308 - Byung-Jae Kwak, Andrew E. Yagle, Joel A. Levitt:
Nonlinear system identification of hydraulic actuator friction dynamics using a finite-state memory model. 1309-1312 - Jean Pierre Da Costa, Luc Pronzato, Eric Thierry:
Nonlinear filtering by kriging, with application to system inversion. 1313-1316
Time/Frequency and Time/Scale Analysis
- Braham Barkat, Boualem Boashash, Ljubisa Stankovic:
Adaptive window in the PWVD for the IF estimation of FM signals in additive Gaussian noise. 1317-1320 - Yiu Tong Chan, K. C. Ho:
Filter design for CWT computation using the Shensa algorithm. 1321-1324 - Arno J. van Leest, Martin J. Bastiaans:
Gabor's signal expansion on a quincunx lattice and the modified Zak transform. 1325-1328 - Xiang-Gen Xia, Leon Cohen:
On analytic signals with nonnegative instantaneous frequencies. 1329-1332 - Gerald Matz, Franz Hlawatsch:
Minimax robust time-frequency filters for nonstationary signal estimation. 1333-1336 - Yimin Zhang, Moeness G. Amin:
Spatial averaging of time-frequency distributions. 1337-1340 - Bradford W. Gillespie, Les E. Atlas:
Optimization of time and frequency resolution for radar transmitter identification. 1341-1344 - Byeong-Gwan Iem, Antonia Papandreou-Suppappola, Gloria Faye Boudreaux-Bartels:
New time-frequency symbol classification. 1345-1348 - Ramdas Kumaresan:
An inverse signal approach to computing the envelope of a real valued signal. 1349-1352 - Aykut Bultan, Ali N. Akansu:
Frames in rotated time-frequency planes. 1353-1356
Signal Modeling and Representation
- J. Scott Goldstein, Joseph R. Guerci, Irving S. Reed:
An optimal generalized theory of signal representation. 1357-1360 - Don H. Johnson, Wei Wang:
Symbolic signal processing. 1361-1364 - Tomasz Przebinda, Victor E. DeBrunner, Murad Özaydin:
Using a new uncertainty measure to determine optimal bases for signal representations. 1365-1368 - André Ferrari, Jean-Yves Tourneret, François-Xavier Schmider:
Detection of extra solar planets using parametric modeling. 1369-1372 - Roger A. Green:
A DSB-SC signal model for nonlinear regression-based quadrature receiver calibration. 1373-1376 - Abdelhak M. Zoubir:
Model selection: a bootstrap approach. 1377-1380 - Glen Andrews, Kie B. Eom:
Color texture synthesis with 2-D moving average model. 1381-1384 - Soonman Kwon, Daniel R. Fuhrmann:
Sampling theorems for linear time-varying systems with bandlimited inputs. 1385-1388
Filterbank and Wavelet Applications
- Douglas E. Driscoll, Stephen D. Howard:
The detection of radar pulse sequences by means of a continuous wavelet transform. 1389-1392 - Conor Heneghan, Steven B. Lowen, Malvin C. Teich:
Analysis of spectral and wavelet-based measures used to assess cardiac pathology. 1393-1396 - Yuan-Pei Lin, See-May Phoong:
Optimal DMT transceivers over fading channels. 1397-1400 - Matthew L. Welborn:
Narrowband channel extraction for wideband receivers. 1401-1404 - Subbarao S. Govardhanagiri, Tanja Karp, Peter N. Heller, Truong Q. Nguyen:
Performance analysis of multicarrier modulation systems using cosine modulated filter banks. 1405-1408 - See-May Phoong, Yuan-Pei Lin:
Optimal ladder-based biorthogonal coder. 1409-1412 - Jie Liang:
The predictive embedded zerotree wavelet (PEZW) coder: low complexity image coding with versatile functionality. 1413-1416 - Sang-il Park, Mark J. T. Smith, Russell M. Mersereau:
A new directional filter bank for image analysis and classification. 1417-1420 - Henrique S. Malvar:
A modulated complex lapped transform and its applications to audio processing. 1421-1424 - Panagiotis D. Hatziantoniou, Dionysis E. Tsoukalas, John Mourjopoulos, Soterios Salamouris:
Time-frequency mapping based on non-uniform smoothed spectral representations. 1425-1428
Source and Signal Separation
- Carine Simon, Philippe Loubaton, Christophe Vignat, Christian Jutten, Guy D'Urso:
Separation of a class of convolutive mixtures: a contrast function approach. 1429-1432 - T. Engin Tuncer:
A new time-domain deconvolution algorithm and its applications. 1433-1436 - James P. Reilly, Lino Coria-Mendoza:
Blind signal separation for convolutive mixing environments using spatial-temporal processing. 1437-1440 - Dragan Obradovic:
Dynamic signal mixtures and blind source separation. 1441-1444 - Anisse Taleb, Christian Jutten:
On underdetermined source separation. 1445-1448 - James R. Hopgood, Peter J. W. Rayner:
Single channel separation using linear time varying filters: separability of non-stationary stochastic signals. 1449-1452 - Vicente Zarzoso, Asoke K. Nandi:
Blind source separation without optimization criteria? 1453-1456 - Lahouari Ghouti, Chi Hau Chen:
Deconvolution of ultrasonic nondestructive evaluation signals using higher-order statistics. 1457-1460
Filterbanks
- Unto K. Laine:
Block-recursive, multirate filterbanks with arbitrary time-frequency plane tiling. 1461-1464 - Wolfgang Niehsen:
Boundary filters without DC leakage for paraunitary filter banks. 1465-1468 - Kunitoshi Komatsu, Kaoru Sezaki:
Lossless filter banks based on two point transform and interpolative prediction. 1469-1472 - Jacob D. Griesbach, Tamal Bose, Delores M. Etter:
Non-uniform filterbank bandwidth allocation for system modeling subband adaptive filters. 1473-1476 - Fabrizio Argenti, Enrico Del Re:
Eigenfilter design of real and complex coefficient prototypes for uniform and nonuniform filter banks. 1477-1480 - María Elena Domínguez Jiménez, Nuria González-Prelcic:
Processing finite length signals via filter banks without border distortions: a non-expansionist solution. 1481-1484 - Kenji Nakayama, Akihiro Hirano, Hiroaki Sakaguchi:
A polyphase and FFT realization of modulation sub-band adaptive filter with minimum sampling rate. 1485-1488 - Soura Dasgupta, Chris W. Schwarz, Brian D. O. Anderson:
Optimum subband coding of cyclostationary signals. 1489-1492 - Haris Vikalo, Babak Hassibi, Thomas Kailath:
On H∞ optimal signal reconstruction in noisy filter banks. 1493-1496 - Larry A. Wasserman, Alan N. Willson Jr.:
A variable-rate filtering system for digital communications. 1497-1500 - Sony Akkarakaran, P. P. Vaidyanathan:
New results and open problems on nonuniform filter-banks. 1501-1504
Emerging Applications and Fast Algorithms
- Mohammed A. Hasan, Jawad A. K. Hasan:
Rational signal subspace approximation with applications to DOA estimation. 1505-1508 - Scott D. Coutts:
3-D emitter localization using inhomogeneous bistatic scattering. 1509-1512 - Luowen Li, Lihua Xie, Gang Li, Yeng Chai Soh:
A new technique to filter reduction for speech signal processing systems. 1513-1516 - René Landry Jr., Vincent Calmettes, Michel Bousquet:
PIRANHA filter for communication system robustness. 1517-1520 - John Garas, Piet C. W. Sommen:
Warped linear time invariant systems and their application in audio signal processing. 1521-1524 - Ryszard M. Stasinski:
Efficiency of radix-K transforms on computers with cache. 1525-1528 - George Keratiotis, Larry Lind, John W. Cook, Minesh Patel, David Croft, Peter T. Whelan, Peter Hughes:
A novel method for power line interference suppression. 1529-1532 - Jaehak Chung, Edward J. Powers, W. Mack Grady, Sid C. Bhatt:
Adaptive power-line disturbance detection scheme using a prediction error filter and a stop-and-go CA CFAR detector. 1533-1536 - Hazem M. Abbas:
Time series analysis for ECG data compression. 1537-1540 - Wei Wang, M. N. S. Swamy, M. Omair Ahmad, Yuke Wang:
A parallel residue-to-binary converter. 1541-1544
Frequence and Phase Estimation
- Andrew E. Yagle:
Fast and recursive algorithms for magnitude retrieval from DTFT phase at irregular frequencies. 1545-1548 - Peter Händel, Anders Høst-Madsen:
On frequency estimation from oversampled quantized observations. 1549-1552 - Sandrine Vaton, Thierry Chonavel:
Estimating the offset parameters of a mixture in the Fourier domain. 1553-1556 - Mark R. Morelande:
Optimal phase parameter estimation of random amplitude linear FM signals using cyclic moments. 1557-1560 - Olivier Besson, Mounir Ghogho, Ananthram Swami:
On estimating random amplitude chirp signals. 1561-1564 - Martin Kristensson, Magnus Jansson, Björn E. Ottersten:
On subspace based sinusoidal frequency estimation. 1565-1568 - Dawei Huang, Simon Sando, Lian Wen:
Least squares estimation of polynomial phase signals via stochastic tree-search. 1569-1572 - Martial Coulon, Jean-Yves Tourneret:
Multiple frequency estimation in additive and multiplicative colored noises. 1573-1576 - Mounir Ghogho, Asoke K. Nandi, Ananthram Swami:
Cramer-Rao bounds and parameter estimation for random amplitude phase modulated signals. 1577-1580 - Jeffrey C. O'Neill, Patrick Flandrin:
Cramer-Rao lower bounds for atomic decomposition. 1581-1584 - Tong Zhou, Yongsub Kim:
Harmonic retrieval in nonstationary noise. 1585-1588
Spectral Analysis and Higher Order Statistics
- Stéphanie Rouquette, Olivier Alata, Mohamed Najim, Charles W. Therrien:
2-D high resolution spectral estimation based on multiple regions of support. 1589-1592 - Yan Zhang, Shuxun Wang:
Harmonic retrieval in colored non-Gaussian noise. 1593-1596 - Piet M. T. Broersen:
Accurate ARMA models with Durbin's second method. 1597-1600 - Philippe Ciuciu, Jérôme Idier, Jean-François Giovannelli:
Markovian high resolution spectral analysis. 1601-1604 - Tatiana Alieva, Martin J. Bastiaans, André Barbé:
Spectral analysis of discrete signals generated by multiplicative and additive iterative procedures. 1605-1608 - I. Vaughan L. Clarkson:
Frequency estimation, phase unwrapping and the nearest lattice point problem. 1609-1612 - Robin L. Murray, Antonia Papandreou-Suppappola, Gloria Faye Boudreaux-Bartels:
A new class of affine higher order time-frequency representations. 1613-1616 - Hakan Tora, D. Mitchell Wilkes:
Identification of noncausal nonminimum phase AR models using higher-order statistics. 1617-1620
Signal Reconstruction
- John Tuthill, Antonio Cantoni:
Automatic digital pre-compensation in IQ modulators. 1621-1624 - Wooshik Kim:
Two-dimensional phase retrieval using a window function. 1625-1628 - Xiao-Ping Zhang, Zhi-Quan Luo:
A new time-scale adaptive denoising method based on wavelet shrinkage. 1629-1632 - Constantine Papageorgiou, Federico Girosi, Tomaso A. Poggio:
Sparse correlation kernel reconstruction. 1633-1636 - Daniel Seidner, Meir Feder:
Optimal generalized sampling expansion. 1637-1640 - Stephen D. Casey, Brian M. Sadler:
Sampling on unions of non-commensurate lattices via complex interpolation theory. 1641-1644 - Hyeokho Choi, Richard G. Baraniuk:
Interpolation and denoising of nonuniformly sampled data using wavelet-domain processing. 1645-1648 - Shahrnaz Azizi, Douglas Cochran:
Reproducing kernel structure and sampling on time-warped Kramer spaces. 1649-1652 - Victor Solo:
Selection of regularisation parameters for total variation denoising. 1653-1655 - Pina Marziliano, Martin Vetterli:
Irregular sampling with unknown locations. 1657-1660
Adaptive Filter Analysis
- Márcio H. Costa, José C. M. Bermudez, Neil J. Bershad:
Statistical analysis of the LMS algorithm with a zero-memory nonlinearity after the adaptive filter. 1661-1664 - Kazushi Ikeda, Hideaki Sakai:
Convergence properties of the block orthogonal projection algorithm. 1665-1668 - Hichem Besbes, Yousra Ben Jemaa, Meriem Jaïdane:
Exact convergence analysis of affine projection algorithm: the finite alphabet inputs case. 1669-1672 - Roberto López-Valcarce, Soura Dasgupta:
On the stability of the inverse time-varying prediction error filter obtained with the RWLS algorithm. 1673-1676 - Mahesh Godavarti, Alfred O. Hero III:
Stability bounds on step-size for the partial update LMS algorithm. 1677-1680 - Kaywan H. Afkhamie, Zhi-Quan Luo:
Adaptive parameter estimation using interior point optimization techniques: convergence analysis. 1681-1684 - Robert A. Soni, Kyle A. Gallivan, W. Kenneth Jenkins:
Affine projection methods in fault tolerant adaptive filtering. 1685-1688 - Shue-Lee Chang, Tokunbo Ogunfunmi:
Performance analysis of third-order nonlinear Wiener adaptive systems. 1689-1692
Transforms and Statistical Estimation
- Jari P. Kaipio, Marko Juntunen:
Deterministic regression smoothness priors TVAR modelling. 1693-1696 - Irina F. Gorodnitsky:
An extension of an interior-point method for entropy minimization. 1697-1700 - Michael K. Schneider, Alan S. Willsky:
A Krylov subspace method for large estimation problems. 1701-1704 - Jean-François Bercher, Christophe Vignat:
Estimating the entropy of a signal with applications. 1705-1708 - Der-Feng Huang, Bor-Sen Chen:
The filter bank approach for the fractional Fourier transform. 1709-1712 - Cagatay Candan, M. Alper Kutay, Haldun M. Özaktas:
The discrete fractional Fourier transform. 1713-1716 - Thuyen Le, Manfred Glesner:
A data-driven scheme for the approximated computing of alias-free generalized discrete time-frequency distributions. 1717-1720 - Dan P. Scholnik, Jeffrey O. Coleman:
Periodically nonuniform bandpass sampling as a tapped-delay-line filtering problem. 1721-1724 - Alban Duverdier, Bernard Lacaze:
New realization method for linear periodic time-varying filters. 1725-1728 - Patrice Abry, Lieve Delbeke, Patrick Flandrin:
Wavelet based estimator for the self-similarity parameter of α-stable processes. 1729-1732
Markov and Bayesian Estimation and Classification
- Simon J. Godsill, Christophe Andrieu:
Bayesian separation and recovery of convolutively mixed autoregressive sources. 1733-1736 - Petar M. Djuric, Joon-Hwa Chun:
Estimation of nonstationary hidden Markov models by MCMC sampling. 1737-1740 - Robert D. Nowak, Eric D. Kolaczyk:
A Bayesian multiscale framework for Poisson inverse problems. 1741-1744 - Rangasami L. Kashyap, Srinivas Sista:
Bayesian framework for unsupervised classification with application to target tracking. 1745-1748 - Matthew Brand:
Structure and parameter learning via entropy minimization, with applications to mixture and hidden Markov models. 1749-1752 - Christian P. Robert, Arnaud Doucet, Simon J. Godsill:
Marginal MAP estimation using Markov chain Monte Carlo. 1753-1756 - Jayesh H. Kotecha, Petar M. Djuric:
Gibbs sampling approach for generation of truncated multivariate Gaussian random variables. 1757-1760 - Joseph Tabrikian, Jeffrey L. Krolik:
Efficient computation of the Bayesian Cramer-Rao bound on estimating parameters of Markov models. 1761-1764
Volume 4
System Identification, Equalization, and Noise Suppression
- Yuexian Zou, S. C. Chan, Tung-Sang Ng:
A robust M-estimate adaptive filter for impulse noise suppression. 1765-1768 - Hakan Öktem, Karen O. Egiazarian, Juha Nousiainen:
Local adaptive de-noising techniques in transform domain for EMCG de-noising. 1769-1772 - Hamid Krim, Yufang Bao:
A stochastic diffusion approach to signal denoising. 1773-1776 - Petre Stoica, Hongbin Li, Jian Li:
Amplitude estimation with application to system identification. 1777-1780 - Channarong Tontiruttananon, Jitendra K. Tugnait:
Performance analysis of two approaches to closed loop system identification via cyclic spectral analysis. 1781-1784 - Takashi Kimura, Hideaki Sasaki, Hiroshi Ochi:
Blind channel identification using RLS method based on second-order statistics. 1785-1788 - Gopal T. Venkatesan, Lang Tong, Mostafa Kaveh, Ahmed H. Tewfik, Kevin M. Buckley:
A deterministic blind identification technique for SIMO systems of unknown model order. 1789-1792 - Rafael Ruiz, Margarita Cabrera:
Blind channel equalization using weighted subspace methods. 1793-1796 - Carlos Avendaño, Jacob Benesty, Dennis R. Morgan:
A least squares component normalization approach to blind channel identification. 1797-1800 - Konstantinos I. Diamantaras, Athina P. Petropulu:
Blind equalization of multiuser CDMA channels: a frequency-domain approach. 1801-1804
Parameter Estimation
- Gabriella Olmo, Letizia Lo Presti, Paolo Severico:
An enhanced TEA algorithm for modal analysis. 1805-1808 - Jean-Jacques Fuchs:
An inverse problem approach to robust regression. 1809-1812 - Arie Yeredor, Ehud Weinstein:
The extended least-squares and the joint maximum-a-posteriori maximum-likelihood estimation criteria. 1813-1816 - Jaume Riba, Gregori Vázquez:
Conditional maximum likelihood timing recovery. 1817-1820 - Paulo M. Oliveira, Victor A. N. Barroso:
Sequential extraction of components of multicomponent PPS signals. 1821-1824 - Paul M. Baggenstoss, Tod Luginbuhl:
An E-M algorithm for joint model estimation. 1825-1828 - Arye Nehorai, Malcolm Hawkes:
Performance measures for estimating vector systems. 1829-1832 - Céline Theys, Michelle Vieira, Gérard Alengrin:
A reversible jump sampler for polynomial-phase signals. 1833-1836
Adaptive Filters: Algorithms and Performance
- Sundar G. Sankaran, A. A. (Louis) Beex:
Balanced-realization based adaptive IIR filtering. 1837-1840 - Carlos Mosquera, Fernando Pérez-González, Roberto López-Valcarce:
Hyperstable polyphase adaptive IIR filters. 1841-1844 - Shin'ichi Koike:
A novel adaptive step size control algorithm for adaptive filters. 1845-1848 - Cheng-Shing Wu, An-Yeu Wu:
A novel multirate adaptive FIR filtering algorithm and structure. 1849-1852 - Monther I. Haddad, Khaled A. Mayyas, Mohammed A. Khasawneh:
Analytical development of the MMAXNLMS algorithm. 1853-1856 - Marcello L. R. de Campos, Stefan Werner, J. A. Apolinário Jr.:
Householder-transform constrained LMS algorithms with reduced-rank updating. 1857-1860 - Mariane R. Petraglia, Rogerio Guedes Alves:
A new adaptive subband structure with critical sampling. 1861-1864 - Junghsi Lee, Sheng-Chieh Chong:
On the convergence properties of multidelay frequency domain adaptive filter. 1865-1868 - Balaji Raghothaman, Darel A. Linebarger, Dinko Begusic:
Analysis of low rank transform domain adaptive filtering algorithm. 1869-1872 - Orlando José Tobias, José C. M. Bermudez, Neil J. Bershad, Rui Seara:
Second moment analysis of the filtered-X LMS algorithm. 1873-1876
DSP Development Tools
- Magnus Lundberg, Khurram Muhammad, Kaushik Roy, Sarah Kate Wilson:
High-level modeling of switching activity with application to low-power DSP system synthesis. 1877-1880 - Russell E. Henning, Chaitali Chakrabarti:
Activity models for use in low power, high-level synthesis. 1881-1884 - Bruce W. Suter, Kenneth S. Stevens, Scott R. Velazquez, Truong Q. Nguyen:
Multirate as a hardware paradigm. 1885-1888 - Sissades Tongsima, Timothy W. O'Neil, Edwin Hsing-Mean Sha:
Unfolding probabilistic data-flow graphs under different timing models. 1889-1892 - Manish Goel, Naresh R. Shanbhag:
Low-power channel coding via dynamic reconfiguration. 1893-1896 - Paul D. Fiore, Li Lee:
Closed-form and real-time wordlength adaptation. 1897-1900 - Darko Kirovski, Miodrag Potkonjak:
Synthesis of DSP soft real-time multiprocessor systems-on-silicon. 1901-1904 - Frantz Lohier, Lionel Lacassagne, Patrick Garda:
A generic methodology for the software managing of caches in multi-processors DSP architectures. 1905-1908
VLSI Building Blocks
- Stephen McInerney, Richard B. Reilly:
Hybrid multiplier/CORDIC unit for online handwriting recognition. 1909-1912 - Hiroshi Suzuki, Yun-Nan Chang, Keshab K. Parhi:
Low-power bit-serial Viterbi decoder for next generation wide-band CDMA systems. 1913-1916 - Wei-Lung Liu, Oscal T.-C. Chen:
A highly-scaleable symmetric/asymmetric FIR processor. 1917-1920 - Ching-Hsien Chang, Chin-Liang Wang, Yu-Tai Chang:
A novel memory-based FFT processor for DMT/OFDM applications. 1921-1924 - John Bonk, Andrew Stone, Elias S. Manolakos:
Synthesis of array architectures for block matching motion estimation: design exploration using the tool DG2VHDL. 1925-1928 - Shen-Fu Hsiao, Wei-Ren Shiue:
A high-throughput, low power architecture and its VLSI implementation for DFT/IDFT computation. 1929-1932 - Gaye Lightbody, Richard L. Walke, Roger F. Woods, John V. McCanny:
Novel mapping of a linear QR architecture. 1933-1936 - Robert A. Freking, Keshab K. Parhi:
An unrestrictedly parallel scheme for ultra-high-rate reprogrammable Huffman coding. 1937-1940 - Stefano Rovetta, Rodolfo Zunino:
Flexible video compression systems using an analog vector quantization chip. 1941-1944
DSP Architectures
- Matthias Weiss, Frank Engel, Gerhard P. Fettweis:
A new scalable DSP architecture for system on chip (SoC) domains. 1945-1948 - Thuyen Le, Manfred Glesner:
A new flexible architecture for variable length DCT targeting shape-adaptive transform. 1949-1952 - Vasily G. Moshnyaga:
An adaptive block-matching algorithm for motion estimation. 1953-1956 - Jarmo Takala, Jouko O. Viitanen, Jukka Saarinen:
Hardware architecture for real-time distance transform. 1957-1960 - Bai-Jue Shieh, Chen-Yi Lee:
An efficient VLC decompression scheme for user-defined coding tables. 1961-1964 - Loris Navoni, Monica Besana, Pier Luigi Rolandi:
An analog associative memory chip for VQ image compression. 1965-1968 - Mario Novell, Steve Molloy:
VLSI implementation of a reversible variable length encoder/decoder. 1969-1972 - Stephen J. Bellis, William P. Marnane, Peter J. Fish:
Parametric spectral estimation on a single FPGA. 1973-1976
DSP System Design
- Jeff Y. F. Hsieh, Teresa H. Meng:
Low-power DV encoder architecture for digital CMOS camcorder. 1977-1980 - Tetsuro Takizawa, Junji Tajime, Hidenobu Harasaki:
High performance and cost effective memory architecture for an HDTV decoder LSI. 1981-1984 - Takafumi Morifuji, Yoshinori Takeuchi, Masaharu Imai:
A programmable processor with multiple functional units and banked registers for general purpose numerical processing. 1985-1988 - Shai Rubin, Moshe Levinger, Randall R. Pratt, William P. Moore:
Fast construction of test-program generators for digital signal processors. 1989-1992 - Darko Kirovski, Miodrag Potkonjak:
Engineering change protocols for behavioral synthesis. 1993-1996 - Thomas Zeitlhofer, Bernhard Wess:
Operation scheduling for parallel functional units using genetic algorithms. 1997-2000 - Timothy W. O'Neil, Sissades Tongsima, Edwin Hsing-Mean Sha:
Extended retiming: optimal scheduling via a graph-theoretical approach. 2001-2004 - Hung-Ying Tyan, Yu Hen Hu:
Minimum initiation interval of multi-module recurrent signal processing algorithm realization with fixed communication delay. 2005-2008 - Shiro Kobayashi, Gerhard P. Fettweis:
A new approach for block-floating-point arithmetic. 2009-2012 - Louis R. Litwin Jr., Thomas J. Endres, Samir N. Hulyalkar, Michael D. Zoltowski:
The effects of finite bit precision for a VLSI implementation of the constant modulus algorithm. 2013-2016
Education
- Delores M. Etter, Geoffrey C. Orsak:
Reflections on a distance education experiment in DSP. 2017-2020 - Thomas P. Barnwell III, James H. McClellan, Russell M. Mersereau, Ronald W. Schafer:
Repackaging signals, systems and circuits in the core ECE curriculum. 2021-2024 - David C. Munson Jr., Douglas L. Jones:
Analog signal processing first [educational course]. 2025-2028 - Saad Lamouri, Yusuf Öztürk, Hüseyin Abut:
A new collaborative active learning tool for signal processing education. 2029-2032 - Virginia L. Stonick, Wojtek Kolodziej, Otto Gygax:
Design of a guided-asynchronous graduate course in multimedia signal processing. 2033-2036 - C. Robert Hewes, Periagaram K. Rajasekaran:
DSPS education: an industry leader's experiences and expectations. 2037-2038
Recent Advances in Sampling Theory and Applications
- John J. Benedetto, Hui-Chuan Wu:
A multidimensional irregular sampling algorithm and applications. 2039-2042 - Paulo Jorge S. G. Ferreira:
The condition number of certain matrices and applications. 2043-2046 - Thomas Strohmer:
On the estimation of the bandwidth of nonuniformly sampled signals. 2047-2050 - David Francis Walnut:
Nonperiodic sampling and reconstruction from averages. 2051-2054 - William J. Fitzgerald:
The restoration of missing data using Bayesian numerical methods. 2055 - Gilbert G. Walter:
Non-uniform sampling in wavelet subspaces. 2057-2059
Steganography: Information Embedding, Digital Watermarking, and Data Hiding
- Brian Chen, Gregory W. Wornell:
An information-theoretic approach to the design of robust digital watermarking systems. 2061-2064 - Min Wu, Matthew L. Miller, Jeffrey A. Bloom, Ingemar J. Cox:
A rotation, scale and translation resilient public watermark. 2065 - Fred Mintzer, Gordon W. Braudaway:
If one watermark is good, are more better? 2067-2069 - Jean-Paul M. G. Linnartz, Ton Kalker, Jaap Haitsma:
Detecting electronic watermarks in digital video. 2071-2074 - Ahmed H. Tewfik, Mitchell D. Swanson, Bin B. Zhu:
Data embedding in audio: where do we stand. 2075 - Paul Jessop:
The business case for audio watermarking. 2077-2078
Speech Under Stress
- Herman J. M. Steeneken, John H. L. Hansen:
Speech under stress conditions: overview of the effect on speech production and on system performance. 2079-2082 - Jean-Claude Junqua, Steven Fincke, Kenneth L. Field:
The Lombard effect: a reflex to better communicate with others in noise. 2083-2086 - Guojun Zhou, John H. L. Hansen, James F. Kaiser:
Methods for stress classification: nonlinear TEO and linear speech based features. 2087-2090 - Raymond E. Slyh, W. Todd Nelson, Eric G. Hansen:
Analysis of mrate, shimmer, jitter, and F0 contour features across stress and speaking style in the SUSAS database. 2091-2094 - Allan J. South:
Some characteristics of speech produced under high G-force and pressure breathing. 2095-2098
Physics-Based Signal Processing
- David C. Munson Jr., Orhan Arikan:
Interpolation and the chirp transform: DSP meets optics. 2099-2102 - Michael Papazoglou, Jeffrey L. Krolik:
Estimation of aircraft altitude and altitude rate with over-the-horizon radar. 2103-2106 - Michael B. Porter, Philippe Roux, Hee-Chun Song, William A. Kuperman:
Tumor treatment by time-reversal acoustics. 2107-2110 - Peter Gerstoft, Donald F. Gingras:
Source and environmental parameter estimation using electromagnetic matched field processing. 2111-2114 - Paul Runkle, Lawrence Carin:
Multi-aspect target identification with wave-based matching pursuits and continuous hidden Markov models. 2115-2118
DSP Chips, Architectures and Implementations
- Uwe Meyer-Baese, Julien Buros, Wolfgang Trautmann, Fred J. Taylor:
Fast implementation of orthogonal wavelet filterbanks using field-programmable logic. 2119-2122 - Chris Dick, Fred Harris:
High-performance FPGA filters using sigma-delta modulation encoding. 2123-2126 - Dongho Kim, Gwangwoo Choe:
AMD's 3DNow!TM vectorization for signal processing applications. 2127-2130 - Kouhei Nadehara, Takashi Miyazaki, Ichiro Kuroda:
Radix-4 FFT implementation using SIMD multimedia instructions. 2131-2134 - Sanjay M. Joshi, Pradeep K. Dubey:
Some fast speech processing algorithms using AltiVec technology. 2135-2138 - José Fridman, William C. Anderson:
A new parallel DSP with short-vector memory architecture. 2139-2142 - Justin G. R. Delva, Ali M. Reza, Robert D. Turney:
FPGA implementation of a nonlinear two dimensional fuzzy filter. 2143-2146
DSP Tools and Rapic Prototyping
- Hao Shi, Roger Arnold, Karl Westerholz:
C/C++ compiler support for Siemens TriCore DSP instruction set. 2147-2150 - Edwin A. Suominen:
A simple, non-invasive probe for reconstructing signals inside a DSP. 2151-2154 - Bogong Su, Jian Wang, Andrew Esguerra:
Source-level loop optimization for DSP code generation. 2155-2158 - Alan C. Moorman, Donald M. Cates Jr.:
A complete development environment for image processing applications on adaptive computing systems. 2159-2162 - Ki-Il Kum, Jiyang Kang, Wonyong Sung:
A floating-point to integer C converter with shift reduction for fixed-point digital signal processors. 2163-2166 - Shahid Masud, John V. McCanny:
Rapid design of discrete orthonormal wavelet transforms using silicon IP components. 2167-2170 - Timothy Bigg, John Owen, Robert W. Stewart, Daniel Garcia-Alis, Moritz Harteneck, Marc Llovet-Vila:
Rapid prototyping library for adaptive signal processing applications. 2171-2174 - Mohamed Ben-Romdhane, Marius S. Vassiliou, Lan-Rong Dung:
Rapid prototyping of multimedia chip sets. 2175-2178 - Hiren C. Bhagatwala, Edward M. Painter, Andreas S. Spanias:
An interactive tool for bit error rate analysis of speech coding algorithms. 2179-2182
Communication Technologies
- Oguz Tanrikulu:
Design and implementation of an efficient point slicing algorithm for the V.34 modem standard. 2183-2186 - Gianmarco Panza, Silvio Cucchi, Daniele Meli:
Buffer control technique for transmission frequency recovery of CBR connections over ATM networks. 2187-2190 - William Jacklin, Stuart Collar, Scott Stratmoen, Brian Fitzpatrick, Jeffrey Stone:
A personal and inter-vehicle cordless communications system. 2191-2194 - Ahmed Mehaoua, Raouf Boutaba:
The impacts of errors and delays on the performance of MPEG2 video communications. 2195-2198 - Ehsan Daeipour:
Clock compensation in a data/fax relay system. 2199-2202 - Hemanth Sampath, Arogyaswami Paulraj:
Space-time processing TDMA wireless testbed. 2203-2206 - Junchen Du, George Warner, Erik J. Vallow, Penny E. Breyer, Tom L. Hollenbach:
GSM EFR implementation for TRAU application on DSP16000. 2207-2210 - Michael Sablatash, John H. Lodge:
Design of a synchronization scheme for a bandwidth-on-demand multiplexer-demultiplexer pair based on wavelet packet tree filter banks. 2211-2214
Image and Video Technologies
- Michael S. Andrews:
Architectures for generalized 2D FIR filtering using separable filter structures. 2215-2218 - Sangeeta Narang, Naresh K. Narang, Kanad K. Biswas:
Segmentation of non-rigid bodies in affine motion: a new framework. 2219-2222 - Madhukar Budagavi, Wendi Rabiner, Jennifer Webb, Raj Talluri:
Wireless MPEG-4 video on Texas Instruments DSP chips. 2223-2226 - Irene Koo, Panos Nasiopoulos, Rabab K. Ward:
Joint MPEG-2 coding for multi-program broadcasting of pre-recorded video. 2227-2230 - Robert D. Turney, Ali M. Reza, Justin G. R. Delva:
FPGA implementation of adaptive temporal Kalman filter for real time video filtering. 2231-2234 - Klaus Illgner, Hans-Georg Gruber, Pedro R. Gelabert, Jie Liang, Youngjun Yoo, Wissam Rabadi, Raj Talluri:
Programmable DSP platform for digital still cameras. 2235-2238 - Yap-Peng Tan, Tinku Acharya:
A robust sequential approach for the detection of defective pixels in an image sensor. 2239-2242 - Glen P. Abousleman:
Adaptive coding of hyperspectral imagery. 2243-2246 - Lowell L. Winger, Anastasios N. Venetsanopoulos:
Temporally scalable motion compensated adaptive temporal subband coding of video. 2247-2250
Automotive Applications / Industrial Signal Processing
- Jörg Velten, Anton Kummert, Dirk Maiwald:
Application of a brightness-adapted edge detector for real-time railroad tie detection in video images. 2251-2254 - Sönke Carstens-Behrens, Michael Wagner, Johann F. Böhme:
Improved knock detection by time variant filtered structure-borne sound. 2255-2258 - Yeshwant K. Muthusamy, Rajeev Agarwal, Yifan Gong, Vishu Viswanathan:
Speech-enabled information retrieval in the automobile environment. 2259-2262 - Abdurrahman Ünsal, Annette R. von Jouanne, Virginia L. Stonick:
A DSP active filter for power conditioning. 2263-2266 - Scott Fornero, Nasser Kehtarnavaz, M. Swaminadham, Don A. Phillips:
Fourier and wavelet transform features for whirl tower diagnostics. 2267-2270 - Ramón Miralles, Juan Morales, Luis Vergara:
An industrial application of signal processing: ceramic microcrack detection. 2271-2274 - Jonathon C. Ralston, David W. Hainsworth:
Application of ground penetrating radar for coal depth measurement. 2275-2278 - Alf Green, Kostas Tsakalis, Ward MacArthur, Sachi Dash:
Control-oriented identification and uncertainty estimation for paper machines. 2279-2282
Speech and Audio Technologies
- Jason Kridner, Mark T. Nadeski, Pedro R. Gelabert:
A DSP powered solid state audio system. 2283-2286 - Hyen-O Oh, Sung-Youn Kim, Dae Hee Youn, Il-Whan Cha:
New implementation techniques of a real-time MPEG-2 audio encoding system. 2287-2290 - Sassan Ahmadi:
An improved residual-domain phase/amplitude model for sinusoidal coding of speech at very low bit rates: a variable rate scheme. 2291-2294 - Ali Erdem Ertan, Emre B. Aksu, Hakki Gökhan Ilk, Mehmet Haydar Karcí, Onder Karpat, Taner Kolcak, Levent Sendur, Mübeccel Demirekler, Ahmet Enis Çetin:
Implementation of an enhanced fixed point variable bit-rate MELP vocoder on TMS320C549. 2295-2298 - Fenghua Liu, Ryan Heidari:
Improving EVRC half rate by the algebraic VQ-CELP. 2299-2302 - Hong Kook Kim, Mi Suk Lee, Hwang Soo Lee:
A 4 kbps adaptive fixed code-excited linear prediction speech coder. 2303-2306 - Amitava Das, Andrew DeJaco, Sharath Manjunath, Ananth Ananthapadmanabhan, Jeff Huang, Eddie Choy:
Multimode variable bit rate speech coding: an efficient paradigm for high-quality low-rate representation of speech signal. 2307-2310 - Costas S. Xydeas, Thomas M. Chapman:
Segmental prototype interpolation coding. 2311-2314
Defense and Security Applications
- Mikio Ikeda, Kazuya Takeda, Fumitada Itakura:
Audio data hiding by use of band-limited random sequences. 2315-2318 - Michael A. Koets, Randolph L. Moses:
Image domain feature extraction from synthetic aperture imagery. 2319-2322 - Stephen M. Kogon, Daniel J. Rabideau, Richard M. Barnes:
Clutter mitigation techniques for space-based radar. 2323-2326 - Ping Gao, Leslie M. Collins, Norbert Geng, Lawrence Carin, Dean A. Keiswetter, I. J. Won:
Classification of landmine-like metal targets using wideband electromagnetic induction. 2327-2330
Biomedical Applications
- Marios S. Pattichis, Constantinos S. Pattichis, Maria Avraam, Alan C. Bovik, Kyriakos Kyriakou:
AM-FM texture segmentation in electron microscopic muscle imaging. 2331-2334 - Amit Kam, Arnon Cohen:
Detection of fetal ECG with IIR adaptive filtering and genetic algorithms. 2335-2338 - Trudy Stetzler, Neeraj Magotra, Pedro R. Gelabert, Preethi Kasthuri, Sridevi Bangalore:
Low power real-time programmable DSP development platform for digital hearing aids. 2339-2342 - Rosana Esteller, George J. Vachtsevanos, Javier R. Echauz, Tom Henry, P. Pennell, C. Epstein, R. Bakay, Christina Bowen, Brian Litt:
Fractal dimension characterizes seizure onset in epileptic patients. 2343-2346 - Kenneth M. Houston, Robert E. Hillman, James B. Kobler, Geoffrey S. Meltzner:
Development of sound source components for a new electrolarynx speech prosthesis. 2347-2350 - Boualem Boashash, Paul Barklem, Mark Keir:
Detection of seizure signals in newborns. 2351-2354 - Ioanna Christoyianni, Evangelos Dermatas, George Kokkinakis:
Fast detection of masses in digitized mammograms. 2355-2358
Voice and Media Processing
- Yuzo Senda, Hidenobu Harasaki:
A realtime software MPEG transcoder using a novel motion vector reuse and a SIMD optimization techniques. 2359-2362 - Debashis Chowdhury, Ser J. Chia:
Distributed processing in the home using a PC with a wireless speech interface. 2363-2366 - Zhemin Tu, Philipos C. Loizou:
Speech recognition over the Internet using Java. 2367-2370 - James C. Abel, Michael A. Julier:
Implementation of a high-quality Dolby Digital decoder using MMX TM technology. 2371-2374 - Venceslav Kafedziski:
Joint source channel coding of images over frequency selective channels using DCT and multicarrier BPAM. 2375-2378 - John E. Kleider, Richard J. Pattison:
Multi-rate speech coding for wireless and Internet applications. 2379-2382
Adaptive Interference Cancellation
- Christopher Brunner, Martin Haardt, Josef A. Nossek:
Adaptive space-frequency RAKE receivers for WCDMA. 2383-2386 - Masaichi Akiho, Miki Haseyama, Hideo Kitajima:
A practical method to reduce a number of reference signals for the ANC system. 2387-2390 - Jamil Chaoui, Sébastien de Gregorio, Guillaume Gallissian, Yves Masse:
DSP-based solution for ambient noise reduction in mobile phones. 2391-2394 - Robert Simon Sherratt, David Townsend, Chris G. Guy:
Cancellation of siren noise from two way voice communications inside emergency vehicles. 2395-2398 - Idil Haskan, Aysin Ertüzün:
System identification using orthogonal functions and application to acoustic echo cancellation. 2399-2402 - Thorsten Ansahl, Imre Varga, Ingrid Kremmer, Wen Xu:
Adaptive acoustic echo cancellation based on FIR and IIR filter banks. 2403-2406 - Duanpei Wu, Miyuki Tanaka, Ruxin Chen, Lex Olorenshaw, Mariscela Amador, Xavier Menéndez-Pidal:
A robust speech detection algorithm for speech activated hands-free applications. 2407-2410
Volume 5
Source Coding and Compression
- Paolo Prandoni, Martin Vetterli:
Optimal bit allocation with side information. 2411-2414 - Amy R. Reibman, Hamid Jafarkhani, Michael T. Orchard, Yao Wang:
Performance of multiple description coders on a real channel. 2415-2418 - Wenqing Jiang, Antonio Ortega:
Multiple description coding via scaling-rotation transform. 2419-2422 - Deepen Sinha, Carl-Erik W. Sundberg:
Unequal error protection methods for perceptual audio coders. 2423-2426 - Alexis Bernard, Xueting Liu, Richard D. Wesel, Abeer Alwan:
Embedded joint source-channel coding of speech using symbol puncturing of trellis codes. 2427-2430 - Marcia G. Ramos, Ricardo L. de Queiroz:
Adaptive rate-distortion-based thresholding: application in JPEG compression of mixed images for printing. 2431-2434 - Hanying Feng, Michelle Effros:
Separable Karhunen Loeve transforms for the weighted universal transform coding algorithm. 2435-2438 - Dimitris Kalogiros, Vassilis Stylianakis:
A theoretical model for time code modulation. 2439-2442
Compression and Modulation
- Kjersti Engan, Sven Ole Aase, John Håkon Husøy:
Method of optimal directions for frame design. 2443-2446 - Wu-Hsiang Jonas Chen, Jenq-Neng Hwang:
Performance of ordered statistics decoding for robust video transmission on the WSSUS channel. 2447-2450 - MoonSeo Park, David J. Miller:
Joint source-channel decoding for variable-length encoded data by exact and approximate MAP sequence estimation. 2451-2454 - Fabrice Labeau, Luc Vandendorpe, Benoît Macq:
Gaussian modeling for channel errors diagnosis in image transmission. 2455-2458 - Farokh Marvasti, M. Hung, Mohammad Reza Nakhai:
The application of Walsh transform for forward error correction. 2459-2462 - Zoran Cvetkovic:
Modulating waveforms for OFDM. 2463-2466 - Hongya Ge, Kun Wang:
Efficient method for carrier offset correction in OFDM system. 2467-2470 - Helmut Ketterer, Friedrich K. Jondral, Antonio H. Costa:
Classification of modulation modes using time-frequency methods. 2471-2474 - Keith S. M. Lee, Michael J. Rowe, Vikram Krishnamurthy:
Pulse train deinterleaving: algorithms and cost criteria. 2475-2478
Channel Estimation and Equalization
- Deva K. Borah, Rodney A. Kennedy, Inbar Fijalkow:
Effects of colored noise on the performance of linear equalizers. 2479-2482 - João Gomes, Victor Barroso:
Performance analysis of a recursive fractional super-exponential algorithm. 2483-2486 - Arnaud Doucet, Christophe Andrieu:
Iterative algorithms for optimal state estimation of jump Markov linear systems. 2487-2490 - Jukka Mannerkoski, Visa Koivunen, Desmond P. Taylor:
Prediction-based adaptive blind equalization: a performance study. 2491-2494 - Tim Clapp, Simon J. Godsill:
Fixed-lag blind equalization and sequence estimation in digital communications systems using sequential importance sampling. 2495-2498