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ORCIDICASSP 1997: Munich, Germany
- 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '97, Munich, Germany, April 21-24, 1997. IEEE Computer Society 1997, ISBN 0-8186-7919-0
Volume 1: Plenary, Expert Summaries, Special, Audio, Underwater Acoustics, VLSI
Plenary Lectures
- Arogyaswami Paulraj:
Space-time processing for wireless communications. 1-4 - Don E. Pearson:
Variability of performance in video coding. 5-8
Expert Summaries
- Rama Chellapa, Renato De Mori, Georgios B. Giannakis, Hans Georg Musmann, Hermann Ney, Mark J. T. Smith, John R. Treichler, Michael D. Zoltowski:
Expert Summaries. 9-10
Signal Processing for Education
- Delores M. Etter, Geoffrey C. Orsak:
Expanding team experiences in DSP education. 11-14 - Hüseyin Abut, Yusuf Öztürk:
Interactive classroom for DSP/communication courses. 15-18 - James H. McClellan, Ronald W. Schafer, Mark A. Yoder:
Experiences in teaching DSP first in the ECE curriculum. 19-22 - David C. Munson Jr.:
Analog signal processing: a replacement for the sophomore-level circuit analysis course. 23-26 - Sanjit K. Mitra:
Re-engineering the electrical engineering curriculum. 27-30
New Methods for Design of SP-Algorithms
- Sanjit K. Mitra:
Structural subband decomposition: a new concept in digital signal processing. 31-34 - Knut Hüper, Uwe Helmke:
A new algorithm for the generalized eigenvalue problem. 35-38 - Soura Dasgupta, Chris W. Schwarz, Minyue Fu:
A lattice structure for perfect reconstruction linear time varying filter banks with all pass analysis banks. 39-42 - Steffen Paul, Josef A. Nossek:
Algorithm design for structured systems: application to pole placement. 43-46 - Klaus Diepold, Rainer Pauli:
Actions of noncompact groups and algorithm design: a case study. 47-50 - Rodney A. Kennedy, Deva K. Borah, Zhi Ding:
Discretization issues for the design of optimal blind algorithms. 51-54 - Zhuquan Zang, Antonio Cantoni, Kok Lay Teo:
Continuous-time envelope-constrained filter design via Laguerre filters and H∞ optimization methods. 55-58 - Jeroen Dehaene, Nanayaa Twum-Danso:
Local adaptive algorithms for information maximization in neural networks, and application to source separation. 59-62 - Kutluyil Dogançay, Vikram Krishnamurthy:
Quick aggregation of Markov chain functionals via stochastic complementation. 63-66 - John B. Moore, Danchi Jiang:
A rank preserving flow algorithm for quadratic optimization problems subject to quadratic equality constraints. 67-70
Speech-to-Speech Translation
- Thomas Bub, Wolfgang Wahlster, Alex Waibel:
Verbmobil: the combination of deep and shallow processing for spontaneous speech translation. 71-74 - Heinrich Niemann, Elmar Nöth, Andreas Kießling, Ralf Kompe, Anton Batliner:
Prosodic processing and its use in VERBMOBIL. 75-78 - Hans Ulrich Block:
The language components in Verbmobil. 79-82 - Michael Finke, Petra Geutner, Hermann Hild, Thomas Kemp, Klaus Ries, Martin Westphal:
The Karlsruhe-Verbmobil speech recognition engine. 83-86 - Jae-Woo Yang, Jun Park:
An experiment on Korean-to-English and Korean-to-Japanese spoken language translation. 87-90 - Bianca Angelini, Mauro Cettolo, Anna Corazza, Daniele Falavigna, Gianni Lazzari:
Multilingual person to person communication at IRST. 91-94 - Tohru Shimizu, Harald Singer, Yoshinori Sagisaka:
Fast word-graph generation for spontaneous conversational speech translation. 95-98 - Alon Lavie, Alex Waibel, Lori S. Levin, Michael Finke, Donna Gates, Marsal Gavaldà, Torsten Zeppenfeld, Puming Zhan:
Janus-III: speech-to-speech translation in multiple languages. 99-102 - Hiyan Alshawi, Adam L. Buchsbaum:
State-transition cost functions and an application to language translation. 103-106 - Manny Rayner, David M. Carter:
Hybrid language processing in the Spoken Language Translator. 107-110 - Enrique Vidal:
Finite-state speech-to-speech translation. 111-114 - Shoji Hiraoka, Masakatsu Hoshimi, Kenji Matsui, Jean-Claude Junqua:
An experimental bidirectional Japanese/English interpreting video phone system using Internet. 115-118
Advanced Neural Applications
- Christian Goerick, Bernhard Sendhoff, Werner von Seelen:
From neural networks to neural strategies. 119-122 - Rosaria Silipo, Giovanni Bortolan:
Neural and traditional techniques in diagnostic ECG classification. 123-126 - Dragan Obradovic, Gustavo Deco:
Unsupervised learning for blind source separation: an information-theoretic approach. 127-130 - Juha Karhunen, Aapo Hyvärinen, Ricardo Vigário, Jarmo Hurri, Erkki Oja:
Applications of neural blind separation to signal and image processing. 131-134 - Mark D. Plumbley
:
Communications and neural networks: theory and practice. 135-138 - Joachim M. Buhmann, Thomas Hofmann:
Robust vector quantization by competitive learning. 139-142 - Thomas Vetter:
Recognizing faces from a new viewpoint. 143-146 - Wolfgang Utschick, Josef A. Nossek:
Hybrid optimization of feedforward neural networks for handwritten character recognition. 147-150 - Yann LeCun, Léon Bottou, Yoshua Bengio:
Reading checks with multilayer graph transformer networks. 151-154 - Martin Schlang, Einar Bröse, Björn Feldkeller, Otto Granckow, Michael Jansen, Thomas Poppe, Clemens Schäffner, Günter Sörgel:
Neural networks for process control in steel manufacturing. 155-158 - Peter Marbach, John N. Tsitsiklis:
A neuro-dynamic programming approach to admission control in ATM networks: the single link case. 159-162
Signal Processing Technology for Multi-Media Human-Machine Interaction
- James L. Flanagan, Ivan Marsic:
Issues in measuring the benefits of multimodal interfaces. 163-166 - Alex Waibel, Bernhard Suhm, Minh Tue Vo, Jie Yang:
Multimodal interfaces for multimedia information agents. 167-170 - Alex Pentland:
Smart rooms, desks and clothes. 171-174 - Nikil Jayant:
Human machine interaction by voice and gesture. 175-177 - Tsuhan Chen, Ram Rao:
Audio-visual interaction in multimedia communication. 179-182 - Fabio Lavagetto, Skjalg Lepsøy, Carlo Braccini, Sergio Curinga:
Lip motion modeling and speech driven estimation. 183-186 - Hong Wang, Peter Chu:
Voice source localization for automatic camera pointing system in videoconferencing. 187-190 - Akihito Akutsu, Yoshinobu Tonomura, Hiroshi Hamada:
Video interface for spatiotemporal interactions based on multi-dimensional video computing. 191-194 - Alexander G. Hauptmann, Howard D. Wactlar:
Indexing and search of multimodal information. 195-198 - Steve J. Young, Martin G. Brown, Jonathan Foote, Gareth J. F. Jones, Karen Spärck Jones:
Acoustic indexing for multimedia retrieval and browsing. 199-202 - Francis Kubala, Hubert Jin, Spyros Matsoukas, Long Nguyen, Richard M. Schwartz:
Broadcast news transcription. 203-206 - Ryohei Nakatsu:
Image/speech processing that adopts an artistic approach-toward integration of art and technology. 207-210
Microphone Array Signal Processing
- Jens Meyer, Carsten Sydow:
Noise cancelling for microphone arrays. 211-213 - Kenji Kiyohara, Yutaka Kaneda, Satoshi Takahashi, Hiroaki Nomura, Junji Kojima:
A microphone array system for speech recognition. 215-218 - Walter Kellermann:
Strategies for combining acoustic echo cancellation and adaptive beamforming microphone arrays. 219-222 - Gary W. Elko, Anh-Tho Nguyen Pong:
A steerable and variable first-order differential microphone array. 223-226 - Maurizio Omologo, Marco Matassoni, Piergiorgio Svaizer, Diego Giuliani:
Microphone array based speech recognition with different talker-array positions. 227-230 - Piergiorgio Svaizer, Marco Matassoni, Maurizio Omologo:
Acoustic source location in a three-dimensional space using crosspower spectrum phase. 231-234 - Peter L. Chu:
Superdirective microphone array for a set-top videoconferencing system. 235-238 - Mattias Dahl, Ingvar Claesson, Sven Nordebo:
Simultaneous echo cancellation and car noise suppression employing a microphone array. 239-242 - Sven Nordholm, Ingvar Claesson:
Analytical evaluation of a self-calibrating microphone array. 243-246 - Yves Grenier, Sofiène Affes:
Microphone array response to speaker movements. 247-250 - Harvey F. Silverman, William R. Patterson III, James L. Flanagan, Daniel V. Rabinkin:
A digital processing system for source location and sound capture by large microphone arrays. 251-254
DSP for Mobile Communication
- Martin Haardt, Josef A. Nossek:
3-D unitary ESPRIT for joint 2-D angle and carrier estimation. 255-258 - Thomas Hindelang, Wen Xu, Christian Erben:
Quality enhancement of coded and corrupted speeches in GSM mobile systems using residual redundancy. 259-262 - Fuyun Ling:
Pilot assisted coherent DS-CDMA reverse-link communications with optimal robust channel estimation. 263-266 - Christian Bergogne, Philippe Sehier, Michel Bousquet:
A new frequency estimator applied to burst transmission. 267-270 - Thorsten Grötker, Rainer Schoenen, Heinrich Meyr:
Unified specification of control and data flow. 271-274 - Jan M. Rabaey:
Reconfigurable processing: the solution to low-power programmable DSP. 275-278 - Gerhard P. Fettweis:
DSP cores for mobile communications: where are we going? 279-282 - Sanjay Kasturia, Raziel Haimi-Cohen, Colin A. Warwick:
DSPs in mobile communication in the United States. 283-286 - Markus Willems, Volker Bürsgens, Thorsten Grötker, Heinrich Meyr:
FRIDGE: an interactive code generation environment for HW/SW codesign. 287-290 - Ravi Subramanian, Marc Barberis, Herbert Dawid, Klaus-Jürgen Koch:
Staying ahead of the game in silicon for digital mobile communications. 291-294
Echo Cancellation
- Christiane Antweiler, Jörn Grunwald, Holger Quack:
Approximation of optimal step size control for acoustic echo cancellation. 295-298 - Shoji Makino, Klaus Strauss, Suehiro Shimauchi, Yoichi Haneda, Akira Nakagawa:
Subband stereo echo canceller using the projection algorithm with fast convergence to the true echo path. 299-302 - Jacob Benesty, Dennis R. Morgan, M. Mohan Sondhi:
A better understanding and an improved solution to the problems of stereophonic acoustic echo cancellation. 303-306 - Valérie Turbin, André Gilloire, Pascal Scalart:
Comparison of three post-filtering algorithms for residual acoustic echo reduction. 307-310
Audio Coding & Transducer
- Wen-Whei Chang, De-Yu Wang, Li-Wei Wang:
Audio coding using sinusoidal excitation representation. 311-314 - Xiang Wei, Martyn J. Shaw, Martin R. Varley:
Optimum bit allocation and decomposition for high quality audio coding. 315-318 - Karine Hay, Laurent Mainard, Samir Saoudi:
The D5 lattice quantization for 64 kbit/s low-delay subband audio coder with a 15 kHz bandwidth. 319-322 - Aki Härmä, Unto K. Laine, Matti Karjalainen:
An experimental audio codec based on warped linear prediction of complex valued signals. 323-326 - William Kurt Dobson, Jiankan Jack Yang, Kevin J. Smart, Feng Kathy Guo:
High quality low complexity scalable wavelet audio coding. 327-330 - Yuichiro Takamizawa, Masahiro Iwadare, Akihiko Sugiyama:
An efficient tonal component coding algorithm for MPEG-2 Audio NBC. 331-334 - Roch Lefebvre, Claude Laflamme:
Spectral amplitude warping (SAW) for noise spectrum shaping in audio coding. 335-338 - Carlos A. Serantes, Antonio S. Pena, Nuria González-Prelcic:
A fast noise-scaling algorithm for uniform quantization in audio coding schemes. 339-342 - Daniele Cadel, Giorgio Parladori:
Pyramid vector coding for high quality audio compression. 343-346 - Karine Gosse, François Moreau de Saint-Martin, Xavier Durot, Pierre Duhamel, Jean-Bernard Rault:
Subband audio coding with synthesis filters minimizing a perceptual distortion. 347-350 - Simon Boland, Mohamed A. Deriche:
New results in low bitrate audio coding using a combined harmonic-wavelet representation. 351-354 - Wolfgang J. Klippel:
Adaptive inverse control of weakly nonlinear systems. 355-358
Microphone Array & Active Noise Control
- Sven Fischer, Karl-Dirk Kammeyer:
Broadband beamforming with adaptive postfiltering for speech acquisition in noisy environments. 359-362 - James G. Ryan, Rafik A. Goubran:
Near-field beamforming for microphone arrays. 363-366 - Osamu Hoshuyama, Akihiko Sugiyama, Akihiro Hirano:
A robust adaptive microphone array with improved spatial selectivity and its evaluation in a real environment. 367-370 - Douglas E. Sturim, Michael S. Brandstein, Harvey F. Silverman:
Tracking multiple talkers using microphone-array measurements. 371-374 - Michael S. Brandstein, Harvey F. Silverman:
A robust method for speech signal time-delay estimation in reverberant rooms. 375-378 - Jie Gu, Sze-Fong Yau:
A model-based approach to active noise cancellation using loudspeaker array. 379-382 - Wolfgang Täger, Yannick Mahieux:
Reverberant sound field analysis using a microphone array. 383-386 - Alberto González, Antonio Albiol, Steve J. Elliott:
Minimisation of the maximum error signal in active control. 387-390 - Jeong Hyeon Yun, Young-Cheol Park, Dae Hee Youn:
Subband active noise control algorithm based on a delayless subband adaptive filter architecture. 391-394 - Paul Strauch, Bernard Mulgrew:
Nonlinear active noise control in a linear duct. 395-398 - Scott C. Douglas:
Fast exact filtered-X LMS and LMS algorithms for multichannel active noise control. 399-402 - Toshifumi Kosakat, Stephen J. Elliott, Christopher C. Boucher:
A novel frequency domain filtered-X LMS algorithm for active noise reduction. 403-406
Hearing Aids and Computer Music
- Dorra Masmoudi, Dominique Dallet, Jean Paul Dom:
Practical supergrain head sized arrays. 407-410 - Todd Schneider, Robert L. Brennan:
A multichannel compression strategy for a digital hearing aid. 411-414 - Paul W. Shields, Douglas R. Campbell:
Multi-microphone sub-band adaptive signal processing for improvement of hearing aid performance: primarily results using normal hearing volunteers. 415-418 - Kenzo Itoh, Masahide Mizushima:
Environmental noise reduction based on speech/non-speech identification for hearing aids. 419-422 - Russell H. Lambert, Anthony J. Bell:
Blind separation of multiple speakers in a multipath environment. 423-426 - Fernando De Bernardinis, Roberto Roncella, Roberto Saletti, Pierangelo Terreni, Graziano Bertini:
A single-chip 1, 200 sinusoid real-time generator for additive synthesis of musical signals. 427-430 - Carlo Drioli, Davide Rocchesso:
A generalized musical-tone generator with application to sound compression and synthesis. 431-434 - Michael W. Macon, Leslie Jensen-Link, James Oliverio, Mark A. Clements, E. Bryan George:
A singing voice synthesis system based on sinusoidal modeling. 435-438 - Khaled N. Hamdy, Ahmed H. Tewfik, Ting Chen, Satoshi Takagi:
Time-scale modification of audio signals with combined harmonic and wavelet representations. 439-442 - Erhard Rank, Gernot Kubin:
A waveguide model for slapbass synthesis. 443-446 - Shao-Po Wu, William Putnam:
Minimum perceptual spectral distance FIR filter design. 447-450 - Xiaoshu Qian, Yinong Ding:
A phase interpolation algorithm for sinusoidal model based music synthesis. 451-454 - Stephan Tassart, Philippe Depalle:
Analytical approximations of fractional delays: Lagrange interpolators and allpass filters. 455-458 - Lauri Savioja, Vesa Välimäki:
Improved discrete-time modeling of multi-dimensional wave propagation using the interpolated digital waveguide mesh. 459-462
Matched Field Processing
- Christoph F. Mecklenbräuker, Peter Gerstoft, Pei-Jung Chung, Johann F. Böhme:
Generalized likelihood ratio test for selecting a geo-acoustic environmental model. 463-466 - Maria-João Rendas, Georges Bienvenu:
Tuning genetic algorithms for underwater acoustics using a priori statistical information. 467-470 - Kerem Harmanci, Jeffrey L. Krolik:
Robust beamformer weight design for broadband matched-field processsing. 471-474 - Brian F. Harrison, Richard J. Vaccaro, Donald W. Tufts:
Fastmap: a fast, approximate maximum a posteriori probability parameter estimator with application to robust matched-field processing. 475-478 - Donald F. Gingras, Peter Gerstoft, Neil L. Gerr, Christoph F. Mecklenbräuker:
Electromagnetic matched field processing for source localization. 479-482
Detection, Classification and Localisation
- Ivars P. Kirsteins, Sanjay K. Mehta, John W. Fay:
Power-law processors for detecting unknown signals in colored noise. 483-486 - Vittorio Rampa, Umberto Spagnolini:
Multitarget detection/tracking of echoes with known waveform: algorithm and applications. 487-490 - Francisco M. Garcia, Isabel M. G. Lourtie:
Detection of Gaussian bandpass transients under impulsive noise: a wavelet transform approach. 491-494 - Gilles Dassot, Roland Blanpain, Claude Jauffret:
Maximum likelihood estimator for magneto-acoustic localisation. 495-498 - Joseph Tabrikian, Jeffrey L. Krolik:
Barankin bound for source localization in shallow water. 499-502 - Zoi-Heleni Michalopoulou:
Underwater transient signal processing: marine mammal identification, localization, and source signal deconvolution. 503-506 - Alexander Goldin:
Numerical optimization of non-adaptive microphone arrays. 507-510 - Jason Goldberg, Ana I. Pérez-Neira, Miguel Angel Lagunas:
Joint direction-of-arrival and array shape tracking for multiple moving targets. 511-514 - Mark L. Krieg, Douglas A. Gray:
Comparison of probabilistic least squares and probabilistic multi-hypothesis tracking algorithms for multi-sensor tracking. 515-518 - Alex B. Gershman, Christoph F. Mecklenbräuker, Johann F. Böhme:
Direction finding with imperfect wavefront coherence: a matrix fitting approach using genetic algorithm. 519-522 - Saman S. Abeysekera, Yee Hong Leung:
Design of an optimum wideband active sonar array with robustness. 523-526
Ocean Signal Processing
- Jean-Jacques Fuchs:
Multipath time-delay estimation. 527-530 - Robert B. MacLeod, Richard J. Vaccaro:
Fast maximum likelihood estimation with multiple signal initialization. 531-534 - Amir W. Habboosh, Richard J. Vaccaro, Steven M. Kay:
An algorithm for detecting closely spaced delay/Doppler components. 535-538 - Shi-Quan Wu, Hing-Cheung So, Pak-Chung Ching:
Improvement of TDOA measurement using wavelet denoising with a novel thresholding technique. 539-542 - Charles W. Therrien, K. L. Frack Jr., Natanael Ruiz Fontes:
A short-time Wiener filter for noise removal in underwater acoustic data. 543-546 - Donald W. Tufts, Edward C. Real, James W. Cooley:
Fast approximate subspace tracking (FAST). 547-550 - Brian E. Freburger, Donald W. Tufts, Tom A. Palka:
A subspace framework for fast parameter estimation with known waveforms. 551-554 - Nirmal Keshava, José M. F. Moura:
Terrain classification in polarimetric SAR using wavelet packets. 555-558 - Michael Papazoglou, Jeffrey L. Krolik:
Electromagnetic matched-field processing for target height finding with over-the-horizon radar. 559-562 - Geoff Roberts, Abdelhak M. Zoubir, Boualem Boashash:
Time-frequency classification using a multiple hypotheses test: an application to the classification of humpback whale signals. 563-566 - Jianguo Huang, Jianping Zhao, Yiqing Xie:
Source classification using pole method of AR model. 567-570 - Hongya Ge:
The LMMSE estimate-based multiuser detector: performance analyses and adaptive implementation. 571-574 - Mark Johnson, Lee E. Freitag, Milica Stojanovic:
Improved Doppler tracking and correction for underwater acoustic communications. 575-578 - Bayan S. Sharif, Jeffrey A. Neasham, David Thompson, Oliver R. Hinton, Alan E. Adams:
A blind multichannel combiner for long range underwater communications. 579-582
DSP Processors
- Jon Mellott, Fred J. Taylor:
Very long instruction word architectures for digital signal processing. 583-586 - Christoph Baumhof, Frank Müller, Otto Müller, Manfred Schlett:
A novel 32 bit RISC architecture unifying RISC and DSP. 587-590 - Hisakazu Sato, Edgar Holmann, Toyohiko Yoshida, Masahito Matsuo, Toru Kengaku:
A dual-issue RISC processor for multimedia signal processing. 591-594 - Elmar Maas, Dirk Herrmann, Rolf Ernst, Peter Rüffer, Sieghard Hasenzahl, Martin Seitz:
A processor-coprocessor architecture for high end video applications. 595-598 - Yasushi Ooi, Osamu Ohnishi, Yutaka Yokoyama, Yoichi Katayama, Masayuki Mizuno, Masakazu Yamashina, Hideo Takano, Naoya Hayashi, Ichiro Tamitani:
An MPEG-2 encoder architecture based on a single-chip dedicated LSI with a control MPU. 599-602 - Xiao-Dong Zhang, Chi-Ying Tsui:
An efficient and reconfigurable VLSI architecture for different block matching motion estimation algorithms. 603-606 - Konstantina Karagianni, George Diamantakos, Vassilis Paliouras, Thanos Stouraitis:
An operation-saving VLSI geometry engine core. 607-610 - Martin Grajcar, Bernhard Sick:
The FFT butterfly operation in 4 processor cycles on a 24 bit fixed-point DSP with a pipelined multiplier. 611-614 - Shen-Fu Hsiao, Chung-Yi Yen:
New unified VLSI architectures for computing DFT and other transforms. 615-618 - Mohit K. Prasad, Paul D'Arcy, Arup Gupta, Marc S. Diamondstein, Hosahalli R. Srinivas:
Half-rate GSM vocoder implementation on a dual mac digital signal processor. 619-622 - Carlos Cabrera, Montserrat Bóo, Javier D. Bruguera:
VLSI implementation of an area-efficient architecture for the Viterbi algorithm. 623-626
DSP Building Blocks & Arithmetic
- Leilei Song, Keshab K. Parhi:
Low-area dual basis divider over GF(2M). 627-630 - Wolfram Drescher, Kay Bachmann, Gerhard P. Fettweis:
VLSI architecture for datapath integration of arithmetic over GF(2 m) on digital signal processors. 631-634 - Seunghyeon Nahm, Wonyong Sung:
A fast direction sequence generation method for CORDIC processors. 635-638 - Chieh-Chih Li, Sau-Gee Chen:
A radix-4 redundant CORDIC algorithm with fast on-line variable scale factor compensation. 639-642 - Jun Ma, Keshab K. Parhi, Ed F. Deprettere:
Pipelining of cordic based IIR digital filters. 643-646 - Chris J. Myers, Hao Zheng:
An asynchronous implementation of the maxlist algorithm. 647-650 - Wolfgang Wilhelm, Tobias G. Noll:
A novel systematic mapping approach for highly efficient multiplexed FIR-filter architectures. 651-654
Design Automation for DSP
- Rainer Schoenen, Vojin Zivojnovic, Heinrich Meyr:
An upper bound of the throughput of multirate multiprocessor schedules. 655-658 - Inki Hong, Miodrag Potkonjak:
Minimizing the number of operations in DSP computations. 659-662 - Shahram Famorzadeh, Vijay K. Madisetti, Thomas Egolf, Tuongvu Nguyen:
BEEHIVE: an adaptive, distributed, embedded signal processing environment. 663-666 - Jan Jonsson, Jonas Vasell:
On objective function selection in list scheduling algorithms for digital signal processing applications. 667-670 - Jean-Philippe Diguet, Olivier Sentieys, Daniel Chillet, Jean Luc Philippe:
VLSI high level synthesis of fast exact least mean square algorithms based on fast FIR filters. 671-674 - John V. McCanny, Douglas Ridge, Yi Hu, Jill K. Hunter:
Hierarchical VHDL libraries for DSP ASIC design. 675-678 - Chunho Lee, Darko Kirovski, Inki Hong, Miodrag Potkonjak:
DSP Quant: design, validation, and applications of DSP hard real-time benchmark. 679-682 - Bernhard Wess, Martin Gotschlich:
Constructing memory layouts for address generation units supporting offset 2 access. 683-686 - Markus Willems, Holger Keding, Vojin Zivojnovic, Heinrich Meyr:
Modulo-addressing utilization in automatic software synthesis for digital signal processors. 687-690 - Werner Kreuzer, Bernhard Wess:
Cooperative register assignment and code compaction for digital signal processors with irregular datapaths. 691-694 - Ashok Sudarsanam, Sharad Malik, Steven W. K. Tjiang, Stan Y. Liao:
Optimization of embedded DSP programs using post-pass data-flow analysis. 695-698 - Hans-Joachim Stolberg, Masao Ikekawa, Ichiro Kuroda:
Code positioning to reduce instruction cache misses in signal processing applications on multimedia RISC processors. 699-702 - Takashi Miyazaki, Edward A. Lee:
Code generation by using integer-controlled dataflow graph. 703-706 - Jiyang Kang, Wonyong Sung:
Fixed-point C compiler for TMS320C50 digital signal processor. 707-710
Volume 2: Speech Processing
Recognizing Broadcast News
- Raimo Bakis, Scott Saobing Chen, Ponani S. Gopalakrishnan, Ramesh Gopinath, Stéphane H. Maes, Lazaros Polymenakos:
Transcription of broadcast news-system robustness issues and adaptation techniques. 711-714 - Jean-Luc Gauvain, Gilles Adda, Lori Lamel, Martine Adda-Decker:
Transcribing broadcast news shows. 715-718 - Philip C. Woodland, Mark J. F. Gales, David Pye, Steve J. Young:
Broadcast news transcription using HTK. 719-722 - Gary D. Cook, Dan J. Kershaw, James Christie, Carl W. Seymour, Steve R. Waterhouse:
Transcription of broadcast television and radio news: the 1996 ABBOT system. 723-726 - Toru Imai, Richard M. Schwartz, Francis Kubala, Long Nguyen:
Improved topic discrimination of broadcast news using a model of multiple simultaneous topics. 727-730
CELP Speech Coding
- Tero Honkanen, Janne Vainio, Kari Järvinen, Petri Haavisto, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul:
Enhanced full rate speech codec for IS-136 digital cellular system. 731-734 - Lei Zhang, Tian Wang, Vladimir Cuperman:
A CELP variable rate speech codec with low average rate. 735-738 - Mustapha Bouraoui, Francois Druilhe, Gang Feng:
HCELP: low bit rate speech coder for voice storage applications. 739-742 - Chul Hong Kwon, Chong Kwan Un:
Low-rate CELP speech coding using an improved weighting function. 743-746 - Pasi Ojala:
Toll quality variable-rate speech codec. 747-750 - Erdal Paksoy, Alan McCree, Vishu Viswanathan:
A variable rate multimodal speech coder with gain-matched analysis-by-synthesis. 751-754 - Kazunori Mano:
Design of a toll-quality 4-kbit/s speech coder based on phase-adaptive PSI-CELP. 755-758 - Soon Y. Kwon, Hochong Park, Hyokang Chang:
A high quality BI-CELP speech coder at 8 kbit/s and below. 759-762 - Jayesh Patel:
Low complexity VQ for multi-tap pitch predictor coding. 763-766 - Hong Kook Kim, Yong Duk Cho, Moo Young Kim, Sang Ryong Kim:
A 4 kbit/s renewal code excited linear prediction speech coder. 767-770 - Kari Järvinen, Janne Vainio, Pekka Kapanen, Tero Honkanen, Petri Haavisto, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul:
GSM enhanced full rate speech codec. 771-774 - Redwan Salami, Claude Laflamme, Bruno Bessette, Jean-Pierre Adoul:
Description of ITU-T Recommendation G.729 Annex A: reduced complexity 8 kbit/s CS-ACELP codec. 775-778
Language modeling
- Reinhard Kneser, Jochen Peters:
Semantic clustering for adaptive language modeling. 779-782 - Hirokazu Masataki, Yoshinori Sagisaka, Kazuya Hisaki, Tatsuya Kawahara:
Task adaptation using MAP estimation in N-gram language modeling. 783-786 - Michael Simons, Hermann Ney, Sven C. Martin:
Distant bigram language modelling using maximum entropy. 787-790 - Eric Sven Ristad, Robert G. Thomas:
Nonuniform Markov models. 791-794 - Thomas Niesler, Philip C. Woodland:
Modelling word-pair relations in a category-based language model. 795-798 - Philip Clarkson, Anthony J. Robinson:
Language model adaptation using mixtures and an exponentially decaying cache. 799-802 - Stefan Besling, Hans-Günter Meier:
Confidence-driven estimator perturbation: BMPC [Best Model Perturbation within Confidence]. 803-806 - Joerg P. Ueberla:
Domain adaptation with clustered language models. 807-810 - Ralf Kompe, Andreas Kießling, Heinrich Niemann, Elmar Nöth, Anton Batliner, Stefanie Schachtl, Tobias Ruland, Hans Ulrich Block:
Improving parsing of spontaneous speech with the help of prosodic boundaries. 811-814 - Cosmin Popovici, Paolo Baggia:
Specialized language models using dialogue predictions. 815-818 - Germán Bordel, Amparo Varona, M. Inés Torres:
K-TLSS(S) language models for speech recognition. 819-822 - Carlos Crespo, Daniel Tapias, Gregorio Escalada, Jorge Alvarez:
Language model adaptation for conversational speech recognition using automatically tagged pseudo-morphological classes. 823-826
Noise Robustness
- Tetsuya Takiguchi, Satoshi Nakamura, Qiang Hou, Kiyohiro Shikano:
Model adaptation based on HMM decomposition for reverberant speech recognition. 827-830 - Driss Matrouf, Jean-Luc Gauvain:
Model compensation for noises in training and test data. 831-834 - Shigeki Sagayama, Yoshikazu Yamaguchi, Satoshi Takahashi, Jun-ichi Takahashi:
Jacobian approach to fast acoustic model adaptation. 835-838 - Mohamed Afify, Yifan Gong, Jean-Paul Haton:
A unified maximum likelihood approach to acoustic mismatch compensation: application to noisy Lombard speech recognition. 839-842 - Beth T. Logan, Anthony J. Robinson:
Enhancement and recognition of noisy speech within an autoregressive hidden Markov model framework using noise estimates from the noisy signal. 843-846 - Hiroki Yamamoto, Tetsuo Kosaka, Masayuki Yamada, Yasuhiro Komori, Minoru Fujita:
Fast speech recognition algorithm under noisy environment using modified CMS-PMC and improved IDMM+SQ. 847-850 - Bhiksha Raj, Vipul N. Parikh, Richard M. Stern:
The effects of background music on speech recognition accuracy. 851-854 - Jenq-Neng Hwang, Chien-Jen Wang:
Joint model and feature space optimization for robust speech recognition. 855-858 - Kuan-Chieh Yen, Yunxin Zhao:
Co-channel speech separation for robust automatic speech recognition: stability and efficiency. 859-862 - Martin P. Cooke, Andrew C. Morris, Phil D. Green:
Missing data techniques for robust speech recognition. 863-866 - Detlef Hardt, Klaus Fellbaum:
Spectral subtraction and RASTA-filtering in text-dependent HMM-based speaker verification. 867-870 - Kari Laurila:
Noise robust speech recognition with state duration constraints. 871-874
Word Spotting with Confidence
- Thomas Schaaf, Thomas Kemp:
Confidence measures for spontaneous speech recognition. 875-878 - Larry Gillick, Yoshiko Ito, Jonathan Young:
A probabilistic approach to confidence estimation and evaluation. 879-882 - Chalapathy Neti, Salim Roukos, Ellen Eide:
Word-based confidence measures as a guide for stack search in speech recognition. 883-886 - Mitch Weintraub, Françoise Beaufays, Zeév Rivlin, Yochai Konig, Andreas Stolcke:
Neural-network based measures of confidence for word recognition. 887-890 - Javier Caminero, Luis A. Hernández Gómez, Celinda de la Torre, Cesar Martín del Alamo:
Improving utterance verification using hierarchical confidence measures in continuous natural numbers recognition. 891-894 - Antonio J. Rubio, Jesús Esteban Díaz Verdejo, Pedro García-Teodoro, José C. Segura:
On the influence of frame-asynchronous grammar scoring in a CSR system. 895-898 - Alexandros S. Manos, Victor W. Zue:
A segment-based wordspotter using phonetic filler models. 899-902 - Bo-Ren Bai, Chiu-yu Tseng, Lin-Shan Lee:
A multi-phase approach for fast spotting of large vocabulary Chinese keywords from Mandarin speech using prosodic information. 903-906 - Rachida El Méliani, Douglas D. O'Shaughnessy:
Accurate keyword spotting using strictly lexical fillers. 907-910 - Martin Holzapfel, Günther Ruske, Harald Höge:
Failure simulation for a phoneme HMM based keyword spotter. 911-914 - Suhardi, Klaus Fellbaum:
Wordspotting using a predictive neural model for the telephone speech corpus. 915-918
Speech Synthesis
- Mat P. Pollard, Barry M. G. Cheetham, Colin C. Goodyear, Mike D. Edgington:
Shape-invariant pitch and time-scale modification of speech by variable order phase interpolation. 919-922 - Fu-Chiang Chou, Chiu-yu Tseng, Keh-Jiann Chen, Lin-Shan Lee:
A Chinese text-to-speech system based on part-of-speech analysis, prosodic modeling and non-uniform units. 923-926 - Eduardo López Gonzalo, Jose M. Rodriguez-Garcia, Luis A. Hernández Gómez, Juan Manuel Villar-Navarro:
Automatic prosodic modeling for speaker and task adaptation in text-to-speech. 927-930 - Gerit P. Sonntag, Thomas Portele, Barbara Heuft:
Prosody generation with a neural network: weighing the importance of input parameters. 931-934 - Hiroshi Ohmura, Kazuyo Tanaka:
Evaluation of a speech synthesis method for nonlinear modeling of vocal folds vibration effect. 935-938 - Heo-Jin Byeon, Yeon-Jun Kim, Yung-Hwan Oh:
Generation of F0 contour using stochastic mapping and vector quantization control parameters. 939-942 - Bryan L. Pellom, John H. L. Hansen:
Spectral normalization employing hidden Markov modeling of line spectrum pair frequencies. 943-946 - Rivarol Vergin, Douglas D. O'Shaughnessy, Azarshid Farhat:
Time domain technique for pitch modification and robust voice transformation. 947-950 - Kimihito Tanaka, Masanobu Abe:
A new fundamental frequency modification algorithm with transformation of spectrum envelope according to F0. 951-954 - Burhan Necioglu, Mark A. Clements, Thomas P. Barnwell III:
Reliability assessment and evaluation of objectively measured descriptors for perceptual speaker characterization. 955-958 - Xuedong Huang, Alex Acero, Hsiao-Wuen Hon, Yun-Cheng Ju, Jingsong Liu, Scott Meredith, Mike Plumpe:
Recent improvements on Microsoft's trainable text-to-speech system-Whistler. 959-962 - Takehiko Kagoshima, Masami Akamine:
Automatic generation of speech synthesis units based on closed loop training. 963-966
Speech Features and Acoustic Modeling
- Tomio Takara, Kazuya Higa, Itaru Nagayama:
Isolated word recognition using the HMM structure selected by the genetic algorithm. 967-970 - Satoshi Takahashi, Kiyoaki Aikawa, Shigeki Sagayama:
Discrete mixture HMM. 971-974 - Luciano Fissore, Pietro Laface, Franco Ravera:
Using word temporal structure in HMM speech recognition. 975-978 - Zhihong Hu, Etienne Barnard:
Smoothness analysis for trajectory features. 979-982 - Srinivasan Umesh, Leon Cohen, Douglas J. Nelson:
Frequency-warping and speaker-normalization. 983-986 - Su-Lin Wu, Michael L. Shire, Steven Greenberg, Nelson Morgan:
Integrating syllable boundary information into speech recognition. 987-990 - Philipp Schmid, Etienne Barnard:
Explicit, N-best formant features for vowel classification. 991-994 - Jayadev Billa:
Dual-channel auditory spectrum modeling. 995-998 - Régine André-Obrecht, Bruno Jacob:
Direct identification vs. correlated models to process acoustic and articulatory informations in automatic speech recognition. 999-1002 - Thierry Soulas, Chafic Mokbel, Denis Jouvet, Jean Monné:
Adapting PSN recognition models to the GSM environment by using spectral transformation. 1003-1006 - Li Deng:
Integrated-multilingual speech recognition using universal phonological features in a functional speech production model. 1007-1010 - Stephen A. Zahorian, Peter L. Silsbee, Xihong Wang:
Phone classification with segmental features and a binary-pair partitioned neural network classifier. 1011-1014
Speaker Adaption and Normalisation
- Tomoko Matsui, Tatsuo Matsuoka, Sadaoki Furui:
Smoothed N-best-based speaker adaptation for speech recognition. 1015-1018 - Michael Schüßler, Florian Gallwitz, Stefan Harbeck:
A fast algorithm for unsupervised incremental speaker adaptation. 1019-1022 - Shigeru Homma, Kiyoaki Aikawa, Shigeki Sagayama:
Improved estimation of supervision in unsupervised speaker adaptation. 1023-1026 - Jen-Tzung Chien, Chin-Hui Lee, Hsiao-Chuan Wang:
Improved Bayesian learning of hidden Markov models for speaker adaptation. 1027-1030 - Venkatesh Nagesha, Larry Gillick:
Studies in transformation-based adaptation. 1031-1034 - Eric Thelen, Xavier L. Aubert, Peter Beyerlein:
Speaker adaptation in the Philips system for large vocabulary continuous speech recognition. 1035-1038 - Puming Zhan, Martin Westphal:
Speaker normalization based on frequency warping. 1039-1042 - Tasos Anastasakos, John W. McDonough, John Makhoul:
Speaker adaptive training: a maximum likelihood approach to speaker normalization. 1043-1046 - David Pye, Philip C. Woodland:
Experiments in speaker normalisation and adaptation for large vocabulary speech recognition. 1047-1050 - Yasuo Ariki:
Effectiveness of speaker normalized HMM by projection to speaker subspace. 1051-1054 - Jun Ishii, Masahiro Tonomura:
Speaker normalization and adaptation based on linear transformation. 1055-1058 - John W. McDonough, Tasos Anastasakos, George Zavaliagkos, Herbert Gish:
Speaker-adapted training on the Switchboard Corpus. 1059-1062
Speaker Verification and Identification
- Françoise Beaufays, Mitch Weintraub:
Model transformation for robust speaker recognition from telephone data. 1063-1066 - Lori Lamel, Jean-Luc Gauvain:
Speaker recognition with the Switchboard corpus. 1067-1070 - Larry P. Heck, Mitch Weintraub:
Handset-dependent background models for robust text-independent speaker recognition. 1071-1074 - Pierre Castellano, Stefan Slomka, Sridha Sridharan:
Telephone based speaker recognition using multiple binary classifier and Gaussian mixture models. 1075-1078 - Stephan Euler, Rainer Langlitz, Joachim Zinke:
Comparison of whole word and subword modeling techniques for speaker verification with limited training data. 1079-1082 - Michael J. Carey, Eluned S. Parris, Stephen J. Bennett, Harvey Lloyd-Thomas:
A comparison of model estimation techniques for speaker verification. 1083-1086 - Seiichi Nakagawa, Konstantin P. Markov:
Speaker verification using frame and utterance level likelihood normalization. 1087-1090 - Jialong He, Li Liu, Günther Palm:
A new codebook training algorithm for VQ-based speaker recognition. 1091-1094 - Stanley J. Wenndt, Sanyogita Shamsunder:
Bispectrum features for robust speaker identification. 1095-1098 - Deb K. Roy, Carl Malamud:
Speaker identification based text to audio alignment for an audio retrieval system. 1099-1102 - Joaquín González-Rodríguez, Javier Ortega-Garcia:
Robust speaker recognition through acoustic array processing and spectral normalization. 1103-1106 - Javier Ortega-Garcia, Joaquín González-Rodríguez:
Providing single and multi-channel acoustical robustness to speaker identification systems. 1107-1110
Language and Speaker Identification
- James Hieronymus, Shubha Kadambe:
Robust spoken language identification using large vocabulary speech recognition. 1111-1114 - Jirí Navrátil, Werner Zühlke:
Double bigram-decoding in phonotactic language identification. 1115-1118 - Etienne Marcheret, Michael I. Savic:
Random walk theory applied to language identification. 1119-1122 - Levent M. Arslan, John H. L. Hansen:
Frequency characteristics of foreign accented speech. 1123-1126 - Mübeccel Demirekler, Afsar Saranli:
A study on improving decisions in closed set speaker identification. 1127-1130 - Bojan Imperl, Zdravko Kacic, Bogomir Horvat:
The use of harmonic features in speaker recognition. 1131-1134 - Vlasta Radová, Josef Psutka:
An approach to speaker identification using multiple classifiers. 1135-1138
Spoken Language Systems
- Jorge Alvarez, Daniel Tapias, Carlos Crespo, Ismael Cortázar, Fernando Martínez:
Development and evaluation of the ATOS spontaneous speech conversational system. 1139-1142 - Giuseppe Riccardi, Allen L. Gorin, Andrej Ljolje, Michael Riley:
A spoken language system for automated call routing. 1143-1146 - Dario Albesano, Paolo Baggia, Morena Danieli, Roberto Gemello, Elisabetta Gerbino, Claudio Rullent:
Dialogos: a robust system for human-machine spoken dialogue on the telephone. 1147-1150 - Kazuhiro Kondo, Charles T. Hemphill:
Surfin' the World Wide Web with Japanese. 1151-1154 - Lee-Feng Chien, Sung-Chien Lin, Jenn-Chau Hong, Ming-Chiuan Chen, Hsin-Min Wang, Jia-Lin Shen, Keh-Jiann Chen, Lin-Shan Lee:
Internet Chinese information retrieval using unconstrained Mandarin speech queries based on a client-server architecture and a PAT-tree-based language model. 1155-1158 - Tatsuya Kawahara, Chin-Hui Lee, Biing-Hwang Juang:
Combining key-phrase detection and subword-based verification for flexible speech understanding. 1159-1162 - Holger Stahl, Johannes Müller, Manfred K. Lang:
Controlling limited-domain applications by probabilistic semantic decoding of natural speech. 1163-1166
Speech Enhancement
- Jörg Meyer, Klaus Uwe Simmer:
Multi-channel speech enhancement in a car environment using Wiener filtering and spectral subtraction. 1167-1170 - Néstor Becerra Yoma, Fergus R. McInnes, Mervyn A. Jack:
Weighted matching algorithms and reliability in noise cancelling by spectral subtraction. 1171-1174 - Michael E. Deisher, Andreas S. Spanias:
HMM-based speech enhancement using harmonic modeling. 1175-1178 - Bruce L. McKinley, Gary H. Whipple:
Model based speech pause detection. 1179-1182 - Andrzej Drygajlo, Benito Carnero:
Integrated speech enhancement and coding in the time-frequency domain. 1183-1186 - Cheung-Fat Chan, Wai-Kwong Hui:
Quality enhancement of narrowband CELP-coded speech via wideband harmonic re-synthesis. 1187-1190 - Futoshi Asano, Satoru Hayamizu:
Speech enhancement using CSS-based array processing. 1191-1194 - Daniel S. Benincasa, Michael I. Savic:
Co-channel speaker separation using constrained nonlinear optimization. 1195-1198 - Te-Won Lee, Reinhold Orglmeister:
A contextual blind separation of delayed and convolved sources. 1199-1202 - Georg F. Meyer, Fabrice Plante, Frédéric Berthommier:
Segregation of concurrent speech with the reassigned spectrum. 1203-1206 - Hector R. Javkin, Michael Galler, Nancy Niedzielski:
Enhancement of esophageal speech by injection noise rejection. 1207-1210 - Neeraj Magotra, Sudheer Sirivara:
Real-time digital speech processing strategies for the hearing impaired. 1211-1214 - Sharon Gannot, David Burshtein, Ehud Weinstein:
Iterative-batch and sequential algorithms for single microphone speech enhancement. 1215-1218 - Patrick Sörqvist, Peter Händel, Björn E. Ottersten:
Kalman filtering for low distortion speech enhancement in mobile communication. 1219-1222
Features for ASR
- Klaus Kasper, Herbert Reininger, Dietrich Wolf:
Exploiting the potential of auditory preprocessing for robust speech recognition by locally recurrent neural networks. 1223-1226 - Tai-Hwei Hwang, Lee-Min Lee, Hsiao-Chuan Wang:
Feature adaptation using deviation vector for robust speech recognition in noisy environment. 1227-1230 - Keith I. Francis, Timothy R. Anderson:
Binaural phoneme recognition using the auditory image model and cross-correlation. 1231-1234 - Daniel J. Mashao, John E. Adcock:
Utterance dependent parametric warping for a talker-independent HMM-based recognizer. 1235-1238 - Johan de Veth, Louis Boves:
Phase-corrected RASTA for automatic speech recognition over the phone. 1239-1242 - Shoji Kajita, Kazuya Takeda, Fumitada Itakura:
A binaural speech processing method using subband-cross correlation analysis for noise robust recognition. 1243-1246 - Michael J. Tomlinson, Martin J. Russell, Roger K. Moore, Andrew P. Buckland, Martin A. Fawley:
Modelling asynchrony in speech using elementary single-signal decomposition. 1247-1250 - Hervé Bourlard, Stéphane Dupont:
Subband-based speech recognition. 1251-1254 - Sangita Tibrewala, Hynek Hermansky:
Sub-band based recognition of noisy speech. 1255-1258 - Brian Kingsbury, Nelson Morgan:
Recognizing reverberant speech with RASTA-PLP. 1259-1262 - Saeed Vaseghi, Naomi Harte, Ben Milner:
Multi-resolution phonetic/segmental features and models for HMM-based speech recognition. 1263-1266 - Javier Hernando:
Maximum likelihood weighting of dynamic speech features for CDHMM speech recognition. 1267-1270 - Richard C. Rose, Eduardo Lleida:
Speech recognition using automatically derived acoustic baseforms. 1271-1274 - Alexandros Potamianos, Richard C. Rose:
On combining frequency warping and spectral shaping in HMM based speech recognition. 1275-1278
Speech Analysis
- John R. Deller Jr., Tsung-Ming Lin, Majid Nayeri:
Recursive linear prediction using OBE identification with automatic bound estimation. 1279-1282 - Martin Birgmeier, Hans-Peter Bernhard, Gernot Kubin:
Nonlinear long-term prediction of speech signals. 1283-1286 - Hywel B. Richards, John S. Bridle, Melvyn J. Hunt, John S. Mason:
Vocal tract shape trajectory estimation using MLP analysis-by-synthesis. 1287-1290 - Wen Ding, Nick Campbell, Norio Higuchi, Hideki Kasuya:
Fast and robust joint estimation of vocal tract and voice source parameters. 1291-1294 - Boris Doval, Christophe d'Alessandro:
Spectral correlates of glottal waveform models: an analytic study. 1295-1298 - Keiichi Funaki, Yoshikazu Miyanaga, Koji Tochinai:
A time varying ARMAX speech modeling with phase compensation using glottal source model. 1299-1302 - Hideki Kawahara:
Speech representation and transformation using adaptive interpolation of weighted spectrum: vocoder revisited. 1303-1306 - Dan Ellis:
The weft: a representation for periodic sounds. 1307-1310 - Markus Hauenstein:
A computationally efficient algorithm for calculating loudness patterns of narrowband speech. 1311-1314 - Ken'ichi Furuya, Yutaka Kaneda:
Two-channel blind deconvolution for non-minimum phase impulse responses. 1315-1318 - Sung-Joo Lee, Hee-Dong Kim, Hyung Soon Kim:
Variable time-scale modification of speech using transient information. 1319-1322 - Jong Won Seok, Keun-Sung Bae:
Speech enhancement with reduction of noise components in the wavelet domain. 1323-1326 - Jiangtao Xi, James P. Reilly:
Blind separation and restoration of signals mixed in convolutive environment. 1327-1330 - Eric D. Scheirer, Malcolm Slaney:
Construction and evaluation of a robust multifeature speech/music discriminator. 1331-1334
Topics in Speech Coding I
- Insung Lee, Hang Chae Woo:
Encoding of speech spectral parameters using adaptive quantization methods. 1335-1338 - Hai Le Vu, László Lois:
Optimal transformation of LSP parameters using neural network. 1339-1342 - Jan Cernocký, Geneviève Baudoin, Gérard Chollet:
Speech spectrum representation and coding using multigrams with distance. 1343-1346 - Ronald P. Cohn, John S. Collura:
Incorporating perception into LSF quantization some experiments. 1347-1350 - Jan Skoglund, Jan Linden:
Predictive VQ for noisy channel spectrum coding: AR or MA? 1351-1354 - Kazuhito Koishida, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai:
Efficient encoding of mel-generalized cepstrum for CELP coders. 1355-1358 - Juin-Hwey Chen:
A candidate coder for the ITU-T's new wideband speech coding standard. 1359-1362 - Benito Carnero, Andrzej Drygajlo:
Perceptual speech coding using time and frequency masking constraints. 1363-1366 - Anil Ubale, Allen Gersho:
A multi-band CELP wideband speech coder. 1367-1370 - Takehiro Moriya, Naoki Iwakami, Akio Jin, Kazunaga Ikeda, Satoshi Miki:
A design of transform coder for both speech and audio signals at 1 bit/sample. 1371-1374 - Kim T. Petersen, Steffen Duus Hansen, John Aasted Sørensen:
Speech quality assessment of compounded digital telecommunication systems; perceptual dimensions. 1375-1378 - Simão Ferraz de Campos Neto, Franklin L. Corcoran, Ara Karahisar:
Performance assessment of tandem connection of cellular and satellite-mobile coders. 1379-1382 - John J. Parry, Ian S. Burnett, Joe F. Chicharo:
The consequences of linguistic perception on low-rate speech coding. 1383-1386 - Martin Hansen, Birger Kollmeier:
Using a quantitative psychoacoustical signal representation for objective speech quality measurement. 1387-1390
Speech Models and Features
- Kazuyo Tanaka, Hiroaki Kojima:
A method of extracting time-varying acoustic features effective for speech recognition. 1391-1394 - Irina Illina, Yifan Gong:
Elimination of trajectory folding phenomenon: HMM, trajectory mixture HMM and mixture stochastic trajectory model. 1395-1398 - Wendy J. Holmes, Martin J. Russell:
Linear dynamic segmental HMMs: variability representation and training procedure. 1399-1402 - Toshiaki Fukada, Yoshinori Sagisaka, Kuldip K. Paliwal:
Model parameter estimation for mixture density polynomial segment models. 1403-1406 - Jan P. Verhasselt, Irina Illina, Jean-Pierre Martens, Yifan Gong, Jean-Paul Haton:
The importance of segmentation probability in segment based speech recognizers. 1407-1410 - Ashvin Kannan, Mari Ostendorf:
Adaptation of polynomial trajectory segment models for large vocabulary speech recognition. 1411-1414 - Chengalv Rathinavelu, Li Deng:
Speaker adaptation experiments using nonstationary-state hidden Markov models: a MAP approach. 1415-1418 - Kyuwoong Hwang:
Vocabulary optimization based on perplexity. 1419-1422 - Philippe Gelin, Christian Wellekens:
REMAP for video soundtrack indexing. 1423-1426 - Jinhai Cai, Zhi-Qiang Liu:
Robust pitch detection of speech signals using steerable filters. 1427-1430 - Tsuyoshi Moriyama, Hideo Saito, Shinji Ozawa:
Evaluation of the relationship between emotional concepts and emotional parameters on speech. 1431-1434 - Wolfgang Wokurek:
Time-frequency analysis of the glottal opening. 1435-1438 - Werner Kozek, Hans Georg Feichtinger:
Time-frequency structured decorrelation of speech signals via nonseparable Gabor frames. 1439-1442 - Filipp Korkmazskiy, Biing-Hwang Juang, Frank K. Soong:
Generalized mixture of HMMs for continuous speech recognition. 1443-1446
Topics in ASR
- Andrew W. Senior, Krishna S. Nathan:
Writer adaptation of a HMM handwriting recognition system. 1447-1450 - Udo Bub, Joachim Köhler, Bojan Imperl:
In-service adaptation of multilingual hidden-Markov-models. 1451-1454 - Vassilios Diakoloukas, Vassilios Digalakis, Leonardo Neumeyer, Jaan Kaja:
Development of dialect-specific speech recognizers using adaptation methods. 1455-1458 - Lin-Shan Lee, Bo-Ren Bai, Lee-Feng Chien:
Syllable-based relevance feedback techniques for Mandarin voice record retrieval using speech queries. 1459-1462 - Doroteo T. Toledano, Luis Villarrubia, Luis A. Hernández Gómez, Jose Maria Elvira:
Automatic alternative transcription generation and vocabulary selection for flexible word recognizers. 1463-1466 - Neeraj Deshmukh, Julie Ngan, Jonathan Hamaker, Joseph Picone:
An advanced system to generate pronunciations of proper nouns. 1467-1470 - Horacio Franco, Leonardo Neumeyer, Yoon Kim, Orith Ronen:
Automatic pronunciation scoring for language instruction. 1471-1474 - Coimbatore S. Ramalingam, Lorin Netsch, Yu-Hung Kao:
Speaker-independent name dialing with out-of-vocabulary rejection. 1475-1478 - Richard M. Schwartz, Scott Miller, David Stallard, John Makhoul:
Hidden understanding models for statistical sentence understanding. 1479-1482 - Frédéric Bimbot, Marc El-Bèze, Michèle Jardino:
An alternative scheme for perplexity estimation. 1483-1486 - Douglas B. Paul:
Extensions to phone-state decision-tree clustering: single tree and tagged clustering. 1487-1490 - Stéphane Lubiarz, Philip Lockwood:
Evaluation of fast algorithms for finding the nearest neighbor. 1491-1494 - Rudolf Kober, Ulrich Harz, Jutta Schiffers:
Fusion of visual and acoustic signals for command-word recognition. 1495-1497 - Yuri Iwano, Yosuke Sugita, Yusuke Kasahara, Shu Nakazato, Katsuhiko Shirai:
Difference in visual information between face to face and telephone dialogues. 1499-1502
Compensation (Speaker, Channel, Noise)
- Alain Biem, Shigeru Katagiri:
Cepstrum-based filter-bank design using discriminative feature extraction training at various levels. 1503-1506 - Wu Chou:
Minimum error rate training for designing tree-structured probability density function. 1507-1510 - Hiroshi Matsumoto, Masanori Ono:
A frequency-weighted HMM based on minimum error classification for noisy speech recognition. 1511-1514 - Daniel Willett, Christoph Neukirchen, Jörg Rottland:
Dictionary-based discriminative HMM parameter estimation for continuous speech recognition systems. 1515-1518 - Ángel de la Torre, Antonio M. Peinado, Antonio J. Rubio, Victoria E. Sánchez:
A DFE-based algorithm for feature selection in speech recognition. 1519-1522 - Vijay Raman, Vidhya Ramanujam:
Robustness issues and solutions in speech recognition based telephony services. 1523-1526 - Vincent Fontaine, Hervé Bourlard:
Speaker-dependent speech recognition based on phone-like units models-application to voice dialling. 1527-1530 - Josef G. Bauer:
Enhanced control and estimation of parameters for a telephone based isolated digit recognizer. 1531-1534 - Douglas A. Reynolds:
HTIMIT and LLHDB: speech corpora for the study of handset transducer effects. 1535-1538 - Michael Galler, Jean-Claude Junqua:
Robustness improvements in continuously spelled names over the telephone. 1539-1542 - Qi Li, Sarangarajan Parthasarathy, Aaron E. Rosenberg:
A fast algorithm for stochastic matching with application to robust speaker verification. 1543-1546 - Qiang Huo, Hui Jiang, Chin-Hui Lee:
A Bayesian predictive classification approach to robust speech recognition. 1547-1550 - Hui Jiang, Keikichi Hirose, Qiang Huo:
Robust speech recognition based on Viterbi Bayesian predictive classification. 1551-1554
Speech Coding at Low Bit Rate
- Charalampos Papanastasiou, Costas S. Xydeas:
Efficient mixed excitation models in LPC based prototype interpolation speech coders. 1555-1558 - Ian A. Atkinson, Suat Yeldener, Ahmet M. Kondoz:
High quality split band LPC vocoder operating at low bit rates. 1559-1562 - Hui Li, Gordon B. Lockhart:
Non-linear techniques for pitch and waveform enhancement in PWI coders. 1563-1566 - Ian S. Burnett, Duong H. Pham:
Multi-prototype waveform coding using frame-by-frame analysis-by-synthesis. 1567-1570 - Khashayar Yaghmaie, Ahmet M. Kondoz:
Multiband prototype waveform analysis synthesis for very low bit rate speech coding. 1571-1574 - Parham Zolfaghari, Tony Robinson:
A formant vocoder based on mixtures of Gaussians. 1575-1578 - Engin Erzin, Arun Kumar, Allen Gersho:
Natural quality variable-rate spectral speech coding below 3.0 kbps. 1579-1582 - Yusuke Hiwasaki, Kazunori Mano:
A new 2-kbit/s speech coder based on normalized pitch waveform. 1583-1586 - Mary A. Kohler:
A comparison of the new 2400 bps MELP Federal Standard with other standard coders. 1587-1590 - Lynn M. Supplee, Ronald P. Cohn, John S. Collura, Alan McCree:
MELP: the new Federal Standard at 2400 bps. 1591-1594 - Jes Thyssen, W. Bastiaan Kleijn, Roar Hagen:
Using a perception-based frequency scale in waveform interpolation. 1595-1598 - Yair Shoham:
Very low complexity interpolative speech coding at 1.2 to 2.4 kbps. 1599-1602 - Michele L. Jamrozik, John N. Gowdy:
Modified multiband excitation model at 2400 bps. 1603-1606 - Eric W. M. Yu, Cheung-Fat Chan:
Variable bit rate MBELP speech coding via V/UV distribution dependent spectral quantization. 1607-1610
Volume 3: Speech Processing, Digital Signal Processing
Topics in Speech Analysis/Synthesis/Production
- Takashi Masuko, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai:
Voice characteristics conversion for HMM-based speech synthesis system. 1611-1614 - Gudrun Klasmeyer:
The perceptual importance of selected voice quality parameters. 1615-1618 - Hani Yehia, Mark Tiede:
A parametric three-dimensional model of the vocal-tract based on MRI data. 1619-1622 - Rashid Ansari:
Inverse filter approach to pitch modification: application to concatenative synthesis of female speech. 1623-1626 - Helen M. Hanson:
Vowel amplitude variation during sentence production. 1627-1630 - Dongbing Wei, Colin C. Goodyear:
Experiments in female voice speech synthesis using a parametric articulatory model. 1631-1634 - Andrew Richard Greenwood:
Articulatory speech synthesis using diphone units. 1635-1638 - David T. Chappell, John H. L. Hansen:
An auditory-based measure for improved phone segment concatenation. 1639-1642 - Douglas J. Nelson:
Correlation based speech formant recovery. 1643-1646 - Steven Greenberg, Brian Kingsbury:
The modulation spectrogram: in pursuit of an invariant representation of speech. 1647-1650 - François Pellegrino, Régine André-Obrecht:
From vocalic detection to automatic emergence of vowel systems. 1651-1654 - David van Kuijk, Louis Boves:
Acoustic characteristics of lexical stress in continuous speech. 1655-1658 - Minsheng Liu, Arild Lacroix:
Pole-zero modeling of vocal tract for fricative sounds. 1659-1662 - Thomas Wittenberg, Patrick Mergell, Monika Tigges, Ulrich Eysholdt:
Quantitative characterization of functional voice disorders using motion analysis of high-speed video and modeling. 1663-1666
Topics in Speech Coding II
- Tim Fingscheidt, Peter Vary:
Robust speech decoding: a universal approach to bit error concealment. 1667-1670 - W. Bastiaan Kleijn:
On optimal and minimum-entropy decoding. 1671-1674 - Sassan Ahmadi, Andreas S. Spanias:
A new sinusoidal phase modeling algorithm. 1675-1678 - Kazuo Nakata, Kin-ichi Higure:
Recursive and adaptive predictive coding of speech. 1679-1682 - Takahiro Unno, Thomas P. Barnwell III, Mark A. Clements:
The multimodal multipulse excitation vocoder. 1683-1686 - Manohar N. Murthi, Bhaskar D. Rao:
Minimum variance distortionless response (MVDR) modeling of voiced speech. 1687-1690 - Xiaoqin Sun, Fabrice Plante, Barry M. G. Cheetham, Kenneth W. T. Wong:
Phase modelling of speech excitation for low bit-rate sinusoidal transform coding. 1691-1694 - John E. Kleider, William M. Campbell:
An adaptive-rate digital communication system for speech. 1695-1698 - Mohammad Zad-issa, Peter Kabal:
Smoothing the evolution of the spectral parameters in linear prediction of speech using target matching. 1699-1702 - Shahrokh Ghaemmaghami, Mohamed A. Deriche, Boualem Boashash:
Comparative study of different parameters for temporal decomposition based speech coding. 1703-1706 - Sara Grassi, Alain Dufaux, Michael Ansorge, Fausto Pellandini:
Efficient algorithm to compute LSP parameters from 10th-order LPC coefficients. 1707-1710 - John Leis, Mark Phythian, Sridha Sridharan:
Speech compression with preservation of speaker identity. 1711-1714 - Juha Backman:
A method for measuring modulation transmission in speech transmitted via a nonlinear channel. 1715-1718 - Vincent van de Laar, W. Bastiaan Kleijn, Ed F. Deprettere:
Perceptual entropy rate estimates for the phonemes of American English. 1719-1722
Acoustic Modeling
- Olivier Oppizzi, Régis Quélavoine:
Rescoring under fuzzy measures with a multilayer neural network in a rule-based speech recognition system. 1723-1726 - Chak-Wai Chau, Sam Kwong, C. K. Diu, Wolfgang R. Fahrner:
Optimization of HMM by a genetic algorithm. 1727-1730 - Sabine Deligne, Frédéric Bimbot:
Inference of variable-length acoustic units for continuous speech recognition. 1731-1734 - Bojan Petek, Ove Andersen, Paul Dalsgaard:
Comparative performance analysis of statistical trajectory models in cellular environment. 1735-1738 - Yu-Hung Kao, Lorin Netsch:
Inter-digit HMM: connected digit recognition using the Macrophone corpus. 1739-1742 - Michael Finke, Ivica Rogina:
Wide context acoustic modeling in read vs. spontaneous speech. 1743-1746 - Jörg Rottland, Christoph Neukirchen, Daniel Willett:
Performance of hybrid MMI-connectionist/HMM systems on the WSJ speech database. 1747-1750 - Don X. Sun:
Statistical modeling of co-articulation in continuous speech based on data driven interpolation. 1751-1754 - John J. Godfrey, Aravind Ganapathiraju, Coimbatore S. Ramalingam, Joseph Picone:
Microsegment-based connected digit recognition. 1755-1758 - Jürgen Fritsch, Michael Finke, Alex Waibel:
Context-dependent hybrid HME/HMM speech recognition using polyphone clustering decision trees. 1759-1762 - Knut Kvale, Ingunn Amdal:
Improved automatic recognition of Norwegian natural numbers by incorporating phonetic knowledge. 1763-1766 - Stéphane Dupont, Hervé Bourlard, Olivier Deroo, Vincent Fontaine, Jean-Marc Boite:
Hybrid HMM/ANN systems for training independent tasks: experiments on Phonebook and related improvements. 1767-1770 - Harald Höge, Herbert S. Tropf, Richard Winski, Henk van den Heuvel, Reinhold Haeb-Umbach, Khalid Choukri:
European speech databases for telephone applications. 1771-1774 - Pak-Chung Ching, Ka-Fai Chow, Tan Lee, Alfred Ying Pang Ng, Lai-Wan Chan:
Development of a large vocabulary speech database for Cantonese. 1775-1778
Large Vocabulary Systems and Implementation
- Qiru Zhou, Wu Chou:
An approach to continuous speech recognition based on layered self-adjusting decoding graph. 1779-1782 - Stefan Ortmanns, Andreas Eiden, Hermann Ney, Norbert Coenen:
Look-ahead techniques for fast beam search. 1783-1786 - Ken Hanazawa, Yasuhiro Minami, Sadaoki Furui:
An efficient search method for large-vocabulary continuous-speech recognition. 1787-1790 - Hermann Ney, Stefan Ortmanns, Ingo Lindam:
Extensions to the word graph method for large vocabulary continuous speech recognition. 1791-1794 - Sarvar Patel:
An O(N√E¯) Viterbi algorithm. 1795-1798 - Tung-Hui Chiang, Chung-Mou Pengwu, Shih-Chieh Chien, Chao-Huang Chang:
CCLMDS'96: towards a speaker-independent large-vocabulary Mandarin dictation system. 1799-1802 - Tatsuo Matsuoka, Katsutoshi Ohtsuki, Takeshi Mori, Kotaro Yoshida, Sadaoki Furui, Katsuhiko Shirai:
Japanese large-vocabulary continuous-speech recognition using a business-newspaper corpus. 1803-1806 - Meinrad Niemöller, Alfred Hauenstein, Erwin Marschall, Petra Witschel, Ulrike Harke:
A PC-based real-time large vocabulary continuous speech recognizer for German. 1807-1810 - Barbara Peskin, Larry Gillick, Natalie Liberman, Michael Newman, Paul van Mulbregt, Steven Wegmann:
Progress in recognizing conversational telephone speech. 1811-1814 - Torsten Zeppenfeld, Michael Finke, Klaus Ries, Martin Westphal, Alex Waibel:
Recognition of conversational telephone speech using the JANUS speech engine. 1815-1818 - Roland Kuhn, Peter Nowell, Caroline Drouin:
Approaches to phoneme-based topic spotting: an experimental comparison. 1819-1822 - Philip N. Garner, Aidan Hemsworth:
A keyword selection strategy for dialogue move recognition and multi-class topic identification. 1823-1826 - Seong-Jin Yun, Yung-Hwan Oh, Gyung-Chul Shin:
Improved lexicon modeling for continuous speech recognition. 1827-1830
Signal Reconstruction
- José M. N. Vieira, Paulo Jorge S. G. Ferreira:
Interpolation, spectrum analysis, error-control coding, and fault-tolerant computing. 1831-1834 - Yannick Deville, Nabil Charkani:
Analysis of the stability of time-domain source separation algorithms for convolutively mixed signals. 1835-1838 - Alban Duverdier, Bernard Lacaze:
Time-varying reconstruction of stationary processes subjected to analogue periodic scrambling. 1839-1842 - Miroslaw Pawlak, Ulrich Stadtmüller:
Signal recovery from grouped data. 1843-1844 - Peter A. Hoeher, Stefan Kaiser, Patrick Robertson:
Two-dimensional pilot-symbol-aided channel estimation by Wiener filtering. 1845-1848 - Howard Hua Yang, Shun-ichi Amari:
Blind equalization of switching channels by ICA and learning of learning rate. 1849-1852 - Buyurman Baykal, Oguz Tanrikulu, Jonathon A. Chambers:
Adaptive soft-constraint satisfaction (SCS) algorithms for fractionally-spaced blind equalizers. 1853-1856 - Akram Aldroubi, Hans Georg Feichtinger:
Complete iterative reconstruction algorithms for irregularly sampled data in spline-like spaces. 1857-1860 - A. Sony John, Uday B. Desai:
Signal de-noising using the wavelet transform and regularization. 1861-1864 - Stephen D. Casey, Carlos Alberto Berenstein, David Francis Walnut:
Exact multichannel deconvolution on radial domains. 1865-1868 - Haralambos Pozidis, Athina P. Petropulu:
Signal reconstruction from phase only information and application to blind system estimation. 1869-1872
DSP Applications
- Thomas M. Panicker, V. John Mathews:
A fast Gauss-Newton parallel-cascade adaptive truncated Volterra filter. 1873-1876 - Alberto Carini, V. John Mathews, Giovanni L. Sicuranza:
Sufficient stability bounds for slowly varying discrete-time recursive linear filters. 1877-1880 - Brian S. Krongold, Michael L. Kramer, Kannan Ramchandran, Douglas L. Jones:
Spread spectrum interference suppression using adaptive time-frequency tilings. 1881-1884 - Paulo A. C. Marques, Fernando Manuel Gomes de Sousa, José M. N. Leitão:
A DSP based long distance echo canceller using short length centered adaptive filters. 1885-1888 - Brian M. Sadler, Laurel C. Sadler, Tien Pham:
Optimal and robust shockwave detection and estimation. 1889-1892 - Gopal T. Venkatesan, Dennis West, Kevin M. Buckley, Ahmed H. Tewfik, Mostafa Kaveh:
Automatic fault monitoring using acoustic emissions. 1893-1896 - Hassan Ezzaidi, Ivan Bourmeyster, Jean Rouat:
A new algorithm for double talk detection and separation in the context of digital mobile radio telephone. 1897-1900 - Supratim Saha, Angarai Ganesan Ramakrishnan:
Transmission of chosen transform coefficients of normalized cardiac beats for compression. 1901-1904 - Ba-Ngu Vo, Thi-Ngoc Ho, Antonio Cantoni, Victor Sreeram:
FIR filters in continuous-time envelope constrained filter design. 1905-1908 - Matteo Bertocco, Dennis Lorenzin, Pietro Paglierani:
Modified cepstral analysis for accurate estimation of echo parameters in telecommunication networks. 1909-1912 - Olivier Meste:
Bispectral reconstruction using incomplete phase knowledge: a neuroelectric signal estimation application. 1913-1916 - Gustavo A. Hirchoren, Dalton Soares Arantes:
Optimal phase-locked loop design with Kalman predictors for synchronous networks. 1917-1920
Adaptive Filters
- Albertus C. den Brinker:
Overparametrization in adaptive filters. 1921-1924 - Petr Tichavský, Peter Händel:
Recursive estimation of linearly or harmonically modulated frequencies of multiple cisoids in noise. 1925-1928 - Tyseer Aboulnasr, Khaled A. Mayyas:
Selective coefficient update of gradient-based adaptive algorithms. 1929-1932 - Quanhong Zhu, Scott C. Douglas, Kent F. Smith:
A pipelined architecture for LMS adaptive FIR filters without adaptation delay. 1933-1936 - Markus Rupp, Scott C. Douglas:
Deterministic stabilty analyses of unit-norm constrained algorithms for unbiased adaptive IIR filtering. 1937-1940 - Parthapratim De, H. Howard Fan:
A modified normalized lattice adaptive filter for fast sampling. 1941-1944 - John S. Bodenschatz:
Symmetric alpha-stable filter theory. 1945-1948 - Owen E. Kelly, Don H. Johnson:
Adaptive channel equalization using context trees. 1949-1952 - Michael L. McCloud, Delores M. Etter:
Subband adaptive filtering with time-varying nonuniform filter banks. 1953-1956 - Sofia Ben Jebara, Meriem Jaïdane-Saïdane:
Best input for optimal tracking randomly time-varying systems: justification of adaptive predictive structure. 1957-1960 - Saul B. Gelfand, Yongbin Wei, James V. Krogmeier:
Stability of variable and random stepsize LMS. 1961-1963
Fast Algorithms
- Erich Fuchs, Klaus Donner:
Fast least-squares polynomial approximation in moving time windows. 1965-1968 - Yi Chu, Wen-Hsien Fang, Shun-Hsyung Chang:
An efficient Haar wavelet-based approach for the harmonic retrieval problem. 1969-1972 - Haitao Guo, C. Sidney Burrus:
Wavelet transform based fast approximate Fourier transform. 1973-1976 - Dragutin Sevic, Miodrag V. Popovic:
On computing the 2-D extended lapped transforms. 1977-1980 - Isabel García, Consuelo Gonzalo, Margarita Pérez-Castellanos, José A. Moreno, José Sánchez-Dehesa:
Efficient computation of the discrete Wigner distribution function through a new iterative algorithm. 1981-1984 - Joseph M. Winograd, S. Hamid Nawab:
Probabilistic complexity analysis of incremental DFT algorithms. 1985-1988 - Cuong Pham, Tokunbo Ogunfunmi:
On the recursive total least-squares. 1989-1992 - Richard J. Kozick, Maurice F. Aburdene:
Parallel-recursive filter structures for the computation of discrete transforms. 1993-1996 - Andreas Klappenecker:
Basefield transforms derived from character tables. 1997-2000 - Gábor Péceli, Annamária R. Várkonyi-Kóczy:
Block-recursive filters and filter-banks. 2001-2004 - Abdulnasir Hossen, Ulrich Heute:
Fast approximate DCT: basic-idea, error analysis, applications. 2005-2008 - Annamária R. Várkonyi-Kóczy, Sergios Theodoridis:
Fast sliding transforms in transform-domain adaptive filtering. 2009-2012
Time-Frequency and Wavelets I
- Hiroshi Kanai, Michie Sato, Noriyoshi Chubachi:
Time dependent autoregressive spectrum estimation of heart wall vibrations. 2013-2016 - Jakob Ängeby:
Properties of the structured auto-regressive time-frequency distribution. 2017-2020 - Chenshu Wang, Moeness G. Amin:
Zero-tracking time-frequency distributions. 2021-2024 - Daniel Seidner, Meir Feder:
Vector sampling expansion: deterministic and stochastic signals. 2025-2028 - Paolo Prandoni, Michael M. Goodwin, Martin Vetterli:
Optimal time segmentation for signal modeling and compression. 2029-2032 - Michael J. Vrhel, Akram Aldroubi:
Pre-filtering for the initialization of multi-wavelet transforms. 2033-2036 - Michael M. Goodwin:
Matching pursuit with damped sinusoids. 2037-2040 - Antonia Papandreou-Suppappola, Robin L. Murray, Gloria Faye Boudreaux-Bartels:
Localized subclasses of quadratic time-frequency representations. 2041-2044 - Jack McLaughlin, James Droppo, Les E. Atlas:
Class-dependent, discrete time-frequency distributions via operator theory. 2045-2048 - Franz Hlawatsch, Teresa Twaroch:
Extending the characteristic function method for joint a-b and time-frequency analysis. 2049-2052 - Ljubisa Stankovic, Srdjan Stankovic, Igor Djurovic:
An architecture for realization of the cross-terms free polynomial Wigner-Ville distribution. 2053-2056 - Michael S. Richman, Thomas W. Parks:
Understanding discrete rotations. 2057-2060
Design of RNS Frequency Sampling Filter Banks
- Uwe Meyer-Bäse, Jon Mellott, Fred J. Taylor:
Design of RNS frequency sampling filter banks. 2061-2064 - William M. Campbell, Thomas W. Parks:
Optimal design of multirate systems with constraints. 2065-2068 - Wayne Lawton, Charles A. Micchelli:
Design of conjugate quadrature filters having specified zeros. 2069-2072 - Jörg Kliewer, Alfred Mertins:
Design of paraunitary oversampled cosine-modulated filter banks. 2073-2076 - Masaaki Ikehara, Truong Q. Nguyen:
Time-domain design of linear-phase PR filter banks. 2077-2080 - Benjamin W. Wah, Yi Shang, Tao Wang, Ting Yu:
QMF filter bank design by a new global optimization method. 2081-2084 - Vijay K. Jain:
Unified approach to the design of quadrature-mirror filters. 2085-2088
Time-Frequency and Wavelets II
- Heinrich Ruser, Martin Vossiek, Alexander von Jena, Valentin Mágori:
Inverse filter technique for high-precision ultrasonic pulsed wave range Doppler sensors. 2089-2092 - Christoph Delfs, Friedrich K. Jondral:
Classification of piano sounds using time-frequency signal analysis. 2093-2096 - John G. Apostolopoulos, Jae S. Lim:
Transform/subband representations for signals with arbitrarily shaped regions of support. 2097-2100 - Martin J. Bastiaans:
On optimum oversampling in the Gabor scheme. 2101-2104 - Gianpaolo Evangelista, Sergio Cavaliere:
The discrete-time frequency warped wavelet transforms. 2105-2108 - Arvid Breitenbach:
Aspects of spectrum and hybrid spectrum analysis for sensor SNR determination. 2109-2112 - Michael Unser, Josiane Zerubia:
Generalized sampling without bandlimiting constraints. 2113-2116 - Jaakko Astola, Karen O. Egiazarian, Heikki Huttunen:
Wavelet packets and genetic algorithms. 2117-2120 - Xiang-Gen Xia, Shie Qian:
An iterative algorithm for time-variant filtering in the discrete Gabor transform domain. 2121-2124 - Patrick J. Loughlin, Dale Groutage, Robert Rohrbaugh:
Time-frequency analysis of acoustic transients. 2125-2128 - Mark Allan Coffey:
Boundary-compensated wavelet bases. 2129-2132 - Israel Cohen, Shalom Raz, David Malah:
Eliminating interference terms in the Wigner distribution using extended libraries of bases. 2133-2136 - Antonio S. Pena, Nuria González-Prelcic, Carlos A. Serantes:
A flexible tiling of the time axis for adaptive wavelet packet decompositions. 2137-2140 - Mehran Jahed, Bijan Najafi, Ali Khamene, Stephen J. Lai-Fook:
Time delay calculation of stress waves using wavelet analysis application in canine edematous lungs. 2141-2144
Digital Filter Design and Implementation
- Alexander Flaig, Gonzalo R. Arce, Kenneth E. Barner:
Affine order statistic filters: a data-adaptive filtering framework for nonstationary signals. 2145-2148 - Dusan M. Kodek:
Limits of finite wordlength FIR digital filter design. 2149-2152 - Guo-Fang Xu, Tamal Bose, Jim Schroeder:
Elimination of limit cycles due to two's complement quantization in normal form digital filters. 2153-2156 - Lina J. Karam:
On the design of multidimensional FIR filters by transformation. 2157-2160 - Hartmut Brandenstein, Rolf Unbehauen:
Approximation of complex-valued 2-D frequency response specifications by separable-denominator digital filters. 2161-2164 - Li Lee, Alan V. Oppenheim:
Properties of approximate Parks-McClellan filters. 2165-2168 - Mathias C. Lang:
Design of nonlinear phase FIR digital filters using quadratic programming. 2169-2172 - Richard Rau, James H. McClellan:
Design of polar-separable FIR filters by radial slice approximations. 2173-2176 - Juha Kauraniemi:
Analysis of limit cycles in the direct form delta operator structure by computer-aided test. 2177-2180 - Yong Ching Lim, Seo-How Low:
The synthesis of sharp diamond-shaped filters using the frequency response masking approach. 2181-2184 - Mitsuhiko Yagyu, Akinori Nishihara, Nobuo Fujii:
Minimization of finite wordlength error in 2-D FIR digital filters in the frequency domain. 2185-2188 - José L. Sanz-González:
Tradeoff between roundoff and overflow errors in digital filter realizations. 2189-2192 - Rui Yang, Yong Ching Lim, Maurice G. Bellanger:
Design of sharp FIR bandstop filters using quadrature masking filters. 2193-2196 - Kamen R. Ralev, Peter H. Bauer:
Implementation options for block floating point digital filters. 2197-2200
DSP: Teaching, IIR Filters, Fractional Sampling
- Ljiljana D. Milic, Miroslav D. Lutovac:
Design of multiplierless elliptic IIR filters. 2201-2204 - Matti Karjalainen, Aki Härmä, Unto K. Laine:
Realizable warped IIR filters and their properties. 2205-2208 - Ivan W. Selesnick:
New exchange rules for IIR filter design. 2209-2212 - Artur Krukowski, Izzet Kale, Richard C. S. Morling:
The design of polyphase-based IIR multiband filters. 2213-2216 - Zhongnong Jiang, Alan N. Willson Jr.:
A pipelined/interleaved IIR digital filter architecture. 2217-2220 - Bojan Djokic, Miroslav D. Lutovac, Miodrag V. Popovic:
A new approach to the phase error and THD improvement in linear phase IIR filters. 2221-2224 - Ashraf Alkhairy:
Design of recursive digital filters with magnitude specifications. 2225-2227 - Ahmet Kirac, Palghat P. Vaidyanathan:
FIR compaction filters: new design methods and properties. 2229-2232 - Anush Yardim, Gerald D. Cain, Arnaud Lavergne:
Performance of fractional-delay filters using optimal offset windows. 2233-2236 - Andrzej Tarczynski, Vesa Välimäki, Gerald D. Cain:
FIR filtering of nonuniformly sampled signals. 2237-2240 - Manuel Duarte Ortigueira:
Fractional discrete-time linear systems. 2241-2244 - N. Paul Murphy, Artur Krukowski, Andrzej Tarczynski:
An efficient fractional sample delayer for digital beam steering. 2245-2248 - Chaouki T. Abdallah, Dalton Soares Arantes, Gregory L. Heileman, Don R. Hush, Ramiro Jordan, Roberto de Alencar Lotufo, Neeraj Magotra, L. Howard Pollard, Edl Schamiloglu, Robert Whitman:
Interactive DSP course development/teaching environment. 2249-2252 - Martti Rahkila, Matti Karjalainen:
An interactive DSP tutorial on the Web. 2253-2256
Novel Adaptive Algorithms
- Daniël W. E. Schobben, Gerard P. M. Egelmeers, Piet C. W. Sommen:
Efficient realization of the block frequency domain adaptive filter. 2257-2260 - Woo-Jin Song, Min-Soo Park:
A complementary pair LMS algorithm for adaptive filtering. 2261-2264 - Marc Moonen, Ian K. Proudler:
Using a lattice algorithm to estimate the Kalman gain vector in fast Newton-type adaptive filtering. 2265-2268 - João Batista Destro Filho, Gérard Favier, João M. T. Romano:
New Bussgang methods for blind equalization. 2269-2272 - Victor Shtrom, H. Howard Fan:
A refined class of cost functions in blind equalization. 2273-2276 - Majid Nayeri, T. M. Lin, John R. Deller Jr.:
Novel blind variants of the OBE algorithm. 2277-2280 - Pi Sheng Chang, Alan N. Willson Jr.:
Conjugate gradient method for adaptive direction-of-arrival estimation of coherent signals. 2281-2284 - Phillip M. S. Burt, Max Gerken:
A polyphase IIR adaptive filter: error surface analysis and application. 2285-2288 - J. William Whikehart, Soura Dasgupta:
Adaptive periodic IIR filters. 2289-2292 - Fernando Gil Vianna Resende Jr., Paulo S. R. Diniz, Mineo Kaneko, Akinori Nishihara:
Adaptive AR spectral estimation based on multi-band decomposition of the linear prediction error with variable forgetting factors. 2293-2296 - François Capman, Jérôme Boudy, Philip Lockwood:
Controlled convergence of QR least-squares adaptive algorithms-application to speech echo cancellation. 2297-2300 - Laurence Tianruo Yang, Man Lin:
Iterative total least squares filter in robot navigation. 2301-2304 - Shigenori Kinjo, Mirai Oshiro, Hiroshi Ochi:
A new two-dimensional parallel block adaptive filter with reduced computational complexity. 2305-2308 - Yoshito Higa, Hiroshi Ochi, Shigenori Kinjo:
An over-sampling subband adaptive filter with the optimal real filter bank. 2309-2312
Adaptive Algorithms - Analytical Models
- Azzedine Zerguine, Maamar Bettayeb, Colin F. N. Cowan:
A hybrid LMS-LMF scheme for echo cancellation. 2313-2316 - Tomas Gänsler:
A robust frequency-domain echo canceller. 2317-2320 - Ling Qin, Maurice G. Bellanger:
Adaptive sub-channel equalization in multicarrier transmission. 2321-2324 - Kensaku Fujii, Juro Ohga:
Sub-RLS algorithm with an extremely simple update equation. 2325-2328 - Noriyuki Hirayama, Hideaki Sakai:
Analysis of a delayless subband adaptive filter. 2329-2332 - Shin'ichi Koike:
A new efficient method of convergence calculation for adaptive filters using the sign algorithm with digital data inputs. 2333-2336 - Maria D. Miranda, Leonardo Aguayo, Max Gerken:
Performance of the a priori and a posteriori QR-LSL algorithms in a limited precision environment. 2337-2340 - Kamal Premaratne, Mohamed Mansour:
Robust stability of time-variant difference equations with restricted parameter perturbations: regions in coefficient-space. 2341-2344 - Ronald D. DeGroat, Dinko Begusic, Eric M. Dowling, Darel A. Linebarger:
Spherical subspace and eigen based affine projection algorithms. 2345-2348 - Sau-Gee Chen, Yung-An Kao, Ching-Yeu Chen:
On the convergence and MSE of Chen's LMS adaptive algorithm. 2349-2352 - Yongbin Wei, Saul B. Gelfand, James V. Krogmeier:
Noise constrained LMS algorithm. 2353-2356 - Shivaling S. Mahant-Shetti, Srinath Hosur, Alan Gatherer:
The log-log LMS algorithm. 2357-2360 - J. Jiang, Christopher D. Schmitz, Bernard A. Schnaufer, W. Kenneth Jenkins:
Improved fault coverage for adaptive fault tolerant filters. 2361-2364
Nonlinear Systems and Signal Analysis
- Alfredo Restrepo, Luis F. Zuluaga, Luis E. Pino:
Optimal noise levels for stochastic resonance. 2365-2368 - C. Emanuel Savin, M. Omair Ahmad, M. N. Shanmukha Swamy:
Lp norm design of weighted order statistic filters. 2369-2372 - Tania Stathaki, Anne Scohyers:
A constrained optimisation approach to the blind estimation of Volterra kernels. 2373-2376 - Olufemi Adeyemi, Gloria Faye Boudreaux-Bartels:
Improved accuracy in the singularity spectrum of multifractal chaotic time series. 2377-2380 - Hans Schurer, Alex G. J. Nijmeijer, Mark A. Boer, Cornelis H. Slump, Otto E. Herrmann:
Identification and compensation of the electrodynamic transducer nonlinearities. 2381-2384 - Robert D. Nowak, Richard G. Baraniuk:
Wavelet-based transformations for nonlinear signal processing. 2385-2388 - Elena Stringa, Carlo S. Regazzoni:
Signal restoration by statistical soft morphology. 2389-2392 - Ted Frison, Henry D. I. Abarbanel:
Identification and quantification of nonstationary chaotic behavior. 2393-2396 - Jean-Pierre Costa, Thierry Pitarque, Eric Thierry:
Using orthogonal least squares identification for adaptive nonlinear filtering of GSM signals. 2397-2400 - Daniel Homm, Rudolf Rabenstein:
Numerical integration of nonlinear multidimensional systems. 2401-2404 - Panos Koukoulas, Nicholas Kalouptsidis:
Third order Volterra system identification. 2405-2408 - Balasubramaniam Santhanam, Petros Maragos:
Demodulation of discrete multicomponent AM-FM signals using periodic algebraic separation and energy demodulation. 2409-2412 - Subhash Challa, Farhan A. Faruqi:
A new approach to optimal nonlinear filtering. 2413-2416 - Murali Tummala, Michael T. Donovan, Bruce E. Watkins, Richard North:
Volterra series based modeling and compensation of nonlinearities in high power amplifiers. 2417-2420
Filter Banks
- Henrique S. Malvar:
Lapped biorthogonal transforms for transform coding with reduced blocking and ringing artifacts. 2421-2424 - Jamal Tuqan, Palghat P. Vaidyanathan:
Optimum low cost two channel IIR orthonormal filter bank. 2425-2428 - Takayuki Nagai, Takaaki Futie, Masaaki Ikehara:
Direct design of nonuniform filter banks. 2429-2432 - Cormac Herley:
Reconstruction for novel sampling structures. 2433-2436 - Dong-Yan Huang, Phillip A. Regalia, Maurice G. Bellanger:
Comparison of two eigenstructure algorithms for lossless multirate filter optimization. 2437-2440 - Frank Hartenstein:
Parametrization of discrete finite biorthogonal wavelets with linear phase. 2441-2444 - Tanja Karp, Alfred Mertins, Truong Q. Nguyen:
Efficiently VLSI-realizable prototype filters for modulated filter banks. 2445-2448 - Palghat P. Vaidyanathan, Ahmet Kirac:
Theory of cyclic filter banks. 2449-2452 - Helmut Bölcskei, Franz Hlawatsch:
Oversampled filter banks: optimal noise shaping, design freedom, and noise analysis. 2453-2456 - Frank Heinle:
A new approach to the compensation of aliasing in transform and subband coders. 2457-2460 - Peter Rieder:
Parameterization of symmetric multiwavelets. 2461-2464 - Ricardo L. de Queiroz, Reiner Eschbach:
Downscaled inverses for M-channel lapped transforms. 2465-2468 - Gerald Schuller:
Time-varying filter banks with variable system delay. 2469-2472 - Jérôme Lebrun, Martin Vetterli:
Balanced multiwavelets. 2473-2476
DSP for Communications
- Michael G. Larimore, Sally L. Wood, John R. Treichler:
Performance costs for theoretical minimal-length equalizers. 2477-2480 - Patrick Grohan, Sylvie Marcos:
Nonlinear channel equalizer using Gaussian sum approximations. 2481-2484 - S. Tateesh, S. A. Atungsiri, Ahmet M. Kondoz:
Link adaptation to channel interference using multi-rate source and channel coding for CDMA mobile communications. 2485-2488 - Ahmad Bahai, Markus Rupp:
Adaptive DFE algorithms for IS-136 based TDMA cellular phones. 2489-2492 - Mohammed Al-Janabi, Izzet Kale, Richard C. S. Morling:
Effective-fourth-order resonator based MASH bandpass sigma-delta modulators. 2493-2496 - Tetsuya Shimamura, Colin F. N. Cowan:
Equalisation of time variant multipath channels using amplitude banded techniques. 2497-2500 - Walter A. Frank, Ulrich Appel:
Efficient equalization of nonlinear communication channels. 2501-2504