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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