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ORCIDICASSP 1986: Tokyo, Japan
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 1986, Tokyo, Japan, April 7-11, 1986. IEEE 1986
- K. Odaka, T. T. Doi, H. Nakajima:
Digital audio magnetic recording: Progress in packing density and key technology. 1-4 - Toshiyuki Suzuki, Takeharu Niioka:
Perpendicular magnetic recording and its application to digital recording. 5-8 - Kenji Hayashi, Takao Arai, Takaharu Noguchi, Hiroo Okamoto, Masaharu Kobayashi:
Error correction method for R-DAT and its evaluation. 9-12 - John P. Stautner, David M. Horowitz:
Efficient data reduction for digital audio using a digital filter array. 13-16 - Roger Lagadec:
Digital data equalizing in multitrack digital audio recording. 17-20 - Joël Soumagne, Philippe Mabilleau, Sarto Morissette, Gerald Chouinard, David Bennett:
A comparative study of the proposed high quality coding schemes for digital music. 21-24 - Yoshiharu Hoshino, Toshiyuki Takegahara:
Permissible value of group delay distortion on tone quality due to low-pass filters. 25-28 - Takao Arai, Takaharu Noguchi, Masaharu Kobayashi, Nobutaka Amada, Yasufumi Yumde, Kuniaki Miura:
A study on the digitization of audio signals for video tape recorder. 29-32 - Yasushi Katsumata, Osamu Hamada:
An audio sampling frequency conversion using digital signal processors. 33-36 - Masao Kasuga:
An approach to high resolution D/A converter utilizing a linear predictive coding. 37-40 - Biing-Hwang Juang, Lawrence R. Rabiner:
Mixture autoregressive hidden Markov models for speaker independent isolated word recognition. 41-44 - Serge Soudoplatoff:
Markov modeling of continuous parameters in speech recognition. 45-48 - Lalit R. Bahl, Peter F. Brown, Peter V. de Souza, Robert L. Mercer:
Maximum mutual information estimation of hidden Markov model parameters for speech recognition. 49-52 - Amir Averbuch, Lalit R. Bahl, Raimo Bakis, Peter F. Brown, A. G. Cole, G. Daggett, Subrata K. Das, Ken Davies, S. DeGennaro, Peter V. de Souza, E. Epstein, D. Fraleigh, Frederick Jelinek, Slava M. Katz, B. Lewis, Robert L. Mercer, Arthur Nádas, David Nahamoo, Michael Picheny, G. Shichman, P. Spinelli:
An IBM PC based large-vocabulary isolated-utterance speech recognizer. 53-56 - Jean-Luc Gauvain:
A syllable-based isolated word recognition experiment. 57-60 - Jean-Pierre Tubach, Louis-Jean Boë:
Quantitative knowledge on word structure, from a phonetic corpus, with application to large vocabularies recognition systems. 61-64 - Roberto Billi, G. Massia, F. Nesti:
Word preselection for large vocabulary speech recognition. 65-68 - Shoji Hiraoka, Shuji Morii, Masakatsu Hoshimi, Katsuyuki Niyada:
Compact isolated word recognition system for large vocabulary. 69-72 - Lynn Wilcox, Bruce T. Lowerre:
Coarse classification using a hierarchical decision tree and top down parsing. 73-76 - Kai-Fu Lee:
Incremental network generation in word recognition. 77-80 - Mitchel Weintraub:
A computational model for separating two simultaneous talkers. 81-84 - Vishu Viswanathan, Claudia M. Henry, Richard M. Schwartz, Salim E. Roucos:
Evaluation of multisensor speech input for speech recognition in high ambient noise. 85-88 - J. W. Kim, C. K. Un:
Enhancement of noisy speech by forward/Backward adaptive digital filtering. 89-92 - Gagan Mirchandani, Richard C. Gaus Jr., L. Kathy Bechtel:
Performance characteristics of a hardware implementation of the cross-talk resistant adaptive noise canceller. 93-96 - Yasuo Ariki, K. Kajimoto, Toshiyuki Sakai:
Acoustic noise reduction by two dimensional spectral smoothing and spectral amplitude transformation. 97-100 - Kuldip K. Paliwal:
Speech enhancement using multi-pulse excited linear prediction system. 101-104 - David J. Goodman, O. G. Jaffe, Gordon B. Lockhart, W. C. Wong:
Waveform substitution techniques for recovering missing speech segments in packet voice communications. 105-108 - Dimitrios P. Prezas, Joe Picone, David L. Thomson:
Fast and accurate pitch detection using pattern recognition and adaptive time-domain analysis. 109-112 - Francis Charpentier:
Pitch detection using the short-term phase spectrum. 113-116 - Masakazu Imai, Seiji Inokuchi:
Frequency identification by complex spectrum. 117-120 - Werner Verhelst, Fernand De Decker, Bart Francq, Oscar Steenhaut:
An adaptive non-uniform sign clipping preprocessor (ANUSC) for real-time autocorrelative pitch detection. 121-124 - Daniel W. Griffin, Jae S. Lim:
A high quality 9.6 kbps speech coding system. 125-128 - Takahiro Saito, Hideya Takeo, Kiyoharu Aizawa, Hiroshi Harashima, Hiroshi Miyakawa:
Adaptive discrete cosine transform image coding using gain/Shape vector quantizers. 129-132 - Tokumichi Murakami, Kohtaro Asai, Atsushi Itoh:
Vector quantization of color images. 133-136 - R. Aravind, Allen Gersho:
Low-rate image coding with finite-state vector quantization. 137-140 - Ho John Lee, Daniel T. L. Lee:
A gain-shape vector quantizer for image coding. 141-144 - J. P. Marescq, Claude Labit:
Vector quantization in transformed image coding. 145-148 - F. Oliveri, G. Conte, Mario Guglielmo:
A technique using a one-dimensional mapping for vector quantisation of images. 149-152 - P. A. Ramamoorthy, T. Tran:
A hybrid coding involving ADM and vector quantization for digital video image compression. 153-156 - R. H. J. M. Plompen, J. G. P. Groenveld, Dick E. Boekee, F. Booman:
The performance of a hybrid videoconferencing coder using displacement estimation in the transform domain. 157-160 - Stefan Carlsson, Christian Reillo:
Contour based representation of the displacement field for motion compensated image coding. 161-164 - Sharaf E. Elnahas, Kou-Hu Tzou:
Hybrid interframe coding of video signals with backward-acting motion detection. 165-168 - Lloyd J. Griffiths:
High-resolution spectral estimation: Rethinking the Fourier transform. 169-172 - Genshen Xu, Yoh-Han Pao:
Single vector approaches to eigenstructure analysis for harmonic retrieval. 173-176 - Hideaki Sakai:
Estimation of frequencies of sinusoids in colored noise. 177-180 - Peter J. Sherman, Arthur E. Frazho:
High resolution spectral estimation of sinusoids in colored noise using a modified Pisarenko decomposition. 181-184 - Eugene Church, P. Z. Takacs:
Spectral and parameter estimation problems arising in the metrology of high performance mirror surfaces. 185-188 - Cory Myers:
On the use of linear programming for spectral estimation. 189-192 - Sun-Yuan Kung, C. K. Lo, R. Foka:
A Toeplitz approximation approach to coherent source direction finding. 193-196 - S. Lawrence Marple Jr.:
Performance of multichannel autoregressive spectral estimators. 197-200 - Jim E. Schroeder, Rao K. Yarlagadda:
Linear predictive spectral analysis via the Lp norm. 201-204 - Yoshikazu Miyanaga, Nobuo Nagai, Nobuhiro Miki:
Refined ARMA digital lattice filter. 205-208 - Yonina Rosen, Boaz Porat:
ARMA parameter estimation based on sample covariances, for missing data. 209-212 - John W. Adams:
A new FFT approach to the interpolation of discrete-time signals. 213-216 - Naoki Suehiro, Mitsutoshi Hatori:
Fast algorithms for the discrete Fourier transform and for other transforms. 217-220 - Peter Kabal, B. Sayar:
Performance of fixed-point FFT's: Rounding and scaling considerations. 221-224 - C. X. Fan, S. H. Wang:
A fast Fourier transform algorithm using Hadamard transform. 225-228 - Pierre Duhamel, Martin Vetterli:
Cyclic convolution of real sequences: Hartley versus Fourier and new schemes. 229-232 - Ramasamy Krishnan, Graham A. Jullien, William C. Miller:
Computation of complex number theoretic transforms using quadratic residue number systems. 233-236 - Wan-Chi Siu, Anthony G. Constantinides:
A hardware efficient realisation of number theoretic convolvers. 237-240 - Vijay K. Jain, T. E. McClellan, Tapan K. Sarkar:
Half-Fourier transform and application to radar signals. 241-244 - Wirendre A. Perera, Peter J. W. Rayner:
Optimal design of multiplierless DFTS and FFTS. 245-248 - Ramesh C. Agarwal, James W. Cooley:
An efficient vector implementation of the FFT algorithm on IBM 3090VF. 249-252 - Yuval Bistritz, Hanoch Lev-Ari, Thomas Kailath:
Immitance-domain Levinson algorithms. 253-256 - Wonyong Sung, Sanjit K. Mitra:
Efficient multi-processor implementation of recursive digital filters. 257-260 - Jin-Der Wang, H. Joel Trussell:
A unified derivation of the fast RLS algorithms. 261-264 - De-Yuan Cheng, Allen Gersho:
A fast codebook search algorithm for nearest-neighbor pattern matching. 265-268 - Hiroshi Nagaoka, Yoshimi Monden, Suguru Arimoto:
The continuous-time limit of the discrete-time stability theory. 269-272 - Cheung Auyeung, Russell M. Mersereau, Ronald W. Schafer:
Maximum entropy deconvolution. 273-276 - Jesús M. Alcázar-Fernández, José R. Casar Corredera, Ramón García Gómez:
L1-norm noisy Tauberian deconvolution. 277-280 - Zhao-Xiong Wu:
A new homomorphic deconvolution system. 281-284 - Atsushi Imiya, Toshio Sasaki, Hidemitu Ogawa:
Generalized Lanczos method for signal smoothing. 285-288 - Zhenyu Li, Henrik V. Sorensen, C. Sidney Burrus:
FFT and convolution algorithms on DSP microprocessors. 289-292 - Paul M. Farrelle, S. Srinivasan, Anil K. Jain:
A unified transform architecture. 293-296 - William J. Vetter:
The array matrix generalization for signal processing. 297-300 - Kazuo Toraichi, Kazuki Katagishi, Ryoichi Mori:
A fast B-spline transform and its applications. 301-304 - Yasuo Sugiyama:
A generalization of Levinson algorithm for solving Toeplitz systems. 305-308 - Kazuyo Tanaka, Satoru Hayamizu, Kozo Ohta:
A demiphoneme network representation of speech and automatic labeling techniques for speech data base construction. 309-312 - Michael F. Guyote, Keith A. Lewis, Donald Lijana:
A speech data base at the united states air force academy. 313-316 - David S. Pallett:
A PCM/VCR speech database exchange format. 317-320 - Shuichi Itahashi:
A Japanese language speech database. 321-324 - Guy Perennou:
B.D.L.E.X. : A data and cognition base of spoken French. 325-328 - Victor W. Zue, D. Scott Cyphers, Robert H. Kassel, David H. Kaufman, Hong C. Leung, Mark Randolph, Stephanie Seneff, John E. Unverferth III, Timothy Wilson:
The development of the MIT Lisp-machine based speech research workstation. 329-332 - Kari Torkkola, Heikki Riittinen:
A microprocessor-based word recognition system for large vocabularies. 333-336 - A. Fukui, Y. Fujihashi, F. Nakagawa:
Signal processor application to voice dialing equipment. 337-340 - Richard V. Cox, D. Bock, K. B. Bauer, James D. Johnston, Jeffrey Snyder:
The analog voice privacy system. 341-344 - N. Morgan, Baruch Mazor:
Development of a 16-kb/s speech codec for telephone applications. 345-348 - Sami Aly:
24-channel 32 kb/s ADPCM transcoder using the CCITT recommendation G.721. 349-352 - Ph. Missakian, M. Milgram, B. Zavidovique:
A special architecture for dynamic programming. 353-356 - William G. Bliss, J. Girard, J. Avery, M. Lightner, Louis Scharf:
A modular architecture for dynamic programming and maximum likelihood sequence estimation. 357-360 - Allen L. Gorin, J. E. Shoenfelt, R. N. Lewine:
Speech recognition on the DADO/DSP multiprocessor. 361-364 - Evert Dijkstra, Christian Piguet:
A new systolic decomposition for the dynamic time warping algorithm. 365-368 - Francis Jutand, Nicolas Demassieux, Dominique Vicard:
VDP : A versatile high performance vector distance processor. 369-372 - Yoshitake Suzuki:
Design of an efficient dynamic time warping LSI. 373-376 - Makoto Morito, Kozo Yamada, Akihiko Fujisawa, Masao Takeuchi:
A single-chip speaker independent voice recognition system. 377-380 - Henri Barral, Nicolas Moreau:
VLSI Architecture for a real-time LPC-based feature extractor. 381-384 - Frans J. van Wijk, Frank P. Welten, Jef L. van Meerbergen, Jan Stoter, Jos A. Huisken, Antoine Delaruelle, Karel E. van Eerdewijk, Josef Schmid, Jan H. Wittek:
On the IC architecture and design of a 2 µm CMOS 8 MIPS digital signal processor with parallel processing capability: The PCB5010/5011. 385-388 - J. L. Laborie, D. F. Martin, J. C. Michalina, A. Picco:
VLSI Digital signal processor (PSI). 389-392 - S. Abiko, M. Hashizume, Y. Matsushita, K. Shinozaki, T. Takamizawa, Cole Erskine, Surendar Magar:
Architecture and applications of a 100-ns CMOS VLSI digital signal processor. 393-396 - John P. Roesgen:
A high performance microprocessor for DSP applications. 397-400 - Takao Kaneko, Hironori Yamauchi, Atsushi Iwata:
A 50ns floating-point signal processor VLSI. 401-404 - Yoshikazu Mori, Toshio Jufuku, Masao Iida, Akira Nomura, Noboru Ichiura, Takao Nakamura:
Architecture of high-speed 22-bit floating-point digital signal processor. 405-408 - Takao Nishitani, Ichiro Kuroda, Yuichi Kawakami, H. Tanaka, Tom Nukiyama:
Advanced single-chip signal processor. 409-412 - Shingo Tsujimichi, TakaHide Ohkami, Yukihiko Shimazu:
A next-generation 32-bit VLSI signal processor. 413-416 - Robert E. Owen, Bruce E. Miller:
Architectural considerations for a sub 10 nanosecond DSP building block family. 417-420 - James R. Boddie, W. Patrick Hays, James Tow:
The architecture, instruction set and development support for the WE®DSP32 digital signal processor. 421-424 - Yoshihiro Tomita, Shigeyuki Unagami, Tomohiko Taniguchi, Yasuhiko Tada, Masahiro Taka:
Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3). 425-428 - Christopher R. Cole, Amine Haoui, Peter L. Winship:
A high performance digital voice echo canceller on a single TMS32020. 429-432 - Eric P. Farges, Mark A. Clements:
Hidden Markov models applied to very low bit rate speech coding. 433-436 - Biing-Hwang Juang:
Design and performance of trellis vector quantizers for speech signals. 437-440 - Yasuo Matsuyama:
Joint time-spectral vector quantization and inverse filter set. 441-444 - Hitoshi Koyama, Allen Gersho:
Fully vector-quantized multipulse LPC at 4800 bps. 445-448 - Hüseyin Abut, Siegfried Ergezinger:
Low-rate speech encoding using vector quantization and subband coding. 449-452 - Richard C. Rose, Thomas P. Barnwell III:
The self excited vocoder - an alternate approach to toll quality at 4800 bps. 453-456 - Kazunori Ozawa, Takashi Araseki:
Low bit rate multi-pulse speech coder with natural speech quality. 457-460 - Daniel Lin:
A novel LPC synthesis model using a binary pulse source excitation. 461-464 - Per Hedelin:
High quality glottal LPC-vocoding. 465-468 - Luis A. Hernández Gómez, Francisco Javier Casajús-Quirós, Aníbal R. Figueiras-Vidal, Ramón García-Gómez:
On the behaviour of reduced complexity code-excited linear prediction (CELP). 469-472 - Joseph P. Campbell Jr., Thomas E. Tremain:
Voiced/Unvoiced classification of speech with applications to the U.S. government LPC-10E algorithm. 473-476 - Karl-Heinz Brenner, Adolf W. Lohmann:
Optical circuitry and architectures for digital optical computing. 477-480 - Yoshiki Ichioka, Jun Tanida:
Optical parallel array logic system. 481-484 - Keikichi Hirose, Hiroya Fujisaki, Yasuhiro Kosugi:
Use of optical signal processing techniques to spectrum analysis of speech. 485-488 - Nobuo Goto, Yasumitsu Miyazaki:
Optical signal processors consisting of collinear acousto-optic channel waveguides. 489-492 - T. Asakura, T. Iwai, Tohru Ifukube, Toshio Kawashima:
Reproduction of the sounds from old wax phonographic cylinders using the laser-beam reflection method. 493-496 - P. P. Vaidyanathan:
New cascaded lattice structures for FIR filters having extremely low coefficent sensitivity. 497-500 - Masami Iwatsuki, Masayuki Kawamata, Tatsuo Higuchi:
Synthesis of minimum sensitivity structures in linear systems using controllability and observability measures. 501-504 - Kou-Hu Tzou:
Embedded max quantization. 505-508 - K. Pahlavan:
Nonlinear quantization and data communication. 509-512 - Sasan H. Ardalan:
Floating point error analysis of recursive least squares and least means squares adaptive filters. 513-516 - Akinori Nishihara:
Design of limit-cycle-free digital biquad filters. 517-520 - Michael E. Lukacs:
Predictive coding of multi-viewpoint image sets. 521-524 - Leonardo Chiariglione, Luigi Corgnier, Mario Guglielmo:
An architecture for the universal video codec. 525-528 - Eugene Walach, Ehud D. Karnin:
A fractal based approach to image compression. 529-532 - Kou-Hu Tzou