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ORCIDICASSP 1990: Albuquerque, New Mexico, USA
- 1990 International Conference on Acoustics, Speech, and Signal Processing, ICASSP '90, Albuquerque, New Mexico, USA, April 3-6, 1990. IEEE 1990
- W. Bastiaan Kleijn
:
Source-dependent channel coding for CELP. 1-4 - Michael S. Brandstein, Peter A. Monta, John C. Hardwick, Jae S. Lim:
A real-time implementation of the improved MBE speech coder. 5-8 - Paul C. Meuse:
A 2400 bps multi-band excitation vocoder. 9-12 - Carmen García-Mateo, Francisco Javier Casajús-Quirós, Luis A. Hernández Gómez:
Multi-band vector excitation coding of speech at 4.8 kbps. 13-16 - Jorge S. Marques, Luís B. Almeida, José M. Tribolet:
Harmonic coding at 4.8 kb/s. 17-20 - Kazunori Mano, Takehiro Moriya:
4.8 kbit/s delayed decision CELP coder using tree coding. 21-24 - Jae H. Chung, Ronald W. Schafer:
Excitation modeling in a homomorphic vocoder. 25-28 - Masami Akamine, Kimio Miseki:
CELP coding with an adaptive density pulse excitation model. 29-32 - Rosario Drogo de Iacovo, Roberto Montagna, Daniele Sereno:
Vector quantization and perceptual criteria in SVD based CELP coders. 33-36 - Saeed Vaseghi:
Finite state CELP for variable rate speech coding. 37-40 - Jean-Claude Junqua:
ORION: a two pass hybrid system for isolated-words automatic speech recognition. 41-44 - Alexander I. Rudnicky, Michelle Sakamoto, Joseph Polifroni:
Spoken language interaction in a goal-directed task. 45-48 - Victor Zue, James R. Glass, David Goodine, Michael Philips, Stephanie Seneff:
The SUMMIT speech recognition system: phonological modelling and lexical access. 49-52 - Toshiyuki Hanazawa, Kenji Kita, Satoshi Nakamura, Takeshi Kawabata, Kiyohiro Shikano:
ATR HMM-LR continuous speech recognition system. 53-56 - Volker Steinbiss, Andreas Noll, A. Peaseler, Hermann Ney, Henning Bergmann, Christian Dugast, Hans-Hermann Hamer, H. Piotrowski, Horst Tomaschewski, Andrea Zielinski:
A 10000-word continuous-speech recognition system. 57-60 - Shinta Kimura:
100000-word recognition using acoustic-segment networks. 61-64 - Lin-Shan Lee, Chiu-yu Tseng, Hung-Yan Gu, Fu-hua Liu, Robert Chen-Hao Chang, Shew-Heng Hsieh, Chian-hung Chen:
A real-time Mandarin dictation machine for Chinese language with unlimited texts and very large vocabulary. 65-68 - Martin J. Russel, Keith Ponting, S. M. Peeling, Sue Browning, John S. Bridle, Roger K. Moore, Isabel Galiano, P. Howell:
The ARM continuous speech recognition system. 69-72 - Victor Zue, James R. Glass, David Goodine, Hong C. Leung, Michael S. Phillips, Joseph Polifroni, Stephanie Seneff:
The VOYAGER speech understanding system: preliminary development and evaluation. 73-76 - Michael Cohen, Hy Murveit, Jared Bernstein, Patti J. Price, Mitch Weintraub:
The decipher speech recognition system. 77-80 - Richard M. Schwartz, Yen-Lu Chow:
The N-best algorithms: an efficient and exact procedure for finding the N most likely sentence hypotheses. 81-84 - Lalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, Dimitri Kanevsky, David Nahamoo:
Constructing groups of acoustically confusable words. 85-88 - Abdulmesih Aktas, Otto Schmidbauer, K. H. Maier, Wolfgang Feix:
Classification of coarse phonetic categories in continuous speech: statistical classifiers vs. temporal flow connectionist network. 89-92 - Stephen E. Levinson, Andrej Ljolje, Laura G. Miller:
Continuous speech recognition from a phonetic transcription. 93-96 - David S. Pallett, William M. Fisher, Jonathan G. Fiscus:
Tools for the analysis of benchmark speech recognition tests. 97-100 - Edmund Lai, G. A. Carrijo, R. Bennett, Roberto Togneri, Michael D. Alder, Yianni Attikiouzel:
An English language speech database at the University of Western Australia. 101-104 - Joseph Picone:
The demographics of speaker independent digit recognition. 105-108 - Charles R. Jankowski Jr., Ashok Kalyanswamy, Sara Basson, Judith Spitz:
NTIMIT: a phonetically balanced, continuous speech, telephone bandwidth speech database. 109-112 - David P. Morgan, Christopher L. Scofield, Theresa M. Lorenzo, Edward C. Real, David P. Loconto:
A keyword spotter which incorporates neural networks for secondary processing. 113-116 - Herbert Gish, Yen-Lu Chow, Jan Robin Rohlicek:
Probabilistic vector mapping of noisy speech parameters for HMM word spotting. 117-120 - Noboru Sugamura:
Continuous speech recognition using large vocabulary word spotting and CV syllable spotting. 121-124 - Ayman Asadi, Richard M. Schwartz, John Makhoul:
Automatic detection of new words in a large vocabulary continuous speech recognition system. 125-128 - Richard C. Rose, Douglas B. Paul:
A hidden Markov model based keyword recognition system. 129-132 - Fritz Class, Alfred Kaltenmeier, Peter Regel-Brietzmann, Karl Trottler:
Fast speaker adaptation for speech recognition systems. 133-136 - Francis Kubala, Richard M. Schwartz, Chris Barry:
Speaker adaptation from a speaker-independent training corpus. 137-140 - Gerhard Rigoll:
Baseform adaptation for large vocabulary hidden Markov model based speech recognition systems. 141-144 - Chin-Hui Lee, Chih-Heng Lin, Biing-Hwang Juang:
A study on speaker adaptation of continuous density HMM parameters. 145-148 - Luc Mathan, Laurent Miclet:
Speaker hierarchical clustering for improving speaker-independent HMM word recognition. 149-152 - Hiroaki Hattori, Satoshi Nakamura, Kiyohiro Shikano:
Supplementation of HMM for articulatory variation in speaker adaptation. 153-156 - Satoshi Nakamura, Kiyohiro Shikano:
A comparative study of spectral mapping for speaker adaptation. 157-160 - S. J. Cox, John S. Bridle:
Simultaneous speaker normalisation and utterance labelling using Bayesian/neural net techniques. 161-164 - John B. Hampshire II, Alexander H. Waibel:
The Meta-Pi network: connectionist rapid adaptation for high-performance multi-speaker phoneme recognition. 165-168 - Katsuhiko Shirai, Naoki Hosaka, Eiichiro Kitagawa:
Speaker adaptive phoneme recognition by multi-level clustering based on mutual information criterion. 169-172 - Natalie Fournier, Yves Grenier:
A real-time vector excited adaptive predictive coder at 9.6 kbit/s. 173-176 - Claude Laflamme, Jean-Pierre Adoul, H. Y. Su, Sarto Morissette:
On reducing computational complexity of codebook search in CELP coder through the use of algebraic codes. 177-180 - Juin-Hwey Chen, Melvin J. Melchner, Richard V. Cox, Duane O. Bowker:
Real-time implementation and performance of a 16 kb/s low-delay CELP speech coder. 181-184 - Frank K. Soong, Biing-Hwang Juang:
Optimal quantization of LSP parameters using delayed decisions. 185-188 - Roar Hagen, Per Hedelin:
Low bit-rate spectral coding in CELP, a new LSP-method. 189-192 - Nam Phamdo, Nariman Farvardin:
Coding of speech LSP parameters using TSVQ with interblock noiseless coding. 193-196 - Philip A. Chou, Tom D. Lookabaugh:
Conditional entropy-constrained vector quantization of linear predictive coefficients. 197-200 - Richard A. Dean:
Global maximum likelihood voice decoding. 201-204 - Alessandro Falaschi, Massimo Giustiniani, Piero Pierucci:
A finite states Markov quantizer for speech coding. 205-208 - F. F. Tzeng:
An analysis-by-synthesis linear predictive model for narrowband speech coding. 209-212 - Masaaki Honda:
Speech coding using waveform matching based on LPC residual phase equalization. 213-216 - Ki Yong Lee, Byung-Gook Lee, Iickho Song, Souguil Ann:
On Bernoulli-Gaussian process modeling of speech excitation source. 217-220 - J. A. Naylor:
A neural network algorithm for enhancing delta modulation/LPC tandem connections. 221-224 - Mahesan Niranjan:
CELP coding with adaptive output-error model identification. 225-228 - S. A. Atungsiri, Ahmet M. Kondoz, Barry G. Evans:
Error detection and control for the parametric information in CELP coders. 229-232 - Renaud J. Di Francesco, Claude Lamblin, Alain Le Guyader, Dominique Massaloux:
Variable rate speech coding with online segmentation and fast algebraic codes. 233-236 - Jerry D. Gibson, Yoon-Chae Cheong, Hong Chae Woo, Wen-Whei Chang:
A comparison of backward adaptive prediction algorithms in low delay speech coders. 237-240 - Tomohiko Taniguchi, Yoshinori Tanaka, A. Sasama, Yasuji Ohta:
Principal axis extracting vector excitation coding: high quality speech at 8 kb/s. 241-244 - Eyal Yair, Kenneth Zeger, Allen Gersho:
Conjugate gradient methods for designing vector quantizers. 245-248 - Robert J. McAulay, Thomas F. Quatieri:
Pitch estimation and voicing detection based on a sinusoidal speech model. 249-252 - Alfredo Restrepo Palacios, Irwin W. Sandberg, Alan C. Bovik:
Non-Euclidean locally monotonic regression. 1201-1204 - Yoram Bresler, Scott P. Litke:
A parametric technique for superresolution image reconstruction. 1205-1208 - Vip Desai, K. S. Arun:
Robust parametric reconstruction of signals by rational modeling. 1209-1212 - Barry J. Sullivan:
Ill-conditioned signal restoration and the conjugate gradient method. 1213-1216 - M. A. Khasawneh, Winser E. Alexander:
Mean square error analysis for the fast LMS-sine algorithm. 1257-1260 - Heping Ding, Moustafa M. Fahmy:
Inverse of linear periodically time-varying filtering. 1217-1220 - Boualem Bouachache, Langford B. White:
Instantaneous frequency estimation and automatic time-varying filtering. 1221-1224 - Laith Naaman, Alan C. Bovik:
Least-squares order statistic filters for signal restoration in dependent noise. 1225-1228 - Francesco Palmieri:
Adaptive recursive order statistic filters. 1229-1232 - John L. Brown Jr., Sergio D. Cabrera:
Multi-channel signal reconstruction using noisy samples. 1233-1236 - Naoki Saito:
Superresolution of noisy band-limited data by data adaptive regularization and its application to seismic trace inversion. 1237-1240 - Bhaskar D. Rao:
Some new properties of the LMS FIR-ALE. 1241-1244 - Nam Ik Cho, Sang Uk Lee:
On the performance analysis of ALE using an IIR lattice notch filter. 1245-1248 - John J. Shynk, Christina K. Chan:
A comparative analysis of the stationary points of the constant modulus algorithm based on Gaussian assumptions. 1249-1252 - Peter J. Voltz, Frank Kozin:
Almost-sure sample stability of the continuous-time LMS algorithm. 1253-1256 - Victor E. DeBrunner, A. A. (Louis) Beex:
An informational approach to the convergence of output error adaptive IIR filter structures. 1261-1264 - Majid Nayeri, Hong Fan:
Asymptotic stability of SMM and RGM for insufficient order IIR adaptive filters. 1265-1268 - Tarek I. Haweel, Peter M. Clarkson:
Analysis and generalization of a median adaptive filter. 1269-1272 - Raziel Haimi-Cohen, Hanan Herzberg, Yair Be'ery:
Delayed adaptive LMS filtering: current results. 1273-1276 - Nancy E. Hubing, S. Thomas Alexander:
Statistical analysis of the soft constrained initialization of recursive least squares algorithms. 1277-1280 - Bowonkoon Chitprasert, K. R. Rao:
Discrete cosine transform filtering. 1281-1284 - Gregory W. Medlin:
A design technique for high order digital differentiators. 1285-1288 - Ali Zilouchian, Nurgun Erdol:
A minimum-order structure for fixed-point digital filters. 1289-1292 - Arnab K. Shaw, Pradeep Misra:
An exact realization of 2-D IIR filters using separable 1-D modules. 1293-1296 - Y. Z. Zhang, T. W. Cole, Y. F. Yao:
Optimum removal of LCOs in digital filters. 1297-1300 - Nevio Benvenuto, Michele Marchesi, Aurelio Uncini:
Results on the application of simulated annealing algorithm for the design of digital filters with powers-of-two coefficients. 1301-1304 - C. Sidney Burrus, Admadji W. Soewito, Ramesh A. Gopinath:
Least squared error FIR filter design with spline transition functions. 1305-1308 - M. Scott Andrews, Joseph Picone, Ronald D. DeGroat:
Robust pitch determination via SVD based cepstral methods. 253-256 - Mohammad Ali Masnadi-Shirazi, Nasir Ahmed:
Laguerre approximation of nonrecursive discrete-time systems. 1309-1312 - Hans Wilhelm Schüßler, Peter Steffen:
On the design of allpasses with prescribed group delay. 1313-1316 - L. Hodgson, M. E. Jernigan, Barry L. Wills:
Nonlinear multiplicative cepstral analysis for pitch extraction in speech. 257-260 - John Oglesby, John S. Mason:
Optimisation of neural models for speaker identification. 261-264 - Wei-Ping Zhu, Zhenya He:
Design of three-dimensional spherically symmetric digital filters. 1317-1320 - Younès Bennani, Françoise Fogelman-Soulié, Patrick Gallinari:
A connectionist approach for automatic speaker identification. 265-268 - Takao Kobayashi, Satoshi Imai:
Complex Chebyshev approximation for IIR digital filters using an iterative WLS technique. 1321-1324 - Aaron E. Rosenberg, Chin-Hui Lee, Frank K. Soong:
Sub-word unit talker verification using hidden Markov models. 269-272 - Ren-Hua Wang, Lingshen He, Hiroya Fujisaki:
A weighted distance measure based on the fine structure of feature space: application to speaker recognition. 273-276 - Virginia L. Stonick, S. Thomas Alexander:
Global optimal IIR filter design and ARMA estimation using homotopy continuation methods. 1325-1328 - Chi-Shi Liu, Wern-Jun Wang, Min-Tau Lin, Hsiao-Chuan Wang:
Study of line spectrum pair frequencies for speaker recognition. 277-280 - Paul T. Yang, Rajeev Jain, Toshiaki Yoshino, Wanda Gass, Ashwin Shah:
A functional silicon compiler for high speed FIR digital filters. 1329-1332 - Leland B. Jackson, Gerald J. Lemay:
A simple Remez exchange algorithm to design IIR filters with zeros on the unit circle. 1333-1336 - Richard J. Hartnett, Gloria Faye Boudreaux:
Design of efficient parallel hybrid FIR filters using dynamic programming and subset selection methods. 1337-1340 - Roy Chapman, M. A. Rahman:
Chebyshev polynomial based transfer functions for orthogonal lattice filters. 1341-1344 - Michael I. Savic, Sunil K. Gupta:
Variable parameter speaker verification system based on hidden Markov modeling. 281-284 - Jonathan S. Abel:
A bound on mean square estimate error. 1345-1348 - Constantin Papaodysseus, Constantin Triantafillou, Elias Koukoutsis, George Carayannis:
Error propagation and numerical recovery for a class of parametric DSP algorithms. 1349-1352 - Yifan Gong, Jean-Paul Haton:
Text-independent speaker recognition by trajectory space comparison. 285-288 - Don H. Johnson, Anand R. Kumar:
Modeling and analyzing fractal point processes. 1353-1356 - Brian H. Maranda, John A. Fawcett:
The performance analysis of a fourth-moment detector. 1357-1360 - Herbert Gish:
Robust discrimination in automatic speaker identification. 289-292 - Herbert Gish:
A probabilistic approach to the understanding and training of neural network classifiers. 1361-1364 - Victor C. Soon, Lang Tong, Y. F. Huang, R. Liu:
An extended fourth order blind identification algorithm in spatially correlated noise. 1365-1368 - Richard C. Rose, Douglas A. Reynolds:
Text independent speaker identification using automatic acoustic segmentation. 293-296 - Michael S. Scordilis, John N. Gowdy:
Neural network control for a cascade/parallel formant synthesizer. 297-300 - S. Kalson:
Adaptive detection with diagonal loading [for adaptive antenna array]. 1369-1372 - Johan de Veth, Bert Cranen, Helmer Strik, Lou Boves:
Extraction of control parameters for the voice source in a text-to-speech system. 301-304 - Georgios B. Giannakis, Amod V. Dandawate:
Linear and non-linear adaptive noise cancelers. 1373-1376 - John J. Shynk, Sumit Roy:
Analysis of a perceptron learning algorithm with momentum updating. 1377-1380 - Mahmood R. Azimi-Sadjadi, Stuart Citrin, Sassan Sheedvash:
Supervised learning process of multi-layer perceptron neural networks using fast least squares. 1381-1384 - V. John Mathews, Zhenhua Xie:
Stochastic gradient adaptive filters with gradient adaptive step sizes. 1385-1388