default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDICASSP 1981: Atlanta, Georgia, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '81, Atlanta, Georgia, USA, March 30 - April 1, 1981. IEEE 1981
Narrow Band Speech Coding I
- David Y. Wong, Biing-Hwang Juang, Augustine H. Gray Jr.:
Recent developments in vector quantization for speech processing. 1-4 - Kun-Shan Lin, Ying Tsui:
LPC compressed speech at 850 bits-per-second. 5-7 - John J. O'Donnell:
A system for very low data rate speech communication. 8-11 - Hüseyin Abut, Robert M. Gray, Guillermo Rebolledo:
Vector quantization of speech waveforms. 12-15 - Eric Dorsey, Jared Bernstein:
Inter-speaker comparison of LPC acoustic space using a minimax distortion measure. 16-19 - James D. Marr, Thomas P. Barnwell III:
Two dimensional prediction and interpolation for data rate compression of LPC parameters. 20-23 - Jesse W. Fussell:
A differential linear predictive voice coder for 1200 bps. 24-27 - Robert J. McAulay:
A low-rate vocoder based on an adaptive subband formant analysis. 28-31 - Bernard Gold:
Experiments with a pattern-matching channel vocoder. 32-34
Algorithms
- Thomas F. Quatieri, Victor T. Tom, Monson H. Hayes, James H. McClellan:
Convergence of iterative signal reconstruction algorithms. 35-38 - Sun-Yuan Kung, D. V. Bhaskar Rao:
Highly parallel architectures for solving linear equations. 39-42 - Peter R. Cappello, Kenneth Steiglitz:
Some intractable problems in digital signal processing. 43-46 - Gérard Thomas:
An improvement of the Van-Cittert's method. 47-49 - H. Joel Trussell, L. A. Schwalbe:
Methods for deconvolving sparse positive delta function series. 50-57 - Michael A. Rodriguez, Richard H. Williams, T. J. Carlow:
Estimation and detection of variable latency signals via unwrapped phase averaging. 58-61 - Hermann Ney:
A dynamic programming technique for nonlinear smoothing. 62-65
Time Varying and Related Spectral Estimation
- Carsten Thomsen, Jens Hee:
Analysis of non-stationary signals-Digital filters vs. FFT. 66-68 - Theo A. C. M. Claasen, Wolfgang F. G. Mecklenbräuker:
Time-frequency signal analysis by means of the Wigner distribution. 69-72 - Nian-Chyi Huang, Jake K. Aggarwal:
Spectral modifications using linear shift-variant digital filters. 73-76 - George A. Lippert, Mostafa Kaveh:
Frequency errors in the spectral estimates of complex sinusoids using the "Tapered" burg technique. 77-80 - Yiu Tong Chan, P. J. McCabe, J. B. Plant:
Nonlinear estimation of frequency and phase of a sinusoid in noise. 81-84 - Maureen Quirk, Bede Liu:
On narrow-band spectrum calculation by direct decimation. 85-88 - Mukta L. Kar, J. O. Hornkohl, W. M. Farmer:
A new approach to Fourier analysis of randomly sampled data using linear regression technique. 89-93 - Lonnie C. Ludeman:
Optimum sampling for estimation of Fourier coefficients in noise. 94-97 - Ernest G. Baxa Jr., Stephen D. Huffman:
On leakage reduction associated with spectral power domain averaging to improve estimate quality. 98-101
Speech Synthesis
- Susan R. Hertz:
SRS text-to-phoneme rules: A three-level rule strategy. 102-105 - Herbert E. Wolf:
Control of prosodic parameters for a formant synthesizer based on diphone concatenation. 106-109 - Helmut Dettweiler:
An approach to demisyllable speech synthesis of German words. 110-113 - Stefano Sandri, Enrico Vivalda:
Automatic stress assignment for Italian text-to-speech synthesis. 114-117 - Sverre Holm:
Automatic generation of mixed excitation in a linear predictive speech synthesizer. 118-120
Aids for the Handicapped and Laryngeal Analysis
- Bruce L. Hicks, Louis D. Braida, Nathaniel I. Durlach:
Pitch invariant frequency lowering with nonuniform spectral compression. 121-124 - Steven V. De Gennaro, Kenneth R. Krieg, Louis D. Braida, Nathaniel I. Durlach:
Third-octave analysis of multichannel amplitude compressed speech. 125-128 - Douglas C. Sargent:
A procedure for synchronizing continuous speech with its corresponding printed text. 129-132 - Ashok Krishnamurthy, Donald G. Childers:
Vocal fold vibratory patterns: Comparison of film and inverse filtering. 133-136 - Alan M. Smith:
Feature extraction for laryngeal evaluation. 137-140
Sonar Signal Processing
- John T. Rickard:
Optimal array processing with 3-D random arrays. 141-144 - Stephen W. Lang, Gregory L. Duckworth, James H. McClellan:
Array design for MEM and MLM array processing. 145-148 - Malcolm T. Stark:
Matched array processing for wideband passive sonar. 149-152 - Georges Bienvenu, Laurent Kopp:
Source power estimation method associated with high resolution bearing estimator. 153-156 - Stanislav B. Kesler, Simon Haykin, Robert S. Walker:
Maximum-entropy field-mapping in the presence of correlated multipath. 157-161 - Ethan Aronoff, David Rivers:
Data turning-Average signal to noise ratio improvement. 162-167 - Steven Kay:
Improved detection performance of an FM signal by autoregressive spectral analysis. 168-170 - A. L. Vyas, P. V. Indiresan:
Performance evaluation of a new robust detector for sonar signals. 171-175 - Roger F. Dwyer:
Asymptotic performance measures for sequential partition detectors. 176-179
Talker Recognition
- Timothy Diller, John F. Siebenand:
Speaker-independent word recognition using sex-dependent clustering. 180-183 - Aaron E. Rosenberg, Kathleen L. Shipley:
Speaker identification and verification combined with speaker independent word recognition. 184-187 - Hermann Ney:
Telephone-line speaker recognition using clipped autocorrelation analysis. 188-192 - Edwin H. Wrench Jr.:
A realtime implementation of a text independent speaker recognition system. 193-196 - Malayappan Shridhar, N. Mohankrishnan, M. R. Baraniecki:
Text-independent speaker recognition using orthogonal linear prediction. 197-200
Narrow Band Speech Coding II
- Manfred R. Schroeder, Bishnu S. Atal:
Rate distortion theory and predictive coding. 201-204 - Per Hedelin:
A tone oriented voice excited vocoder. 205-208 - Peter L. Chu, David G. Messerschmitt:
An allpass transformed lattice filter with improved sensitivity properties. 209-212 - Luís B. Almeida, José M. Tribolet:
A model for short-time phase prediction of speech. 213-216 - Vijay K. Jain:
Linear predictor with a new error criterion. 217-219
Digital Filter Design
- Daniel J. Esteban, Claude R. Galand:
HQMF: Halfband quadrature mirror filters. 220-223 - Gulamabbas A. Merchant, Thomas W. Parks:
Inverse filtering for systems with unit circle zeroes. 224-227 - Tariq S. Durrani, Roy Chapman:
Constrained optimization solutions to I.I.R filter design using discrete prolate spheroidal wave functions. 228-231 - Robert A. Gabel:
On the design and performance of equiripple IIR interpolators. 232-235 - Tapio Saramäki, Yrjö Neuvo, Tapio Saarinen:
Equal ripple amplitude and maximally flat group delay digital filters. 236-239 - Mark W. Smith, David C. Farden:
Thinning the impulse response of FIR digital filters. 240-242 - Bernard Widrow, Paul F. Titchener, Richard P. Gooch:
Adaptive design of digital filters. 243-246 - Benjamin Friedlander, Martin Morf:
Least-squares algorithms for adaptive linear-phase filtering. 247-250 - A. S. Ramnarayanan, Fred J. Taylor:
On the structure of IIR filters using residue arithmetic. 251-254 - Stephen C. Pohlig, Gary A. Shaw, Theodore Bially, Thomas F. Quatieri:
A nested algorithm for improving the accuracy of chirp-Fourier transform implementations. 255-258 - Michel Feldmann, Pierre Duhamel:
The multibridge charge coupled filter: General synthesis using bias conservative structure. 259-262
Adaptive Processing I
- William S. Hodgkiss:
The adaptive lattice array processor. 263-266 - Michael L. Honig, David G. Messerschmitt:
Convergence models for adaptive gradient and least squares algorithms. 267-270 - Dennis R. Morgan:
Response of a delay-constrained adaptive linear predictor filter to a sinusoid in white noise. 271-274 - Arye Nehorai, Martin Morf:
Enhancement of sinusoids in colored noise and the whitening performance of exact least-squares predictors. 275-278 - Frank W. Symons Jr.:
Use of linear prediction in detecting narrowband signals in colored noise. 279-282 - Kevin M. Buckley, S. Rao:
A comparison of adaptive gradient and adaptive least-squares algorithms. 283-286 - Frank K. Soong, S. Shankar Narayan, Allen M. Peterson:
On the asymptotic behavior of a complex adaptive line enchancer (CALE). 287-292 - Raymond S. Medaugh, Lloyd J. Griffiths:
A comparison of two fast linear predictors. 293-296 - Tariq S. Durrani, N. L. M. Murukutla, Ken C. Sharman:
Constrained algorithms for multi input adaptive lattices in array processing. 297-301 - James A. Cadzow, Thomas P. Bronez:
An algebraic approach to super-resolution adaptive array processing. 302-305
Transform and Convolution Methods
- Yoshiaki Tadokoro, Tatsuo Higuchi:
Another discrete Fourier transform computation with small multiplications via the Walsh transform. 306-309 - Masud Arjmand, Richard A. Roberts:
Multifactor algorithms for noncyclic digital convolution. 310-314 - Henri J. Nussbaumer:
Inverse polynomial transform algorithms for DFTs and convolutions. 315-318 - Thomas A. Kriz:
Corner-turn complexity properties of polynomial transform 2D convolution methods. 319-322 - S. Prakash, V. V. Rao:
Fixed-point error bound for convolution by polynomial transforms, with application to FIR filtering. 323-326 - Meghanad D. Wagh, Salvatore D. Morgera:
Cyclic convolution algorithms over finite fields: Multidimensional considerations. 327-330 - David P. Maher:
Long convolutions using transforms over reducible extensions of fermat number rings. 331-334 - C. Sidney Burrus:
A new prime factor FFT algorithm. 335-338 - G. Robert Redinbo, William J. Hunnebeck:
On the simulation of residue number systems. 339-342
Speech Analysis
- Benjamin Friedlander, Sidhartha Maitra:
Speech deconvolution by recursive ARMA lattice filters. 343-346 - B. Yegnanarayana:
A pole-zero model for cepstrally smoothed speech spectra. 347-350 - David H. Friedman:
Estimation of formant parameters by sum-of-poles modeling. 351-354 - Fumio Sugiyama, Makoto Nakamura:
An LPC vocoder for efficient implementation. 355-358 - Harald Höge:
Estimation of the dynamics of vocal tract parameters. 359-361 - Elaine Cohen:
A spline approach to speech analysis/Synthesis. 362-365 - Per Hedelin, Gunnar Hult:
QD-an algorithm for non-linear inverse filtering. 366-369 - Tai-Yi Huang, Cai-Fei Wang, Yoh-Han Pao:
Speech analysis for Chinese Putonghua (mandarin). 370-373
LSI Advances in Speech and Signal Processing
- Louis Schirm IV:
A family of high speed, floating point arithmetic chips. 374-377 - Bernard New:
The Am29500 signal processing family. 378-381 - Eric Dorsey, Jim Caldwell:
Application of the PDSP chip set to LPC synthesis. 382-385 - Akira Ichikawa, Kazuo Nakata, Akio Komatsu, Yoshiaki Kitazume:
Conceptual system design for continuous speech recognition LSI. 386-389 - Gideon Amir, R. Gregorian, Gwyn Edwards:
The implementation of a speech synthesis algorithm. 390-393
Time Varying and Related Spectral Estimation
- James W. Cooley, Shmuel Winograd:
On the use of filter design programs for generating spectral windows. 394-396
Sonar Signal Processing
- Mauro J. Dentino, H. M. Huey, James R. Zeidler:
Comparative performance of adaptive and conventional detectors for finite bandwidth signals. 397-400
Speech Hardware
- R. Geppert:
Hardware implementation of a 15-channel filter bank. 451-454 - Ronald E. Crochiere, Mark A. Randolph, John W. Upton, James D. Johnston:
Real-time implementation of sub-band coding on a programmable integrated circuit. 455-458 - Daniel J. Esteban, Claude R. Galand:
Multiport implementation of real time 16kbps sub-band coder. 459-462 - K. Moidin Mohiuddin, S. Shankar Narayan, K. Chen, Allen M. Peterson:
A general purpose real time digital speech processor. 463-466 - David Vetter, John Stork, Klaus Skoge, Paul Ahrens:
LPC speech I.C. using a 12-pole cascade digital filter. 467-470 - David J. Burr, Bryan D. Ackland, Neil Weste:
A high speed array computer for dynamic time warping. 471-474
ARMA and MEM Spectral Estimation
- James A. Cadzow, Koji Ogino:
Adaptive ARMA spectral estimation. 475-479 - Yoh-Han Pao, Dennis T. Lee:
Additional results on the Cadzow ARMA method for spectrum estimation. 480-483 - James A. Cadzow, Randolph L. Moses:
A superresolution method of ARMA spectral estimation. 484-487 - Benjamin Friedlander:
A recursive maximum likelihood algorithms for ARMA line enhancement. 488-491 - Alberto Mordojovich, Richard A. Roberts:
A comparison of spectral estimators for real data. 492-495 - Chrysostomos L. Nikias, Peter D. Scott:
Improved spectral resolution by energy-weighted prediction method. 496-499 - Henry Stark, Chander S. Sarna:
Pattern recognition of waveforms using modern spectral estimation techniques and its application to earthquake/Explosion data. 500-502 - T. Sen Lee:
Identification and spectral estimation of noisy multivariate autoregressive processes. 503-507 - Robert H. Wilkinson:
Will the real MEM please stand up? 508-511 - Louis L. Scharf, Aloysius A. Beex, T. von Reyn:
Modal decomposition of covariance sequences for parametric spectrum analysis. 512-517
Adaptive Processing II
- S. M. Sharpe, Loren W. Nolte:
Adaptive MSE estimation. 518-521 - Lloyd J. Griffiths, Donald W. Cooley:
Nonstationary effects in adaptive filtering. 522-525 - David B. Harris:
Recursive least squares with linear constraints. 526-529 - David C. Farden, Mark W. Smith:
Bounds on tracking errors for adaptive signal processing algorithms. 530-533 - C. Richard Johnson Jr., Brian D. O. Anderson:
Sufficient excitation and stable reduced-order adaptive IIR filtering. 534-537 - C. Richard Johnson Jr., I. D. Landau, T. Taylor, Luc Dugard:
On adaptive IIR filters and parallel adaptive identifiers with adaptive error filtering. 538-541 - David L. Soldan:
A comparison of adaptive algorithms used for designing phase compensation filters. 542-545 - Jelisaveta Kesler, Simon Haykin:
An adaptive interference canceller using Kalman filtering. 546-549 - David Mansour, Augustine H. Gray Jr.:
Frequency domain non-linear adaptive filter. 550-553