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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
DSP: Applications and Techniques
- John R. Treichler, Michael G. Larimore, C. Richard Johnson Jr., Sally L. Wood:
The application of SHARF to adaptive removal of TV ghosting. 554-559 - O. A. Horna:
Correction algorithms for extended range echo cancellers. 560-563 - Tariq S. Durrani, C. J. Macleod, J. Pearson, Gordon Hayward:
Processing techniques for the inspection of offshore structures. 564-567 - John M. Milan:
An architecture for an unattended radar signal processor. 568-571 - Michael G. Larimore, B. J. Langland:
Recursive linear prediction for clock synchronization. 572-575 - Paul Lansky, Kenneth Steiglitz:
Synthesis of timbral families by warped linear prediction. 576-578 - Robert H. Sperry, David C. Farden:
A microprogrammed signal processor. 579-582
Digital Audio
- Barry A. Blesser:
Perceptual issues in digital processing of music. 583-586 - Kees A. Immink:
Modulation systems for digital audio discs with optical readout. 587-589 - Guy W. McNally, T. A. Moore:
A modular signal processor for digital filtering and dynamic range control of high quality audio signals. 590-594 - Roger Lagadec, Henry O. Kunz:
A universal, digital sampling frequency converter for digital audio. 595-598
Medium Band Speech Coding I
- Bishnu S. Atal, Joel R. Remde:
Split-band APC System for low bit-rate encoding of speech. 599-602 - James L. Melsa, Arun Pande:
Mediumband speech encoding using time domain harmonic scaling and adaptive residual coding. 603-606 - Jared J. Wolf, Kenneth D. Field:
Real-time speech coder implementation at 9.6 and 16 kb/s. 607-610 - John I. Makhoul, Michael G. Berouti, Michael A. Krasner:
Time and frequency domain noise shaping in speech coding. 611-614 - R. Viswanathan, William Russell, Alan L. Higgins:
Noisy-channel performance of 16 kb/s APC coders. 615-618 - Jean E. Menez, J. F. Galliano, Claude R. Galand:
Comparative study of 16 kbps adaptive predictive coders. 619-622 - Thomas E. Eger, L. William Varner:
A novel adaptive predictor DPCM system for speech signals. 623-626 - Michael A. Krasner, Michael G. Berouti, John I. Makhoul:
Stability analysis of APC systems. 627-630 - Ronald S. Cheung, S. Y. Kwon:
The design of a 16 kb/s split-band adaptive predictive coder for noisy channels. 631-635 - Subrata K. Das:
Considerations in the implementation of an adaptive transform coder for speech in a 16-bit fixed point arithmetic microprocessor. 636-639 - Legand L. Burge Jr.:
Efficient coding of the prediction residual. 640-643
Digital Signal Processing Architecture and Hardware
- Akira Kanemasa, Rikio Maruta, Kenji Nakayama, Y. Sakamura, S. Tanaka:
An LSI chip set for DSP hardware implementation. 644-647 - Hassan M. Ahmed, Martin Morf, Daniel T. L. Lee, Ping Ang:
A VLSI speech analysis chip set based on square root normalized ladder forms. 648-653 - Victor Godbole, Zaheer Ali:
Realtime spectrum analysis using a microprocessor peripheral. 654-657 - Stanley A. White:
Architecture for a digital programmable image processing element. 658-661 - John A. Eldon, Craig Robertson:
Single-board digital signal processor. 662-665 - Dan I. Moldovan:
A feasibility study of microprocessor-based digital filters. 666-669 - Rose Gibson, Donald P. McCabe:
Fourier transform algorithm implementations on a general-purpose microprocessor. 670-672 - Robert A. Collesidis, Todd A. Dutton, Joseph R. Fisher, W. T. Metcalf:
Control of multiprocessor SPS-1000 configurations using principles of data-flow architecture. 673-676 - Peter Alexander:
Fast convolution using modularized multiprocessor architectures. 677-680 - Antonio Ruiz:
On the hardware implementation of a multi-processor environment for several DSP applications. 681-684
Two-Dimensional Filter Design
- K. S. Thyagarajan:
Design of 2-D IIR digital filters with circular symmetry by transformation of the variable. 685-687 - Samy A. H. Aly, Moustafa M. Fahmy:
2-D separable digital filters. 688-691 - Harnatha C. Reddy, P. Karivaratha Rajan, M. N. Shanmukha Swamy:
Design of two-dimensional digital filters using analog reference filters without second kind singularities. 692-695 - David B. Harris:
Design of 2-D rational digital filters. 696-699 - John W. Woods, I. Paul, N. Sangal:
2-D direct form, recursive filter design with magnitude and phase specifications. 700-703 - Majid Ahmadi, Venkat Ramachandran:
A method for the design of stable (N-D) analog and digital filters. 704-707 - Thomas F. Quatieri, Dan E. Dudgeon:
Extensions of 2-D iterative digital filters. 708-711 - P. A. Ramamoorthy:
Two-dimensional hexagonal digital recursive filters. 712-715 - Leonid M. Roytman, M. N. Shanmukha Swamy:
An ℓ2-stability theorem for multidimensional IIR digital filters. 716-719
Discrete Utterance Recognition
- Hermann Ney:
An optimization algorithm for determining the endpoints of isolated utterances. 720-723 - Lawrence R. Rabiner, Jay G. Wilpon:
Isolated word recognition using a two-pass pattern recognition approach. 724-727 - N. Rex Dixon, Harvey F. Silverman:
What are the significant variables in dynamic programming for discrete utterance recognition? 728-731 - Dean P. McCullough:
Variations on Itakura's spectral match score. 732-735 - Michael H. Kuhn, Horst Tomaschewski, Hermann Ney:
Fast nonlinear time alignment for isolated word recognition. 736-740 - Robert J. Fontana, Michael S. Fox:
A composite source model for speaker and isolated word recognition. 741-745 - John G. Ackenhusen, Lawrence R. Rabiner:
Microprocessor implementation of an LPC-based isolated word recognizer. 746-749 - John L. Anderson:
Optical processing techniques for real-time isolated word recognition. 750-752 - James H. Clark, Patricia Collins, Bruce T. Lowerre:
A formalization of performance specifications for discrete utterance recognition systems. 753-757 - Gérard Chollet, Alain Astier, Mario Rossi:
Evaluating the performance of speech recognisers at the acoustic-phonetic level. 758-761
Audio
- J. Robert Ashley:
The ASSP sound reinforcement systems. 762-765 - Jont B. Allen:
Cochlear modeling - 1980. 766-769 - J. Robert Ashley:
Echoes, reverberation, speech intelligibility and musical performance. 770-772 - Tracy Petersen, Steven F. Boll:
Critical band analysis-synthesis. 773-775 - Vijay K. Jain, W. Marshall Leach Jr., Ronald W. Schafer:
Time-domain measurement of vented-box loudspeaker system parameters. 776-781 - Glen Cascino, J. Robert Ashley:
Digital simulation of loudspeaker systems. 782-785 - Austin J. Brouns:
Second-order gradient noise-cancelling microphone. 786-789 - Douglas Preis:
Phase equalization for magnetic recording. 790-795 - Dean Wallraff:
The effects of primitive size on the overall design of real-time digital audio signal processors. 796-799 - W. Stephen Bussey, Robert Haigler:
Tubes versus transistors in electric guitar amplifiers. 800-803
Medium and Wide Band Speech Coding
- Philippe Mabilleau, Jean-Pierre Adoul:
Medium band speech coding using a dictionary of waveforms. 804-807 - Thomas P. Barnwell III:
An experimental study of sub-band coder design incorporating recursive quadrature filters and optimum ADPCM. 808-811 - Barry M. Abzug:
Using the prediction residual to improve LPC synthesis for 9600 bps applications. 812-815 - David L. Cohn, James L. Melsa, Arvind Arora, James M. Kresse, Arun Pande:
Practical considerations for variable length source coding. 816-819 - George S. Kang, Lawrence J. Fransen, Evans L. Kline:
Mediumband speech processor with baseband residual spectrum encoding. 820-823 - Harald Katterfeldt:
A DFT-based residual-excited linear predictive coder (RELP) for 4.8 and 9.6kb/s. 824-827 - Maurizio Copperi, Neviano Dal Degan, J. Roberto B. de Marca:
An adaptive delta modulator with noise spectral shaping. 828-831 - V. Ramamoorthy:
Speech coding using modulo-PCM with side information. 832-835 - Jerry D. Gibson:
Bounds on performance and dynamic bit allocation for sub-band coders. 836-839 - Joseph Tierney, Marilyn L. Malpass:
Enhanced CVSD-An Embedded speech coder for 64-16 kbps. 840-843 - P. J. Patrick, Costas S. Xydeas, Raymond Steele, Wai-kuen Cham:
Wideband quality speech encoders with bit rates of 16-32 kbits/s. 844-847 - Georges Bonnerot, Jean-Marie Raulin, Maurice G. Bellanger:
Performance of a 32 kbit/s ADPCM coder for digital long-haul telephone transmission. 848-851
Fast Recursive Estimation Methods in System Identification
- Larry Marple:
Fast least squares FIR system identification. 852-855 - Martin Morf, Carlos H. Muravchik, Daniel T. L. Lee:
Hilbert space array methods for finite rank process estimation and ladder realizations for adaptive signal processing. 856-859 - Hanoch Lev-Ari, Thomas Kailath:
Schur and Levinson algorithms for nonstationary processes. 860-864 - Benjamin Friedlander:
A modified lattice algorithm for deconvolving filtered impulsive processes. 865-868 - John R. Deller Jr.:
Computer simulation studies of a simplified model for a certain class of autoregressive systems. 869-872 - Jeffrey D. Klein, Bradley W. Dickinson:
A time-recursive lattice algorithm for autoregressive estimation. 873-876 - Boaz Porat, Benjamin Friedlander, Martin Morf:
Square-root covariance ladder algorithms. 877-880 - Claude Guéguen:
Linear prediction in the singular case and the stability of eigen models. 881-885 - William J. Done:
Use of the fast Kalman estimation algorithm for adaptive system identification. 886-889 - Yoshimi Monden, Masashi Yamada, Suguru Arimoto:
A new recursive method for identification of multivariate linear systems from impulse response and output covariance information. 890-893 - J. Bee Bednar:
Automatic order selection in the design of IIR filters. 894-896 - Vijay K. Jain, Tapan K. Sarkar, Donald D. Weiner:
Rational modeling by pencil-of-functions method. 897-900
Adaptive Processing II
- John Y. Cheung:
Error smoothing in adaptive LMS algorithms. 901-904
Connected-Word and Subword-Segment Recognition
- Cory S. Myers, Lawrence R. Rabiner:
Connected word recognition using a level building dynamic time warping algorithm. 951-955 - Cory S. Myers, Stephen E. Levinson:
Connected word recognition using a syntax-directed dynamic programming temporal alignment procedure. 956-959 - Rainer Zelinski, Fritz Class:
A segmentation procedure for connected word recognition based on estimation principles. 960-963 - James D. Marr:
Comparison of several clustering algorithms for data rate compression of LPC parameters. 964-966 - Aaron E. Rosenberg, Lawrence R. Rabiner, Stephen E. Levinson, Jay G. Wilpon:
A preliminary study on the use of demisyllables in automatic speech recognition. 967-970 - Günther Ruske, Thomas Schotola:
The efficiency of demisyllable segmentation in the recognition of spoken words. 971-974 - Heikki Riittinen, Seppo Haltsonen, Erkki Reuhkala, Matti Jalanko:
Experiments on an isolated-word recognition system for multiple speakers. 975-978 - Matthew Yuschik, Hinrich R. Martens:
An evaluation of hierarchical features in speech recognition. 979-982
Two-Dimensional Digital Signal Processing
- Samy A. H. Aly, Moustafa M. Fahmy:
Symmetry in two-dimensional rectangularly-sampled digital filters. 983-986 - Robert A. King, Ahmet H. Kayran:
A new stabilization technique for 2-dimensional recursive digital filters. 987-990 - Leonid M. Roytman, James F. Delansky, M. N. Shanmukha Swamy:
A technique for coefficient minimum word length estimation sufficient for stability maintenance in N-D filters. 991-994 - Hyokang Chang, Jake K. Aggarwal:
Implementation of two-dimensional semicausal recursive digital filters. 995-999 - Jae S. Lim, Naveed A. Malik:
A new algorithm for one-dimensional and two-dimensional maximum entropy power spectrum estimation. 1000-1005 - Theresa C. Speake, Russell M. Mersereau:
Evaluation of two-dimensional discrete Fourier transforms via generalized FFT algorithms. 1006-1009 - Theresa C. Speake, Russell M. Mersereau:
An interpolation technique for periodically sampled two-dimensional signals. 1010-1013 - Monson H. Hayes:
Multidimensional signal reconstruction from phase or magnitude. 1014-1017 - Sergei Fogel, Rui J. P. de Figueiredo:
Thematic filtering and scene interpretation based on complete invariant systems of features. 1018-1021 - Hiroyuki Mizutani, Naoyuki Kasagi, Noriaki Hirayama:
Movement and deformation prediction for pluvial distributions. 1022-1025
Sonar, Radar, and Seismology: Models and Processing
- Anthony I. Eller, H. Joseph Venne Jr.:
Evaluation procedure for environmental acoustic models. 1026-1029 - Alastair D. McAulay, W. Clay Choate, R. N. Shurtleff:
Accurate modeling for shallow water wave propagation. 1030-1033 - Magnus Moll:
Effects of input power fluctuations on passive sonar receiver performance. 1034-1037 - Jude Franklin, Robert J. Urick:
Conventional and binary time series models of sonar fluctuations. 1038-1041 - Robert S. Walker, D. V. Crowe, W. R. Mayo:
Synthetic multichannel time series for simulating underwater acoustic noise. 1042-1045 - C. J. Macleod, Tariq S. Durrani, Gordon Hayward:
A new model of the piezoelectric ultrasonic transducer. 1046-1049 - Rajendar Bahl, P. V. Indiresan:
A novel beamformer for circular sonar arrays. 1050-1053 - Carey Gibson, Simon Haykin:
Performance studies of adaptive lattice prediction-Error filters for target detection in a radar environment using real data. 1054-1057 - Samuel D. Stearns, Luke J. Vortman:
Seismic event detection using adaptive predictors. 1058-1061 - Kou-Yuan Huang, Clare D. McGillem, Paul E. Anuta:
Analytic signal representation of the synthetic seismogram of bright spots. 1062-1065
Speech Enhancement and Quality
- Elliot Singer:
The effects of microphones and facemasks on LPC vocoder performance. 1066-1069 - Charles F. Teacher, David C. Coulter:
Operation of LPC vocoders in noisy environments. 1070-1073 - Dakshesh Parikh, David Mansour, John D. Markel:
Study of echo cancelling algorithms for full duplex telephone networks with vocoders. 1074-1077 - Robert D. Preuss:
Low complexity robust linear predictive coding of speech signals. 1078-1081 - Chong Kwan Un, K. Y. Choi:
Improving LPC analysis of noisy speech by autocorrelation subtraction method. 1082-1085 - Tracy Petersen, Steven F. Boll:
Acoustic noise suppression in the context of a perceptual model. 1086-1088 - Richard V. Cox, David Malah:
A technique for perceptually reducing periodically structured noise in speech. 1089-1092 - Yves Grenier, Kalle-J. Bry, Joël Le Roux, M. Sulpis:
Autoregressive models for noisy speech signals. 1093-1096 - Mark A. Richards:
A system for helium speech enhancement using the short-time Fourier transform. 1097-1100 - P. Breitkopf, Thomas P. Barnwell III:
Segmental preclassification for improved objective speech quality measures. 1101-1104
Digital Image Processing
- S. Hamid Nawab, Alan V. Oppenheim, Jae S. Lim:
Improved spectral subtraction for signal restoration. 1105-1108 - J. A. Ponnusamy, Mandyam D. Srinath:
Sequential image enhancement using 2-D adaptive estimation. 1109-1112 - Anil K. Jain, Surendra Ranganath:
Application of two dimensional spectral estimation in image restoration. 1113-1116 - Tamar Peli, Jae S. Lim:
Adaptive filtering for image enhancement. 1117-1120 - William A. Pearlman, Priyadarshan Jakatdar:
Hybrid DFT/DPCM interframe image quantization. 1121-1124 - Sankar K. Pal, Robert A. King:
Application of fuzzy set theory in detecting x-ray edges. 1125-1128 - Vito Cappellini, Luigi Odorico:
A new operator for edge detection. 1129-1131 - Bernd Girod:
Objective quality measures for the design of digital image transmission systems. 1132-1135 - Thomas S. Huang, Roger Y. Tsai:
Three-dimensional motion estimation from image-space shifts. 1136-1139 - Graham A. Jullien, William C. Miller:
A two-dimensional finite field processor for image filtering. 1140-1143 - Paul S. Schenker, E. G. Cande, Kon Max Wong, William R. Patterson III:
New sensor geometries for image processing: Computer vision in the polar exponential grid. 1144-1148
Continuous Speech Recognition
- Lalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer:
Continuous parameter acoustic processing for recognition of a natural speech corpus. 1149-1152 - Arthur Nádas, Robert L. Mercer, Lalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis:
Continuous speech recognition with automatically selected acoustic prototypes obtained by either bootstrapping or clustering. 1153-1155 - Michael Wagner:
Automatic labelling of continuous speech with a given phonetic transcription using dynamic programming algorithms. 1156-1159 - Seppo Haltsonen:
Improvement and comparison of three phonemic segmentation methods of speech. 1160-1163 - Kiyohiro Shikano:
Acoustic processing in the conversational speech recognition system. 1164-1167 - Lalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer:
Speech recognition of a natural text read as isolated words. 1168-1171 - Katsuhiko Shirai:
Vowel identification in continuous speech using articulatory parameters. 1172-1175 - Edward C. Bronson, Leah J. Siegel:
A parallel architecture for speech understanding. 1176-1179
Finite Wordlength Effects in Digital Signal Processing
- A. Jalali, K. R. Rao:
Limited wordlength and FDCT processing accuracy. 1180-1183 - Ganapati Panda, Ranendra N. Pal, B. Chatterjee:
Fixed-point error analysis of Winograd basic DFT algorithms considering correlation between noise sources. 1184-1188 - Amnon Aliphas, Allen M. Peterson:
Narrowband filtering with frequency sampling filters. 1189-1192 - Douglas Preis, Carey D. Bunks:
Computational and performance aspects of minimax equalizers. 1193-1196 - Kenji Nakayama:
Permuted difference coefficient digital filters. 1197-1200 - Adly T. Fam:
Multiplicative linear phase FIR filters. 1201-1204 - Anil Mahanta, R. C. Agarwal, S. C. Dutta Roy:
FIR filter structures having low sensitivity and roundoff noise. 1205-1208 - T. L. Chang:
A unified analysis of roundoff noise reduction in digital filters. 1209-1212 - J. W. K. Lam, Venkatanarayana Ramachandran, M. N. Shanmukha Swamy:
Comparison of the effects of quantization on digital filters. 1213-1216 - Alan G. Bolton:
Transfer functions available with a new second order digital filter structure. 1217-1220 - Pei-Hwa Lo, Yih-Chyun Jenq:
On the overflow problem in a second order digital filter. 1221-1226 - Ahmad I. Abu-El-Haija:
On limit cycle amplitudes in error-feedback digital filters. 1227-1230 - David C. Munson Jr.:
Determining exact maximum amplitude limit cycles in digital filters. 1231-1234
Time Delay and Coherence Estimation
- Albert A. Gerlach:
A high-speed algorithm for mapping 2-D ambiguity surfaces. 1235-1238 - Leon H. Sibul, Lawrence J. Ziomek:
Generalized wideband crossambiguity function. 1239-1242 - Robert Lugannani:
Distribution of the sample magnitude-squared coherence obtained using overlapped Fourier transforms. 1243-1246 - Jacques Hay:
Improvement of the imaging of moving acoustic sources by the knowledge of their motion. 1247-1252 - Delores M. Etter, M. M. Masukawa:
A comparison of algorithms for adaptive estimation of the time delay between sampled signals. 1253-1256 - Dae Hee Youn, Nasir Ahmed, G. Clifford Carter:
A method for generating a class of time-delayed signals. 1257-1260 - Kent Scarbrough, Nasir Ahmed, G. Clifford Carter:
Some considerations of time delay estimation. 1261-1264 - Azizul H. Quazi:
Effects of signal and noise spectral slopes on time delay estimates in passive localization. 1265-1268 - Mati Wax:
The estimate of time delay between two signals with random relative phase shift. 1269-1272
Connected Word Recognition
- Yves Grenier, Laurent Miclet, J. C. Maurin, H. Michel:
Speaker adaptation for phoneme recognition. 1273-1275
Sonar, Radar, and Seismology: Models and Processing
- Driss Aboutajdine, Zine El Abidine Amri, Mohamed Najim:
Robust arrival time determination of seismic waves in noise. 1276-1279
Late Papers
- Mauro J. Dentino, Harry M. Huey:
Comparison of detection performance of a time domain and frequency domain adaptive algorithm. 1280-1283 - Kenneth R. Perry:
A comparison of the effects of component nonidealities on the performance of analog and digital LMS adaptive noise cancellers. 1284-1287 - David Malah:
Efficient spectral matching of the LPC residual signal. 1288-1291
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