default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDICASSP 1980: Denver, Colorado, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '80, Denver, Colorado, USA, April 9-11, 1980. IEEE 1980
Session Plenary
- Marcian E. Hoff Jr.:
IC Technology: Trends and impact on digital signal processing. 1-6
Speech
Narrowband Speech - I
- Richard H. Wiggins, James H. Parry:
Effect of corruption within the recursive estimation of spectral parameters for LPC. 7-10 - James D. Marr, Thomas P. Barnwell III:
Two-dimensional prediction of area functions for coding of LPC speech parameters. 11-14 - Andres Buzo, Augustine H. Gray Jr., Robert M. Gray, John D. Markel:
Speech coding based upon vector quantization. 15-18 - Nick G. Kingsbury, W. A. Amos:
A robust channel vocoder for adverse environments. 19-22 - Sidhartha Maitra, Charles R. Davis:
Improvements in the classical model for better speech quality. 23-27 - Panos Papamichalis, Thomas P. Barnwell III:
A dynamic programming approach to variable rate speech transmission. 28-31 - Richard M. Schwartz, John W. Klovstad, John Makhoul, John Aasted Sørensen:
A preliminary design of a phonetic vocoder based on a diphone model. 32-35 - Jesse W. Fussell:
The Karhunen-Loeve transform applied to the log area ratios of a linear predictive speech coder. 36-39
Pitch Detection and Vocal Cord Models
- Daniel T. L. Lee, Martin Morf:
A novel innovation based time domain pitch detector. 40-44 - Robert J. Sluyter, H. J. Kotmans, A. V. Leeuwaarden:
A novel method for pitch extraction from speech and a hardware model applicable to vocoder systems. 45-48 - Bruce Fette, Rose Gibson, E. Greenwood:
Windowing functions for the average magnitude difference function pitch extractor. 49-52 - Leah J. Siegel, Alan C. Bessey:
A decision tree procedure for voiced/Unvoiced/Mixed excitation classification of speech. 53-56 - V. Ramamoorthy:
Voice/Unvoice detection based on a composite-Gaussian source model of speech. 57-60 - Raymond Descout, Jean-Yves Auloge, Bernard Guérin:
Continuous model of the vocal source. 61-64 - Donald G. Childers, J. S. Mott, G. P. Moore:
Automatic parameterization of vocal cord motion from ultra high speed films. 65-68
Digital Signal Processing
Quantization Effects
- David C. Munson Jr., Bede Liu:
Floating point error bound in the prime factor FFT. 69-72 - Dusan M. Kodek:
An algorithm for the design of optimal finite word-length FIR digital filters. 73-76 - Chrysostomos L. Nikias, Adly T. Fam:
Precise pole realization by unobservable digital filters. 77-80 - P. Ananthakrishna, Sanjit K. Mitra:
Block-state recursive digital filters with minimum round-off noise. 81-84 - Masayuki Kawamata, Tatsuo Higuchi:
A sufficient condition for absence of overflow oscillations in arbitrary digital filters based on the element equations. 85-88 - Peter L. Chu, David G. Messerschmitt:
Zero sensitivity analysis of the digital lattice filter. 89-93
Computational Complexity and Fast Algorithms (invited)
- Shmuel Winograd:
Signal processing and complexity of computation. 94-101 - Thomas Kailath:
Generalized Levinson algorithms and Ladder filters for nonstationary signal processing. 102
Underwater Acoustics and Adaptive Filtering
The Acoustic Medium
- William J. Vetter:
A raypath reflection model for layered media with source and receiver in different layers. 103-106 - G. Beresford-Smith, I. M. Mason:
Seismic imaging of faults in multi-moded coal seams. 107-110 - Alastair D. McAulay, W. Clay Choate, R. N. Shurtleff:
Digital generation of accurate synthetic seismograms. 111-114 - Anthony I. Eller, John F. Miller:
Environmental influences on acoustic array design and performance in shallow water. 115-119 - James P. Reilly, Simon Haykin:
An experimental study of the MEM applied to array antennas in the presence of multipath. 120-123 - Magnus Moll:
A compound random process for underwater ambient acoustic noise. 124-127
Speech
Narrowband Speech - II
- Bernard Gold:
Formant representation of parameters for a channel vocoder. 128-130 - John M. Turner, Bradley W. Dickinson, Daniel Lai:
Characteristics of reflection coefficient estimates based on a Markov chain model. 131-134 - John D. Markel, Augustine H. Gray Jr.:
An experimental comparison of two scalar quantization methods. 135-137 - Nils Rydbeck, Per Tjernlund, Jan Uddenfeldt:
A 4.8 KBPS voice excited DFT vocoder with time encoded baseband. 138-141 - Chong Kwan Un, Wonyong Sung:
A 4800 bps LPC vocoder with improved excitation. 142-145
Speech Analysis and Reconstruction
- Michel Chafcouloff, Gérard Chollet, Paul P. Durand, Jacques Guizol, Xavier Rodet:
Observation and modelling of "Formant" transitions using ISASS. 146-149 - R. C. Cox, David M. Robinson:
Some notes on phase in speech signals. 150-153 - Sidhartha Maitra, Scott H. Foster, Charles R. Davis:
A maximum peakiness criterion for deconvolving speech waveforms. 154-157 - Frank K. Soong, Allen M. Peterson:
Fast spectral estimation of speech signal in analytic form. 158-161 - Kil Ho Song, Chong Kwan Un:
On pole-zero modeling of speech. 162-165
Discrete and Connected Word Recognition
- Edward P. Neuburg:
Frequency-axis warping to improve automatic word recognition. 166-168 - Harvey F. Silverman, N. Rex Dixon:
State constrained dynamic programming (SCDP) for discrete utterance recognition. 169-172 - Cory S. Myers, Lawrence R. Rabiner, Aaron E. Rosenberg:
An investigation of the use of dynamic time warping for word spotting and connected speech recognition. 173-177 - Subrata K. Das:
Some experiments in discrete utterance recognition. 178-181 - Lawrence R. Rabiner, Jay G. Wilpon, Aaron E. Rosenberg:
Application of isolated word recognition to a voice controlled repertory dialer system. 182-185 - Herbert Bierfert, M. Kirstein, D. Lance:
Some aspects of evaluating the performance of a speech recognition system in real applications. 186-189 - John R. Welch, Sheldon C. Oxenberg:
Reduction of minimum word-boundary gap lengths in isolated word recognition. 190-193 - Lawrence R. Rabiner, C. E. Schmidt:
A connected digit recognizer based on dynamic time warping and isolated digit templates. 194-198 - Ryohei Nakatsu:
A speech recognition machine for connected words. 199-202 - Stephen E. Levinson, Kathleen L. Shipley:
A conversational mode airline information and reservation system using speech input and output. 203-208 - Robert E. Wohlford, A. Richard Smith, Marvin R. Sambur:
The enhancement of wordspotting techniques. 209-212
DFTs and FFTs
- James W. Cooley, Shmuel Winograd:
A limited range discrete Fourier transform algorithm. 213-217 - David P. Maher:
Mathematical background for generalized, partial, and incomplete discrete Fourier transforms. 218-221 - Norman Brenner:
Rapidly "Bit-reversing" data for the past Fourier transform. 222-223 - Gordon L. DeMuth:
A scaling approach for FFT processing. 224-226 - Bradley W. Dickinson, Kenneth Steiglitz:
An approach to the diagonalization of the discrete Fourier transform. 227-230 - Stephen A. Dyer, Nasir Ahmed, Donald R. Hummels:
Computation of the discrete cosine transform via the arcsine transform. 231-234 - Henri J. Nussbaumer:
Fast polynomial transform methods for multidimensional DFTs. 235-237 - Chao H. Huang, Fred J. Taylor:
High speed DFT's using residue numbers. 238-242 - Daniel Minoli, Wendell Nakamine:
Mersenne numbers rooted on 3 for number theoretic transforms. 243-247
Digital Filter Design
- Gloria Faye Boudreaux, Thomas W. Parks:
Digital filters with thinned numerators. 248-251 - Kenneth Steiglitz:
Design of FIR digital phase networks. 252-255 - Robert A. Gabel:
On asymmetric FIR interpolators with minimum Lperror. 256-259 - A. A. (Louis) Beex, Louis L. Scharf:
Covariance sequence approximation for recursive digital filter design. 260-263 - Charles K. Chui, Andrew K. Chan:
A new approach to causal filter design by Padé approximants. 264-267 - Adly T. Fam:
A multiplicative realization of FIR systems that is logarithmically efficient. 268-270 - Eugene B. Hogenauer:
A class of digital filters for decimation and interpolation. 271-274 - Tapio Saramäki:
Optimum recursive digital filters with zeros on the unit circle. 275-278 - B. Yegnanarayana:
Pole-zero decomposition: A new technique for design of digital filters. 279-282 - Tapio Saramäki, Yrjö Neuvo:
Equal ripple amplitude and group delay digital filters. 283-286 - J. Bee Bednar, William A. Coberly:
Order selection for lowpass IIR filters. 287-290 - James D. Johnston:
A filter family designed for use in quadrature mirror filter banks. 291-294
Underwater Acoustics and Adaptive Filtering
Array Processing
- A. M. Vural:
On the problem of fixed shading in conjunction with an optimal/Adaptive array processor. 295-298 - Bernard J. Repasky, Ben R. Breed:
Application of ridge regression analysis to optimum array processing. 299-302 - Andrew C. Callahan:
Interference removal for random arrays: Beam decoupling approaches. 303-306 - Georges Bienvenu, Laurent Kopp:
Adaptivity to background noise spatial coherence for high resolution passive methods. 307-310 - William S. Hodgkiss:
Dynamic beamforming of a random acoustic array. 311-314
Speech
Medium Band Coding - I
- John Ben O'Neal Jr., R. Rao Koneru, Jagannath P. Agrawal:
Digital encoding of phase shift keying voiceband data signals. 315-318 - Ronald S. Cheung, Raimond L. Winslow:
High quality 16 kb/s voice transmission: The subband coder approach. 319-322 - Subrata K. Das:
A technique for speech coding using dynamic programming. 323-326 - Michael A. Krasner:
The critical band coder-Digital encoding of speech signals based on the perceptual requirements of the auditory system. 327-331 - Claude R. Galand, Daniel J. Esteban:
16kbps Real time QMF sub-band coding implementation. 332-335 - José M. Tribolet, Ronald E. Crochiere:
A modified adaptive transform coding scheme with post-processing-enhancement. 336-339 - Ronald E. Crochiere, José M. Tribolet, Lawrence R. Rabiner:
On the measurement of waveform coder distortion using the log likelihood ratio. 340-343 - L. E. Bergeron, Aaron J. Goldberg, Soon Young Kwon, M. Miller:
A robust, adaptive transform coder for 9.6 kb/s speech transmission. 344-347 - R. Viswanathan, Alan L. Higgins, William Russell, John Makhoul:
Baseband LPC coders for speech transmission over 9.6 kb/s noisy channels. 348-351 - H. Ravindra:
Speech articulation rate change using recursive bandwidth scaling. 352-355 - Michael G. Berouti, John Makhoul:
An embedded-code multirate speech transform coder. 356-359 - Aspi B. Wadia:
Error correction scheme for telephone line transmission of RELP vocoder. 360-363 - Michael J. McLane, James L. Melsa, David L. Cohn:
A single chip speech codec and filter. 364-367
Digital Signal Processing
VLSI - The Real Hope for Digital Signal Processing? (invited)
- Earl E. Swartzlander Jr.:
Signal processing architectures with VLSI. 368-371 - John R. Mick, Bernard New:
Bit slice devices for signal processing. 372-375 - Shlomo Waser:
Survey of VLSI for digital signal processing. 376-379 - Bill Koral, Louis Schirm IV:
Floating-point arithmetic for digital signal processing. 380-382 - John S. Thompson, James R. Boddie:
An LSI digital signal processor. 383-385 - Takao Nishitani, Yuichi Kawakami, Rikio Maruta, Akira Sawai:
LSI signal processor development for communications equipment. 386-389 - Matt Townsend, Marcian E. Hoff Jr.:
A single chip NMOS signal processor. 390-393 - Gwyn Edwards:
A speech/Speaker recognition and response system. 394-397 - Richard Wiggins:
An integrated circuit for speech synthesis. 398-401 - Dennis Morris, David Weinrich:
A new speech synthesis chip set. 402-405
Image Processing
- John W. Woods, Vinay K. Ingle, R. Hingorani, G. Juskovic:
Experimental comparison of reduced update Kalman filters and Wiener filters for two-dimensional LMMSE estimation. 406-409 - Fernand S. Cohen, David B. Cooper, Howard Elliott, Peter F. Symosek:
Two-dimensional image boundary estimation by use of likelihood maximization and Kalman filtering. 410-413 - Sarah A. Rajala, Rui J. P. de Figueiredo:
Adaptive nonlinear image restoration by a modified Kalman filtering approach. 414-417 - Roger Y. Tsai, Thomas S. Huang:
Moving image restoration and registration. 418-421 - Uwe L. Haass, Thomas A. Brubaker:
Estimation of cloud motion from satellite pictures. 422-425 - Richard E. Twogood:
2-D Digital signal processing with an array processor. 426-429 - Thomas A. Kriz, Dale F. Bachman:
A number theoretic transform approach to image rotation in parallel array processors. 430-433 - R. Lynn Kirlin:
Median filter and 102422D FFT with an FPS AP-120B array processor. 434-436 - Monson H. Hayes, Jae S. Lim, Alan V. Oppenheim:
Phase-only signal reconstruction. 437-440 - Richard L. Frost, Craig K. Rushforth:
A new non-linear superresolution algorithm. 441-443 - Jae S. Lim:
Image restoration by short space spectral subtraction. 444-448 - James H. McClellan:
Artifacts in alpha-rooting of images. 449-452
Underwater Acoustics and Adaptive Filtering
Adaptive Filters - I
- Stephen D. Huffman, Loren W. Nolte:
Adaptive linear estimation based on time domain orthogonality. 453-456 - Dennis R. Morgan:
An analysis of multiple correlation cancellation loops with a filter in the auxiliary path. 457-461 - C. Y. Chang:
Adaptive multichannel filtering. 462-465 - David C. Farden, Khalid Sayood:
Tracking properties of adaptive signal processing algorithms. 466-469 - Michael J. Coker, Donald N. Simkins:
A nonlinear adaptive noise canceller. 470-473 - Colin F. N. Cowan, H. Martin Reekie, John Mavor, John W. Arthur, Peter B. Denyer:
Miniature CCD-based analog adaptive filters. 474-477 - Arye Nehorai, David Malah:
On the stability and performance of the adaptive line enhancer. 478-481 - Candace M. Anderson:
ALE Gain performance for narrowband signals in white Gaussian noise. 482-485 - John Y. Cheung:
Coherent gain through a frequency domain adaptive LMS algorithm. 486-489
Audio
Psycho and Electro Acoustics
- Douglas Preis:
Measures and perception of phase distortion in electroacoustical systems. 490-493 - W. Marshall Leach Jr.:
The spatial alignment of loudspeaker drivers on a baffle effects on system amplitude and phase responses. 494-497 - Matti Otala:
Conversion of amplitude nonlinearities to phase nonlinearities in feedback audio amplifiers. 498-499 - P. Jeffrey Bloom:
Evaluation of a dereverberation process by normal and impaired listeners. 500-503
Speech
Medium Band Coding - II
- David Malah:
Combined time-domain harmonic compression and CVSD for 7.2 kbit/s transmission of speech signals. 504-507 - Jerry D. Gibson, Louis C. Sauter:
Experimental comparison of forward and backward adaptive prediction in DPCM. 508-511 - Jean-Pierre Adoul, Jean-Louis Debray, Daniel Dalle:
Spectral distance measure applied to the optimum design of DPCM coders with L predictors. 512-515 - Raymond Steele, James D. Johnston:
Slope limiting filters for enhancing noisy channel performance of codecs. 516-519 - R. Viswanathan, William Russell, Alan L. Higgins, Michael G. Berouti, John Makhoul:
Speech-quality optimization of 16 kb/s adaptive predictive coders. 520-525 - Joel A. Feldman, Robert J. McAulay, Elliot Singer:
A split band adaptive predictive coding (SBAPC) speech system. 526-529 - Elliot Singer:
Techniques for improving the robustness of an adaptive predictive coder in the presence of channel errors. 530-534 - Bishnu S. Atal, Manfred R. Schroeder:
Improved quantizer for adaptive predictive coding of speech signals at low bit rates. 535-538 - Neviano Dal Degan, Carlo Scagliola:
Optimal noise shaping in adaptive predictive coding of speech. 539-542 - H. F. Vanlandingham, J. F. Bogdanski Jr.:
An adaptively sampled delta modulator. 543-546 - Heinz G. Fehn, Peter Noll:
Tree and trellis coding of speech and stationary speech-like signals. 547-551
Speech Synthesis
- Keith Blanton:
An efficient method for formant to reflection coefficient conversion. 552-556 - Satoshi Imai, Yoshiharu Abe:
Cepstral synthesis of Japanese from CV syllable parameters. 557-560 - Catherine P. Browman:
Rules for demisyllable synthesis using Lingua, a language interpreter. 561-564 - Marian J. Macchi:
A phonetic dictionary for demisyllabic speech synthesis. 565-567 - Joseph Olive:
A scheme for concatenating units for speech synthesis. 568-571 - David B. Pisoni, Sharon Hunnicutt:
Perceptual evaluation of MITalk: The MIT unrestricted text-to-speech system. 572-575 - Jared Bernstein, David B. Pisoni:
Unlimited text-to-speech system: Description and evaluation of a microprocessor based device. 576-579
Digital Signal Processing
Spectrum Analysis and Parametric Methods
- Otis L. Frost:
High resolution 2-D spectral analysis at low SNR. 580-583 - Lloyd J. Griffiths:
High resolution spectral estimates obtained using data extrapolation. 584-587 - Larry Marple:
Exponential energy spectral density estimation. 588-591 - Donald W. Tufts, Ramdas Kumaresan:
Improved spectral resolution II. 592-597 - James A. Cadzow:
ARMA Spectral estimation: A model equation error procedure. 598-602 - Kenneth Abend:
Spectrum analysis and resolution enhancement by band limited extrapolation. 603-606 - Jean-Pierre Dugré, Louis L. Scharf, A. A. (Louis) Beex:
A note on the measurement of spectral flatness and the calculation of prediction error variances. 607-611 - Dimitri Kazakos, P. Papantoni-Kazakos:
Spectral distance measures between Gaussian processes. 612-613 - Joël Le Roux, Yves Grenier:
An iterative procedure for moving average models estimation. 614-617 - Claude Guéguen, Yves Grenier, F. Giannella:
Factorial linear modelling, algorithms and applications. 618-621 - William L. Mills, Clifford T. Mullis, Richard A. Roberts:
An iterative estimation technique for power spectra by an ARMA model. 622-625 - D. E. Wight, Francis X. Bostick:
Cascade decimation-A technique for real time estimation of power spectra. 626-629
Underwater Acoustics and Adaptive Filters
Detection and Estimation
- J. B. Plant, Yiu Tong Chan:
A two-threshold detector for pulses of unknown duration and doppler. 630-633 - Thomas C. Cantwell, Richard D. Wilmot:
Comparison of sampling techniques for automatic detection of pulse signals with unknown time of arrival. 634-637 - Roger F. Dwyer:
Detection of partitioned signals by discrete cross-spectrum analysis. 638-641 - Jhong S. Lee, Leonard E. Miller:
Detection performance of an FM correlator. 642-645 - Yiu Tong Chan, J. M. Riley, J. B. Plant:
A Wiener filter approach to coherence estimation. 646-649 - John W. Fay:
Confidence bounds for signal-to-noise ratios from magnitude-squared coherence estimates. 650-653 - William P. Whyland:
Reconstruction of discrete-time signals from a subset of weighted DFT outputs. 654-657 - Steven Kay:
Detection of a sinusoid in white noise by autoregressive spectrum analysis. 658-661 - Vijay K. Jain, William L. Collins, David C. Davis:
DFT Interpolation for estimation of tone amplitudes and phases. 662-665
Audio
Sound in Performing Spaces
- James B. Lee:
Information theory and the concert hall problem: The development of tone, ensemble, and diffusion. 670-673 - Theodore J. Schultz:
University centre - A "Slightly surround" concert hall. 674-677 - Jerald R. Hyde, A. Harold Marshall:
Requirements for successful concert hall design: Need for lateral and ensemble reflections. 678-681 - David L. Klepper:
Four outdoor classical music sound reinforcement systems compared. 682-685 - W. J. Gelow, W. Steven Bussey:
Big sound with small things on a medium scale. 686-689 - Wayne R. Lund:
"Return to forever": A touring sound system for concert halls. 690-691
Speech
Speech Enhancement and Noise Reduction
- Steven F. Boll:
Adaptive noise cancelling in speech using the short-time transform. 692-695 - Sidhartha Maitra:
Reducing the effect of background noise for low-bit-rate voice digitizers. 696-698 - Robert J. McAulay, Marilyn L. Malpass:
A real-time noise suppression filter for speech enhancement and robust channel vocoding. 699-702 - William D. Voiers:
Interdependencies among measures of speech intelligility and speech "Quality". 703-705 - Thomas P. Barnwell III:
Correlation analysis of subjective and objective measures for speech quality. 706-709 - Thomas P. Barnwell III:
A comparison of parametrically different objective speech quality measures using correlation analysis with subjective quality results. 710-713 - Roberto Billi, Carlo Scagliola:
An identification method for objective quality measurements on speech waveform coders. 714-718 - J. D. Tomcik, James L. Melsa:
CVSD to LPC conversion using noise tolerant analysis. 719-724 - David Y. Wong:
On understanding the quality problems of LPC speech. 725-728
Digital Signal Processing
Two-Dimensional Digital Filtering
- Russell M. Mersereau, Tae H. Joo, Theresa C. Speake:
A comparison of hexagonally and rectangularly-sampled two-dimensional FIR digital filters. 729-732 - Richard R. Kurth, Michael T. McCallig:
2-D FIR Filter design via semipolynomial approximation. 733-736 - John H. Lodge, Moustafa M. Fahmy:
An efficient ℓp optimization technique for the design of 2-D linear phase FIR digital filters. 737-740 - Dan E. Dudgeon:
An iterative implementation for 2-D digital filters. 741-744 - William H. Haas, Claude S. Lindquist:
A frequency domain approach to synthesizing near optimum edge detection filters. 745-748 - Jean-François Abramatic, S. U. Lee:
Singular value decomposition of 2-D impulse responses. 749-752 - C. H. Reddy, P. Karivaratha Rajan, M. N. S. Swamy:
Studies on N-dimensional filter transfer functions without second kind singularities. 753-757 - Samy A. H. Aly, Moustafa M. Fahmy:
Spatial-domain design of two-dimensional recursive digital filters. 758-761 - Gary A. Shaw, Russell M. Mersereau:
Comparing 2-D recursive and nonrecursive least-square-error approximation filters. 762-765 - S. Y. Hwang:
Computation of correlations in 1-D and 2-D digital signals and systems. 766-769 - Anil K. Jain, Surendra Ranganath:
Image coding by auto regressive synthesis. 770-773
Signal Processing Hardware
- C. S. Joshi, Jack F. McDonald, Randy H. Steinvorth:
A video rate two dimensional FFT processor. 774-777 - Reinder Nouta:
On the development of an integrated circuit for parallel processing of digital filter flow-diagrams. 778-779 - Masud Arjmand, Clifford T. Mullis, Richard A. Roberts:
A modular hardware structure for digital filtering. 780-783 - Robert A. Collesidis, Todd A. Dutton, Joseph R. Fisher:
An ultra-high speed FFT processor. 784-787 - Graham A. Jullien, William C. Miller:
A hardware realization of an NTT convolver using ROM arrays. 788-791 - Fred J. Taylor:
Large moduli multipliers. 792-795 - Shalhav Zohar:
Outline of a fast hardware implementation of Winograd's DFT algorithm. 796-799
Underwater Acoustics and Adaptive Filtering
Time Delay Estimation and Source Location
- Lonnie C. Ludeman:
Multisignal time difference estimator with application to the sound ranging problem. 800-803 - R. Lynn Kirlin:
Improvement of delay measurements from sonar arrays via sequential state estimation. 804-806 - Kent Scarbrough, Nasir Ahmed, G. Clifford Carter:
An experimental comparison of the cross correlation and SCOT techniques for time delay estimation. 807-810 - Joseph C. Hassab, Brian W. Guimond, Steven C. Nardone:
A structure for the combined reduction of bias and variance in estimating source location and motion. 811-817 - Jean-Marc Delosme, Martin Morf, Benjamin Friedlander:
Source location from time differences of arrival: Identifiability and estimation. 818-824 - Jorge I. Galdos, T. Sen Lee:
Nonlinear filtering lower bound evaluation of passive tracking systems. 825-828 - Stephen W. Lang:
Near optimal frequency/Angle of arrival estimates based on maximum entropy spectral techniques. 829-832 - Barry L. Clark:
A comparative evaluation of several bearings-only tracking filters. 833-847 - Webster P. Dove, Alan V. Oppenheim:
Event location using recursive least squares signal processing. 848-850
Speech
Hardware for Speech Processing
- Matti Karjalainen, Unto K. Laine, Raimo Toivonen, Ken R. Haymond, R. Jerome Folmar, Joe Wood:
Aids for the handicapped based on "Synte 2" speech synthesizer. 851-854 - C. J. M. Hodges, Thomas P. Barnwell III, Daniel McWhorter:
The implementation of an all digital speech synthesizer using a multimicroprocessor architecture. 855-858 - Michael McLaughlin, Frank Hudziak, Ira A. Gerson, Kevin L. Kloker:
High performance processor for real-time speech applications. 859-863 - Guy Hochgesang, Robert V. Lemay, Harvey F. Silverman:
The attached processor for speech. 864-867 - James L. Caldwell:
Programmable synthesis using a new "Speech microprocessor". 868-871
Automatic Speech, Language, and Speaker Recognition
- Lalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer:
Further results on the recognition of a continuously read natural corpus. 872-875 - Renato De Mori, Giovanna Giordano:
A parser for segmenting continuous speech into pseudo-syllabic nuclei. 876-879 - Melvyn J. Hunt, Matthew Lennig, Paul Mermelstein:
Experiments in syllable-based recognition of continuous speech. 880-883 - K. P. Li, T. J. Edwards:
Statistical models for automatic language identification. 884-887 - Yves Grenier:
Speaker adaptation through canonical correlation analysis. 888-891 - Jean-Marie Pierrel, Jean-Paul Haton:
Syntactic-Semantic interpretation of sentences in the MYRTILLE II speech understanding system. 892-895 - B. Groc, Denis Tuffelli:
A continuous speech recognition system for data base consultation. 896-899 - Yutaka Kobayashi, Yasuhisa Niimi:
Word boundary detection by pitch contours in an artificial language. 900-903 - M. R. Baraniecki, Malayappan Shridhar:
A speaker verification algorithm for speech utterances corrupted by noise with unknown statistics. 904-907 - Robert E. Wohlford, Edwin H. Wrench Jr., B. Patrick Landell:
A comparison of four techniques for automatic speaker recognition. 908-911
Digital Signal Processing
Architecture / Software
- Fred Mintzer:
Attributes of parallel and cascade microprocessor implementations of digital signal processing. 912-915 - Markku Renfors, Yrjö Neuvo:
Fast multiprocessor realizations of digital filters. 916-919 - Nicholas Roethe, Joachim Lenzer:
An extensible high level language signal processor. 920-923 - Elwood L. Seifert, Frank Cornett:
Architecture analysis for a communications signal processor. 924-926 - Ken Davies:
Why not a high level assembly language? 927-930 - Gervasio Prado, R. K. Pearson:
Interactive software systems for digital signal processing applications. 931-934 - Robert B. Fisher III, Allen M. Peterson:
A facility for interactive digital signal processing. 935-938 - Joachim Lenzer, Gerald Wieber:
On design strategies for parallel algorithms in signal processing using graph models. 939-942
Algorithms, Deconvolution and Coding
- Carey D. Bunks, Douglas Preis:
Minimax time-domain deconvolution for transversal filter equalizers. 943-946 - Gérard Thomas:
Application of the optimal control theory to the deconvolution problem. 947-949 - Dietmar Achilles:
Deconvolution algorithms based on spline interpolation. 950-953 - Martin Morf:
Doubling algorithms for Toeplitz and related equations. 954-959 - Leah J. Siegel:
Parallel processing algorithms for linear predictive coding. 960-963 - Abraham Peled, Antonio Ruiz:
Frequency domain data transmission using reduced computational complexity algorithms. 964-967 - G. Robert Redinbo, Wai Yuen Cheung:
Signal processing techniques in error control systems. 968-973 - Robert J. Fontana:
Universal coding for quasi-stationary processes. 974-977
Underwater Acoustics and Adaptive Filtering
Adaptive Filters - II
- Carey Gibson, Simon Haykin:
A comparison of algorithms for the calculation of adaptive lattice filters. 978-983 - Michael L. Honig, David G. Messerschmitt:
Convergence properties of an adaptive digital lattice filter. 984-988 - Dae Hee Youn, Nasir Ahmed:
Frequency domain considerations of an adaptive escalator predictor. 989-992 - Loren R. McMurray:
Rapid detection of weak signals using adaptive recursive filter weights. 993-996 - John R. Treichler, Michael G. Larimore, C. Richard Johnson Jr.:
On the convergence properties of the simple hyperstable adaptive recursive filter (SHARF). 997-1000 - C. Richard Johnson Jr.:
A stable family of adaptive IIR filters. 1001-1004 - Daniel T. L. Lee, Martin Morf:
Recursive square-root ladder estimation algorithms. 1005-1017 - David L. Soldan, Nasir Ahmed, Samuel D. Stearns:
On using the sequential regression (SER) algorithm for long-term signal processing. 1018-1021 - Eli Fogel, Y. F. Huang:
Adaptive algorithms for non-stastical parameter estimation in linear models. 1022-1025
Underwater Acoustics and Adaptive Filtering
The Acoustic Medium
- Richard B. Lauer:
Wavenumber acoustics: Passive localization and multipath decomposition. 1026-1029
Audio
Psycho and Electro Acoustics
- J. Robert Ashley:
Group and phase delay requirements for loudspeaker systems. 1030-1033 - John Charles Cox:
Time delay effects on speech intelligibility. 1034-1036 - Brian Atkinson:
On the use of operational amplifiers in loudspeaker analogs. 1037-1039 - Martha E. Hean:
Noise pollution, phase VI: The effects of the onset of industrial noise on the physiological and psyshological functions. 1040
Late Papers
- Miron Derkach:
Deductive approach to automatic recognition of Russian spoken sentences. 1041-1044 - Mary Jane Irwin:
Reduction of broadband noise in speech by spectral weighting. 1045-1051 - Bharat B. Madan:
A rapidly converging algorithm for adaptive beam forming. 1052-1055 - Hans Gethöffer:
SIPROL: A high level language for digital signal processing. 1056-1059 - Sadaoki Furui, Aaron E. Rosenberg:
Experimental studies in a new automatic speaker verification system using telephone speech. 1060-1062
manage site settings
To protect your privacy, all features that rely on external API calls from your browser are turned off by default. You need to opt-in for them to become active. All settings here will be stored as cookies with your web browser. For more information see our F.A.Q.
last updated on 2024-10-31 20:20 CET by the dblp team
all metadata released as open data under CC0 1.0 license
see also: Terms of Use | Privacy Policy | Imprint
dblp was originally created in 1993 at:
since 2018, dblp has been operated and maintained by:
the dblp computer science bibliography is funded and supported by:
