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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