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ORCIDICASSP 1984: San Diego, California, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '84, San Diego, California, USA, March 19-21, 1984. IEEE 1984
Narrowband and Low Rate Speech Coding
- Salim E. Roucos, Alexander MacLeod Wilgus:
Speaker normalization algorithms for very-low-rate speech coding. 1-4 - Gérard Benbassat, Xavier Delon:
Low bit rate speech coding by concatenation of sound units and prosody coding. 5-8 - Sharad Singhal, Bishnu S. Atal:
Improving performance of multi-pulse LPC coders at low bit rates. 9-12 - Vijay K. Jain, R. Hangartner:
Efficient algorithm for multi-pulse LPC analysis of speech. 13-16 - A. Parker, S. Thomas Alexander, Henry J. Trussell:
Low bit rate speech enhancement using a new method of multiple impulse excitation. 17-20 - Per Hedelin:
A glottal LPC-vocoder. 21-24 - George S. Kang, Stephanie Everett:
Improvement of the narrowband LPC synthesis. 25-28 - Matthew I. Noah:
Optimal Lloyd-Max quantization of LPC speech parameters. 29-32 - J. Koljonen, Matti Karjalainen:
Use of computational psychoacoustical models in speech processing: Coding and objective performance evaluation. 33-36 - Frank K. Soong, Biing-Hwang Juang:
Line spectrum pair (LSP) and speech data compression. 37-40 - Chia-Chuan Hsiao, Robert W. Brodersen:
A multirate root LPC speech synthesizer. 41-44 - Jean-Pierre Adoul, Claude Lamblin, Alain Le Guyader:
Baseband speech coding at 2400 bps using "Spherical vector quantization". 45-48
Speech Analysis and Synthesis
- Osamu Kakusho, Masuzo Yanagida, Riichiro Mizoguchi:
A sample selective linear prediction analysis of speech. 49-52 - Hynek Hermansky, Hiroya Fujisaki, Yasuo Sato:
Spectral envelope sampling and interpolation in linear predictive analysis of speech. 53-56 - Z. Shpiro, David Malah:
An algebraic approach to discrete short-time Fourier transform analysis and synthesis. 57-60 - Daniel W. Griffin, Douglas S. Deadrick, Jae S. Lim:
Speech synthesis from short-time Fourier transform magnitude and its application to speech processing. 61-64 - Melvyn J. Hunt, C. P. Swail:
Time alignment of natural speech to synthetic speech. 65-68 - P. Jeffrey Bloom:
Use of dynamic programming for automatic synchronization of two similar speech signals. 69-72 - Hong C. Leung, Victor W. Zue:
A procedure for automatic alignment of phonetic transcriptions with continuous speech. 73-76 - Mark Anderson, Janet B. Pierrehumbert, Mark Liberman:
Synthesis by rule of english intonation patterns. 77-80 - Rodolfo Delmonte, Gian Antonio Mian, Graziano Tisato:
A text-to-speech system for italian. 81-84 - Juan Carlos Olabe, Andrés Santos
, R. Martínez, Elias Munoz-Merino, M. Martínez, A. Quilis, Jared Bernstein:
Real time text-to-speech conversion system for spanish. 85-87 - Roger M. Meli, Frank Fallside:
A system for converting teletext into speech. 88-90 - M. Remmel, T. Tago:
Towards increasing the commercial success of speech synthesizers. 91-93
Adaptive Filtering I
- D. Marginedes:
Fast frequency tracking using an adaptive lattice filter for a vortex flowmeter signal. 94-97 - John R. Treichler, Michael G. Larimore:
A real-arithmetic implementation of the constant modulus algorithm. 98-101 - John Y. Cheung:
Convergence conditions for an optimal solution in adaptive recursive filters. 102-105 - C. R. South, A. V. Lewis:
Extension facilities and performance of an LSI adaptive filter. 106-109 - Neil J. Bershad, Lian Zuo Qu:
On the probability density functions of the weight for the complex scalar LMS adaptive algorithm. 110-113 - Neil J. Bershad, Yuh-Huu Chang:
Time correlation statistics of the LMS adaptive algorithm weights. 114-117 - Fuyun Ling, John G. Proakis:
Nonstationary learning characteristics of least squares adaptive estimation algorithms. 118-121 - Yan Chen:
Stable time- and order-recursive algorithm for the adaptive lattice filter. 122-125 - Cumhur Cengiz Evci, Maurice G. Bellanger:
Characteristics of adaptive filters with leakage. 126-129 - Giovanni L. Sicuranza, Andrea Bucconi, Paolo Mitri:
Adaptive echo cancellation with nonlinear digital filters. 130-133 - Dae Hee Youn, S. Prakash:
On realizations and related algorithms for adaptive linear phase filtering. 134-137
2-D Spectral Estimation
- Ahmet H. Kayran, Sydney R. Parker, D. J. Klich:
Two-dimensional spectral estimation with autoregressive lattice parameters. 138-141 - Nan Miao, Zong-Zhi Chen:
Application of SVD to 2-D spectral estimation. 142-145 - Richard M. Leahy, Constantinos E. Goutis, S. Drossos:
Tomographic and spectral analysis using noise statistics. 146-149 - Govind Sharma, Ramalingam Chellappa:
An iterative algorithm for robust 2-D spectrum estimation. 150-153 - Ramalingam Chellappa, Govind Sharma:
A model based approach for 2-D MEPS analysis. 154-157 - David C. Munson Jr., Jorge L. C. Sanz:
The importance of random phase for image reconstruction from frequency offset Fourier data. 158-161 - Li-He Zou, Bede Liu:
Improvement of resolution and reduction of computation in 2D spectral estimation using decimation. 162-165 - Brian L. Hinman, Jared Bernstein, David H. Staelin:
Short-space Fourier transform image processing. 166-169 - Soo-Chang Pei, Eng-Fong Huang:
2D Discrete cosine transform computation by fast polynomial transform algorithms. 170-173 - D. V. Bhaskar Rao, Sun-Yuan Kung:
A state space approach for the 2-D harmonic retrieval problem. 174-176 - Nuthalapati U. Chowdary, Willem J. D. Steenaart:
A high speed two-dimensional FFT processor. 177-180 - S. Hamid Nawab, Farid U. Dowla, R. T. Lacoss:
A new method for wideband sensor array processing. 181-184
Parameter Estimation I
- Vijay K. Jain:
Accurate pole estimation by modified linear prediction. 185-188 - A. R. Gondeck, Vijay K. Jain:
Comparison of three auto-regressive modeling methods. 189-192 - Farid U. Dowla, Jae S. Lim:
Relationship between maximum-likelihood-method and autoregressive modeling in multidimensional power spectrum estimation. 193-196 - Michael T. Manry:
Parameter estimation algorithms utilizing implicit phase. 197-199 - Hossny El-Sherief:
Adaptive least-squares for parametric spectral estimation and its application to pulse estimation and deconvolution of seismic data. 200-203 - Guaning Su:
Signal subspace analysis and improvement of spectral estimation algorithms. 204-207 - Konstantinos Konstantinides, Kung Yao:
Applications of singular value decomposition to system modeling in signal processing. 208-211 - Finley R. Shapiro, Stephen D. Senturia, David Adler:
The analysis of non-oscillating transients using the covariance method with many different orders. 212-215 - Siew Kok Hui, Yong Ching Lim:
A new spectrum analysis approach using autocorrelation technique and MEM. 216-219 - Tryphon T. Georgiou:
Topological aspects of the caratheodory problem. 220-223
Parameter Estimation II
- Richard J. Vaccaro:
On adaptive implementations of Pisarenko's harmonic retrieval method. 224-227 - Yu Hen Hu, Yao-Cheng Ling:
Constrained lattice structures for harmonic retrieval. 228-231 - Mati Wax, Thomas Kailath:
Determining the number of signals by information theoretic criteria. 232-235 - Carlos H. Muravchik, Martin Morf:
Square root normalized feedback ladder algorithm for the identification of moving average systems. 236-239 - J. E. Hudson:
Fitting sinusoids to sampled data using a hybrid S/Z plane moment method. 240-243 - Robert D. Preuss, Rao K. Yarlagadda:
Autoregressive spectral estimation in noise with application to speech analysis. 244-247 - H. Madala:
A new harmonical algorithm for digital signal processing. 248-251 - Neil J. Malloy:
Non-uniform sampling for high resolution spectrum analysis. 252-255 - Fred D. Powell:
Sampling a broad band sparse spectrum without anti-aliassing filters. 256-258 - Koh-ichi Hashimoto, Akira Sano:
On-line trend detection based on ARI modeling. 259-262
Detection and Estimation
- Eugenio J. Tacconi, Michel Bouvet, Bernard C. Picinbono:
A new N.A.R power estimation for adaptive detection. 263-266 - S. L. Earp, Loren W. Nolte:
Multichannel adaptive array processing for optimal detection. 267-270 - Johann F. Böhme:
Estimation of source parameters by maximum likelihood and nonlinear regression. 271-274 - Ratnam V. Raja Kumar, Ranendra N. Pal:
Separation of sinusoids using the constrained adaptive line enhancer. 275-278 - Hong Wang, Mostafa Kaveh:
Estimation of angles-of-arrival for wideband sources. 279-282 - Albert A. Gerlach:
Performance characteristics of biased estimators. 283-286 - D. E. Ohlms, M. L. Hampton:
Track likelihood statistic for the I Q locked loop. 287-289 - Frank Cornett:
Kalman estimation of frequency-domain interference reduction coefficients for angle-modulated signals. 290-293 - Roger F. Dwyer:
Essential limitations to signal detection and estimation: An application to the arctic under ice environmental noise problem. 294-297
Design Methodology and Tools
- Gary E. Kopec:
The integrated signal processing system ISP. 298-301 - S. N. Terepin:
A graphical notation for describing data flow in digital filters. 302-305 - D. A. Schwartz, Thomas P. Barnwell III:
A graph theoretic technique for the generation of systolic implementations for shift-invariant flow graphs. 306-309 - H. V. Jagadish, Thomas Kailath, John A. Newkirk, Robert G. Mathews:
On hardware description from block diagrams. 310-313 - Juan-Manuel Jover, Thomas Kailath:
Design framework for systolic-type arrays. 314-317 - Don H. Johnson:
Signal processing software tools. 318-320 - Brian R. Mears:
A modular method for designing custom signal processing integrated circuits. 321-324 - Clifford T. Mullis, Richard A. Roberts:
Digital processing structures for VLSI implementation. 325-327 - Marc Beacken, Hungwen Li:
Evaluation of signal processing architectures via an integrated simulation environment. 328-331
Isolated-Word Speech Recognition I
- Y. J. Liu:
On creating averaging templates. 332-335 - Vishwa Gupta, Matthew Lennig, Paul Mermelstein:
Decision rules for speaker-independent isolated word recognition. 336-339 - Yasuhiro Matsuda, Shu Tezuka, Mitsuhiko Kanoh, Masafumi Nishimura, Toyohisa Kaneko:
A method for recognizing Japanese monosyllables by using intermediate cumulative distance. 340-343 - David K. Burton, Joseph T. Buck, John E. Shore:
Parameter selection for isolated word recognition using vector quantization. 344-347 - Aaron E. Rosenberg, Kathleen L. Shipley:
Evaluation of an isolated word recognizer in talker-dependent and talker-independent modes using a large telephone-band data base. 348-351 - J. Woodard, Wayne A. Lea:
New measures of performance for speech recognition systems. 352-355 - James K. Baker, Janet M. Baker, Robert Roth, Pard Bamberg:
Cost-effective speech processing. 356-359 - Seppo Haltsonen:
An endpoint relaxation method for dynamic time warping algorithms. 360-363 - B. Yegnanarayana, Sarat Chandran:
Performance of isolated word recognition system for degraded speech. 364-367 - George Vysotsky:
Speaker-independent isolated word recognition using a one-pass analysis. 368-371 - De-Yuan Cheng, Allen Gersho, Bhaskar Ramamurthi, Yair Shoham:
Fast search algorithms for vector quantization and pattern matching. 372-375 - Henry M. Dante:
On the problem of dimensionality and sample size in multi-stage pattern classifiers. 376-379 - Wendy J. Kessler, N. Rex Dixon, Harvey F. Silverman:
An experiment with a non-head-mounted microphone for discrete utterance recognition (DUR). 380-383
Mediumband Speech Coding I
- Michael G. Berouti, H. Garten, Peter Kabal, Paul Mermelstein:
Efficient computation and encoding of the multipulse excitation for LPC. 384-387 - G. A. Senensieb, A. J. Milbourn, A. H. Lloyd, I. M. Warrington:
A non-iterative algorithm for obtaining multi-pulse excitation for linear-predictive speech coders. 388-391 - Isabel Trancoso, Ramón García-Gómez, José M. Tribolet:
A study on short-time phase and multipulse LPC. 392-395 - Peter Kroon, Ed F. Deprettere:
Experimental evaluation of different approaches to the multi-pulse coder. 396-399 - Maurizio Copperi, Daniele Sereno:
9.6 kbit/s Piecewise LPC residual excited coder using multiple-stage vector quantization. 400-403 - Hüseyin Abut, Stephen A. Luse:
Vector quantizers for subband coded waveforms. 404-407 - Allen Gersho, Tor A. Ramstad, I. Versvik:
Fully vector-quantized subband coding with adaptive codebook allocation. 408-411 - Dietrich Wolf, Herbert Reininger, Jürgen Dziumbla:
A comparative study of various speech encoding schemes using vector quantization. 412-415 - Allen Gersho, Yair Shoham:
Hierarchical vector quantization of speech with dynamic codebook allocation. 416-419 - Amine Haoui, David G. Messerschmitt:
Predictive vector quantization. 420-423 - Torbjørn Svendsen:
Tree encoding of the LPC residual. 424-427 - Joël Soumagne, Jean-Pierre Adoul, Sarto Morissette:
A new concept for encoding speech amplitude time quantization. 428-431
Filter Design
- Palghat P. Vaidyanathan, Sanjit K. Mitra:
A new class of very low sensitivity cascade-form digital-filters based on "passive" second order single-input single-output building blocks. 432-435 - James F. Kaiser:
Some properties of a family of generalized time-limited window functions. 436 - Henri J. Nussbaumer, Martin Vetterli:
Computationally efficient QMF filter banks. 437-440 - Marc Beacken:
Efficient implementation of highly variable bandwidth filter banks with highly decimated output channels. 441-443 - Paul C. Millar:
Mirror filters with minimum delay responses for use in subband coders. 444-447 - Hans Wilhelm Schüßler, Peter Steffen:
On the reconstruction of a smooth function out of its samples. 448-451 - Alfred Fettweis, Josef A. Nossek, Klaus Meerkötter:
Reconstruction of signals after filtering and sampling rate reduction. 452-455 - Kenji Nakayama, T. Seki:
A design method for multirate filters. 456-459 - Kazunori Sugahara, Katsuhiko Hayashi, Kotaro Hirano, Sanjit K. Mitra:
N-path digital filters. 460-463 - Sailesh K. Rao, Thomas Kailath:
Pipelined orthogonal digital lattice filters. 464-467 - Adly T. Fam:
A multi-signal bus architecture for FIR filters with single bit coefficients. 468-470 - Yoram Bresler, Albert Macovski:
3-D reconstruction from projections based on dynamic object models. 471-474 - Mehrdad Soumekh, Mostafa Kaveh:
Image reconstruction from frequency domain data on arbitrary contours. 475-478 - Narciso García, Alberto R. Calero:
Faster phase only image reconstruction. 479-482 - R. S. Acharya, Richard A. Robb, Harry Wechsler:
An image reconstruction algorithm for time varying X-ray projection image sequences. 483-486 - Susan R. Curtis, Jae S. Lim, Alan V. Oppenheim:
Signal reconstruction from one bit of Fourier transform phase. 487-490 - A. A. (Louis) Beex:
Soft constraint iterative reconstruction from noisy projections. 491-494 - Anil K. Jain, Siamak Ansari:
Radon transform theory for random fields and optimum image reconstruction from noisy projections. 495-498 - Xinhua Zhuang, Kai-Bor Yu, Robert M. Haralick:
A new approach to the solution of the maximum entropy image reconstruction problem. 499-502 - P. Borghesi, Vito Cappellini, Alberto Del Bimbo, Alessandro Mecocci:
Two special digital processing systems for recognition of moving objects. 503-506 - Steven G. Kratzer, Anthony P. Reeves:
Image restoration by parallel processing. 507-510
Image Reconstruction
- Daniele Pelloni, Roger Lagadec:
Enhancement of audio signals based on digital techniques. 511-514 - A. J. E. M. Janssen, Lodewijk B. Vries:
Interpolation of band-limited discrete-time signals by minimizing out-of-band energy. 515-518 - James Anderson Moorer:
Algorithm design for real-time audio signal processing. 519-522 - Guy W. McNally, Philip S. Gaskell:
Editing digital audio. 523-526 - Wil J. W. Kitzen, P. M. Boers:
Applications of a digital audio-signal processor in T.V. sets. 527-530
Methods for Spectral Estimation
- Benjamin I. Helme, Chrysostomos L. Nikias:
A high-resolution modified Burg algorithm for spectral estimation. 531-534 - Taiho Koh, Edward J. Powers:
Efficient maximum entropy spectral estimation using non-multiplication methods. 535-538 - Giuseppe Martinelli, Gianni Orlandi:
ARMA spectrum estimation by extended lattice. 539-541 - M. Halimi, Francis Castanie:
A comparison of several optimal random quantization algorithms for correlation estimation. 542-545 - James E. Gaby, Monson H. Hayes:
Artificial intelligence applied to spectrum estimation. 546-549 - Rodney W. Johnson, Bruce R. Musicus:
Multichannel relative-entropy spectrum analysis. 550-553 - Boaz Porat, Benjamin Friedlander:
ARMA spectral estimation of time series with missing observations. 554-557 - Rodney W. Johnson, John E. Shore:
Power spectrum estimation by means of relative-entropy minimization with uncertain constraints. 558-561 - Kai-Bor Yu, Xinhua Zhuang, Robert M. Haralick:
Maximum entropy spectrum estimation from noisy correlation measurements. 562-565 - Mustafa Aktar, Bülent Sankur, Yorgo Istefanopulos:
Maximum likelihood and bartlett spectrum estimates. 566-568 - Moeness G. Amin, Lloyd J. Griffiths:
Lag-invariant adaptive spectrum estimation. 569-572
Models and Performance
- Benjamin Friedlander:
On the computation of the Cramer Rao bound for ARMA parameter estimation. 573-576 - Stephen W. Lang:
Confidence regions for spectral bounds. 577-580 - Shaoann Shon, Kishan Mehrotra:
Performance comparison of autoregressive estimation methods. 581-584 - Herbert Gish:
The magnitude squared coherence estimate: A geometric view. 585-587 - Don H. Johnson, Darel A. Linebarger:
Signal processing models for point processes. 588-591 - Juha Karhunen:
Adaptive algorithms for estimating eigenvectors of correlation type matrices. 592-595 - Bernard C. Picinbono, Michel Bouvet:
Complex white noises and autoregressive signals. 596-599 - Thomas P. Bronez, James A. Cadzow:
Signal rank and model order determination. 600-603 - Donald F. Gingras:
On the asymptotic normality of autoregressive spectral density estimates for the noise corrupted case. 604-607 - Miguel Angel Lagunas, Antoni Gasull:
Measuring true spectral density from ML filters (NMLM and q-NMLM spectral estimates). 608-611
Time Delay Estimation
- Kent Scarbrough, G. Clifford Carter, Roger J. Tremblay:
Implications of threshold effects for coherent and incoherent processing techniques of time delay estimation. 612-615 - Yiu Tong Chan, P. A. Yansouni, S. Crozier:
Time delay estimation with low signal to noise ratios. 616-619 - J. P. Ianniello:
Comparison of the Ziv-Zakai lower bound on multipath time delay estimation with autocorrelator performance. 620-623 - Alfred O. Hero III, Stuart C. Schwartz:
Alternatives to the generalized cross correlater for time delay estimation. 624-627 - John W. Betz:
Performance of the deskewed short-time correlator. 628-631 - E. J. Modugno, M. L. Hampton, G. W. Johnson, D. E. Ohlms:
An efficient Doppler compensated correlation and low doppler interference removal. 632-635 - P. A. Yansouni:
Bias free estimation of the variance of time of arrival differences. 636-639 - Chao-Huan Hou:
A new method for the passive locating of unknown source by using a vertical dipole receiver. 640-643 - Julius O. Smith III, Benjamin Friedlander:
Adaptive multipath delay estimation. 644-647 - Dae Hee Youn, Shen-Neng Chiou, V. John Mathews:
Adaptive realization of the phase transform for time delay estimation. 648-651 - Jeffrey L. Krolik, M. Joy, Subbarayan Pasupathy, Moshe Eizenman:
A comparative study of the LMS adaptive filter versus generalized correlation method for time delay estimation. 652-655 - J. Schiller:
Motion parameter estimation of a fast moving sound source using the retardation effect. 656-659
VLSI Digital Signal Processors
- Richard W. Linderman, Peter P. Reusens, Paul M. Chau, Walter H. Ku:
Digital signal processing capabilities of CUSP, a high performance bit-serial VLSI processor. 660-663 - Gottfried Ungerboeck, Dietrich Maiwald, Hans-Peter Kaeser, Pierre R. Chevillat:
The SP16 signal processor. 664-667 - Manfred Immendörfer, Dieter Kopp, Gebhard Thierer:
SASP-A digital signal processor system for speech processing applications. 668-671 - Mohsin M. Jamali, Graham A. Jullien, William C. Miller, S. I. Ahmad:
A real time general purpose signal processor. 672-675 - Matthew Yuschik, Hideaki Kobayashi:
A VLSI digital filter bank. 676-679 - Bob Woo, Lyon Lin, F. A. Ware:
A high-speech 32 bit IEEE floating-point chip set for digital signal processing. 680-683 - J. Nuttall, J. Oxxal:
A 16-bit three port arithmetic logic and shift unit. 684-687 - T. E. Curtis, Anthony G. Constantinides, Y. S. Wu:
VLSI Architecture for signal processing with alternate low-level primitive structures (ALPS). 688-691 - Alois Rainer, Walter Ulbrich, Lajos Gazsi:
Adder-based digital signal processor architecture for 80 NS cycle time. 692-695
Isolated-Word Speech Recognition II
- Lawrence R. Rabiner, M. Mohan Sondhi, Stephen E. Levinson:
A vector quantizer incorporating both LPC shape and energy. 1-4 - Heikki Riittinen:
Short-cut algorithms for the learning subspace method. 5-8 - Jean-Sylvain Liénard, Frank K. Soong:
On the use of transient information in speech recognition. 9-12 - Bernhard R. Kämmerer, Wolfgang A. Küpper, Helmut Lagger:
Special feature vector coding and appropriate distance definition developed for a speech recognition system. 13-16 - S. Raman, B. Yegnanarayana:
Performance of isolated word recognition system for confusable vocabulary. 17-20 - Moshé J. Lasry, Richard M. Stern:
Unsupervised adaptation to new speakers in feature-based letter recognition. 21-24 - Patrick Fonsale:
Feature-based speaker-independent word recognition without oral learning. 25-28 - Shozo Makino, Ken'iti Kido:
A speaker independent word recognition system based on phoneme recognition for a large size (212 words) vocabulary. 29-32 - Mats Blomberg, Rolf Carlson, Kjell Elenius, Björn Granström:
Auditory models in isolated word recognition. 33-36 - Ioannis Dologlou, Jean Marc Dolmazon:
Comparison of a model of the peripheral auditory system and L.P.C. analysis in a speech recognition system. 37-40 - Marcia A. Bush, Gary E. Kopec, Niels Lauritzen:
Segmentation in isolated word recognition using vector quantization. 41-44 - Takeshi Fukabayashi, Chiu-Kuang Chuang:
Speech segmentation and recognition using adaptive linear prediction algorithm. 45-48
Speech Enhancement
- Thomas E. Eger, James Su, L. William Varner:
A nonlinear spectrum processing technique for speech enhancement. 49-52 - Jack E. Porter, Steven F. Boll:
Optimal estimators for spectral restoration of noisy speech. 53-56 - Vishu Viswanathan, Kenneth F. Karnofsky, Kenneth N. Stevens, Michael N. Alakel:
Multisensor speech input for enhanced immunity to acoustic background noise. 57-60 - William Harrison, Jae S. Lim, Elliot Singer:
Adaptive noise cancellation in a fighter cockpit environment. 61-64 - Brian A. Hanson, David Y. Wong:
The harmonic magnitude suppression (EMS) technique for intelligibility enhancement in the presence of interfering speech. 65-68 - Mitchel Weintraub:
The GRASP sound separation system. 69-72
Pitch Detection and Speaker Identification
- Wolfgang J. Hess, Helge Indefrey:
Accurate pitch determination of speech signals by means of a laryngograph. 73-76 - Philippe Specker:
A powerful post-processing algorithm for time-domain pitch trackers. 77-80 - Denis Tuffelli:
A pitch detection algorithm with hypothesis and test strategy by means of fast surface AMDF. 81-84 - Kyong-Ae Oh, Chong Kwan Un:
A performance comparison of pitch extraction algorithms for noisy speech. 85-88 - Michael A. Krasner, Jared J. Wolf, Kenneth F. Karnofsky, Richard M. Schwartz, Salim E. Roucos, Herbert Gish:
Investigation of text-independent speaker identification techniques under conditions of variable data. 89-92 - Panos E. Papamichalis, George R. Doddington:
A speaker recognizability test. 93-96 - Chieh Tsao, Robert M. Gray:
An endpoint detector for LPC speech using residual error look-ahead for vector quantization applications. 97-100
Audio Processing
- Scott Robertson, William Feger, Richard K. Hester:
Analog interface chips for audio band digital signal processing. 101-104 - James H. Snyder:
Experimental hardware for real-time wideband speech coding. 105-107 - Richard V. Cox, Jeffrey Snyder, Ronald E. Crochiere, D. Bock, James D. Johnston:
Testing of wideband digital coders. 108-111 - Julius O. Smith III, Phil Gossett:
A flexible sampling-rate conversion method. 112-115 - To Russell Hsing:
A new digital voice summing technique for teleconferencing. 116-119 - George S. Kang, Mark L. Lidd:
Automatic gain control. 120-123 - Sally G. Smith:
Modelling musical instruments in the digital domain. 124-127 - David R. Fischell, Cecil H. Coker:
A speech direction finder. 128-131 - Matti Karjalainen:
Sound quality measurements of audio systems based on models of auditory perception. 132-135 - John Charles Cox:
The minimum detectable delay of speech and music. 136-139 - Michael J. Ready, V. Ralph Algazi:
Application of a simple hearing model to the design of audio filters. 140-143 - J. Robert Ashley:
Requirements for loudspeaker crossover networks. 144-147
Design and Stability
- Winser E. Alexander:
Initial condition transient suppression for two dimensional recursive digital filters. 148-151 - George A. Lampropoulos, Moustafa M. Fahmy:
On 2-D FIR and IIR filter design. 152-155 - Roy Chapman, Tariq S. Durrani:
Circularly symmetric filter design using 2D prolate spheroidal sequences. 156-159 - Anastasios N. Venetsanopoulos, Chrysostomos L. Nikias:
Realization of two-dimensional digital filters by LU decomposition of their transfer function. 160-163 - Roland Wilson:
Uncertainty, eigenvalue problems and filter design. 164-167 - M. N. S. Swamy, Leonid M. Roytman, Eugene I. Plotkin:
Computation of the threshold of stability for N-dimensional digital filters. 168-170 - Hon Keung Kwan:
Two-dimensional passive state-space digital filter design. 171-174 - Marek Domanski:
Structural pseudolosslessness and structural pseudopassivity of 1-D and 2-D digital filters. 175-178 - M. N. S. Swamy, Leonid M. Roytman, Eugene I. Plotkin:
Two stability tests for two-dimensional digital filters. 179-182 - M. Omair Ahmad, Majid Ahmadi, Venkat Ramachandran:
Transfer function realization of a class of doubly-terminated two-variable lossless networks and their application in linear-phase 2-dimensional digital filter design. 183-186 - Gonzalo R. Arce, Regis J. Crinon:
Median filters: Analysis for 2 dimensional recursively filtered signals. 187-190
Adaptive Filtering II
- Bernard Widrow, Eugene Walach:
Adaptive signal processing for adaptive control. 191-194 - R. A. David:
Detection of multiple sinusoids using a parallel ALE. 195-198 - Nasir Ahmed, Don R. Hush, G. R. Elliott, R. Joseph Fogler:
Detection of multiple sinusoids using an adaptive cascaded structure. 199-202 - John R. Treichler:
Adaptive algorithms that restore signal properties. 203-206 - Otis M. Soloman, James A. Cadzow, Samuel D. Stearns:
A parametric method for computing magnitude squared coherence. 207-210 - Michael G. Larimore, John R. Treichler:
Suppression of FM adjacent channel interference. 211-214 - L. A. Poole, G. E. Warnaka, R. C. Cutter:
The implementation of digital filters using a modified Widrow-Hoff algorithm for the adaptive cancellation of acoustic noise. 215-218 - S. Lawrence Marple Jr.:
A fast least squares linear phase adaptive filter. 219-222
Signals with Time-Varying Parameters
- Chrysostomos L. Nikias:
A new realization scheme of periodically time-varying digital filters. 223-226 - Ken C. Sharman, Benjamin Friedlander:
Time-varying autoregressive modeling of a class of nonstationary signals. 227-230 - Gloria Faye Boudreaux-Bartels, Thomas W. Parks:
Signal estimation using modified Wigner distributions. 231-234 - Y. Zhou, Richard L. Frost, Craig K. Rushforth:
The design of well-conditioned discrete imaging systems. 235-238 - Boualem Bouachache, Francisco Rodriguez:
Recognition of time-varying signals in the time-frequency domain by means of the Wigner distribution. 239-242 - Evangelos E. Milios:
Fast sequential least-squares processing. 243-246 - Manuel Duarte Ortigueira, José M. Tribolet:
A framework for the evaluation of spectral analysis techniques. 247-250
Image Processing I
- E. Russell Ritenour, T. R. Nelson, U. Raff:
Applications of the median filter to digital radiographic images. 251-254 - Thomas A. Nodes, Guang-Yu Liao, Neal C. Gallagher Jr.:
Statistical analysis of two dimensional median filtered images. 255-258 - Subrahmanyam Dravida, John W. Woods, W. Shen:
A comparison of image filtering algorithms. 259-262 - James V. Aanstoos, W. Howard Ruedger, B. D. Meredith, M. E. Beatty III:
Algorithms for the simulation of geometric distortion in a satellite imaging system. 263-266 - B. L. Yen, Thomas S. Huang:
Solving for 3D motion parameters of a rigid body by a vector geometrical approach: Uniqueness and numerical results. 267-270 - Margie H. Groves, Sarah A. Rajala, Wesley E. Snyder:
Calculation of displacement fields by means of the motion detection transform. 271-274 - Jan P. Allebach:
Filtering and correlation of time-sequentially sampled spatiotemporal signals. 275-278 - David Cyganski, John A. Orr:
Object identification and orientation determination in 3-space with no point correspondence information. 279-282 - Serge Castan, Jun Shen:
A method for finding straight line and plane correspondences in stereopair images. 283-286 - K. Mike Tao:
Adaptive image smoothing algorithms for edge and texture preservation. 287-290
Geophysics and Channel Modelling
- Tariq S. Durrani, J. L. Bowie:
New results in seismic deconvolution using lattice processors. 291-294 - A. M. Bisbee, Edmund A. Quincy, Daniel J. Tomich:
Beam-steered vertical seismic arrays for wave classification. 295-298 - David J. Scheibner, Thomas W. Parks:
Slowness aliasing in the Radon transform. 299-302 - C. Sherwin, W. V. McCollough, D. D. McCrady, S. Johnson:
A model for sonar bottom sounding processing. 303-306 - William S. Hodgkiss, Richard K. Brienzo:
Coherent recombination of sediment borne and water path acoustic signals. 307-310 - Changxue Ren, Chengqi Xu:
The influence of Doppler effect on information transmission in sound channel. 311-314 - Kenneth B. Theriault:
Spatial properties of pulsed-Doppler current profiling systems. 315-318 - Marwan A. Simaan:
A frequency domain method for time-shift estimation and alignment of discrete time signals. 319-322 - Chi Hau Chen:
Characterization and recognition of underwater transient signals. 323-326 - Fred Harris:
Observer adaptive PRF for synthetic aperture radar imaging. 327-330
Architectures/Algorithms
- Leah J. Siegel:
Highly parallel architectures and algorithms for speech analysis. 331-334 - Sun-Yuan Kung:
An algorithm basis for systolic/Wavefront array software. 335-338 - R. Yoho, Douglas Preis:
Systolic architectures for deconvolution algorithms. 339-342 - Peter R. Cappello, Kenneth Steiglitz:
A fast tally structure and applications to signal processing. 343-346 - Earl E. Swartzlander Jr., George Hallnor:
Fast transform processor implementation. 347-350 - M. R. Baraniecki, Ramdas Kumaresan, Anna Z. Baraniecki, Malayappan Shridhar:
Multiprocessor system for speech processing and telecommunications. 351-354
VLSI Speech Processors
- Thomas S. Anantharaman, M. Annaratone, Roberto Bisiani:
A family of custom VLSI circuits for speech recognition. 355-358 - Young-Hwan Oh, John G. Ackenhusen, L. M. Breda, L. F. Rosa, Michael K. Brown, Les T. Niles:
Architecture for a real-time LPC-based feature measurement integrated circuit. 359-362 - Patrice Frison, Patrice Quinton:
A VLSI parallel machine for speech recognition. 363-366 - Joel A. Feldman, Steven L. Garverick, F. Matthew Rhodes, James R. Mann:
A wafer scale integration systolic processor for connected word recognition. 367-370 - Michael K. Brown, Reed Thorkildsen, Young-Hwan Oh, Syed S. Ali:
The DTWP: An LPC based dynamic time warping processor for isolated word recognition. 371-374 - Robert Kavaler, Robert W. Brodersen, Tobias G. Noll, Menahem Lowy, Hy Murveit:
A dynamic time warp IC for a one thousand word recognition system. 375-378
Large Vocabulary Speech Recognition
- Anne-Marie Derouault, Bernard Mérialdo:
Recognition complexity with large vocabulary. 379-382 - Yi-Teh Lee, Harvey F. Silverman, N. Rex Dixon:
Preliminary results for an operational definition and methodology for predicting large vocabulary DUR confusability from phonetic transcriptions. 383-386 - Alex Waibel:
Suprasegmentals in very large vocabulary isolated word recognition. 387-390 - Daniel P. Huttenlocher, Victor W. Zue:
A model of lexical access from partial phonetic information. 391-394 - Lalit R. Bahl, Subrata K. Das, Peter V. de Souza, Frederick Jelinek, Slava M. Katz, Robert L. Mercer, Michael A. Picheny:
Some experiments with large-vocabulary isolated-word sentence recognition. 395-396 - Jean-François Mari, Jean Paul Haton:
Some experiments in automatic recognition of a thousand word vocabulary. 397-400 - Jared Bernstein, Richard Becker, David Elliott Bell, Hy Murveit, Fausto Poza, G. Stevens:
Telephone communication between deaf and hearing persons. 401-404 - Yuji Kijima, Yasuhiro Nara, Atsuhito Kobayashi, Shinta Kimura:
Speaker adaptation in large-vocabulary voice recognition. 405-408 - Katsuhiko Shirai, Tetsunori Kobayashi:
Phrase speech recognition of large vocabulary using feature in articulatory domain. 409-412 - Hervé Bourlard, Christian Wellekens, Hermann Ney:
Connected digit recognition using vector quantization. 413-416 - B. Patrick Landell, J. A. Naylor, Robert E. Wohlford:
Effect of vector quantization on a continuous speech recognition system. 417-420
Mediumband Speech Coding II
- Mark J. T. Smith, Thomas P. Barnwell III:
A procedure for designing exact reconstruction filter banks for tree-structured subband coders. 421-424 - Michael D. Cowing, Jesse W. Fussell:
16 kbps APC with hybrid quantization. 425-428 - Randall C. Reininger, Jerry D. Gibson:
Backward adaptive lattice and transversl predictors for ADPCM. 429-432 - Cumhur Cengiz Evci, P. J. Patrick:
Performance analysis of DPCM codecs operating on adaptively frequency mapped 7.6 khz speech. 433-436 - Luís B. Almeida, Fernando M. Silva:
Variable-frequency synthesis: An improved harmonic coding scheme. 437-440 - Robert J. McAulay, Thomas F. Quatieri:
Magnitude-only reconstruction using a sinusoidal speech modelMagnitude-only reconstruction using a sinusoidal speech model. 441-444 - Yasuhiko Arai, Toshio Yagi, Takeshi Miyagawa:
Peak-and-valley impulse excited LPC voice. 445-448 - Mark Dankberg, Ronald A. Iltis, David Saxton, Phillip Wilson:
Implementation of the RELP vocoder using the TMS320. 449-452 - Wanda Gass, Masud Arjmand:
Real-time 9600 bits/sec speech coding on the TI Professional Computer. 453-456 - Claude R. Galand, Chantal Couturier, Guy Platel, Robert Vermot-Gauchy:
Voice excited predictive coder (VEPC) implementation on 10 MIPS signal processor. 457-460 - S. Tanaka, Y. Wake, Takashi Araseki, T. Sato, K. Kage:
A 16kbps APC codec using digital signal processors. 461-464 - R. B. Hanes, F. A. Westall, C. Goody:
A 16 kbit/s speech coder using a single DSP device. 465-468
DFT and FFT Algorithms
- Junho Choi, J. R. Johnson:
A modified discrete fourier transform algorithm: A delta rotation algorithm. 469-472 - Howard W. Johnson, C. Sidney Burrus:
An in-order, in-place radix-2 FFT. 473-476 - Byeong Gi Lee:
FCT - A fact cosine transform. 477-480 - Nasser M. Nasrabadi, Robert A. King:
Complex number theoretic transform in p-adic field. 481-483 - Michael A. Soderstrand, Gregory D. Poe:
Application of Quadratic-Like complex residue number system arithmetic to ultrasonics. 484-487 - J. C. Weed, R. C. Polge:
An efficent implementation of a hexagonal FFT. 488-491 - Michael T. Heideman, C. Sidney Burrus, Howard W. Johnson:
Prime factor FFT algorithms for real-valued series. 492-495
Deconvolution
- H. Joel Trussell, M. Reha Civanlar:
Signal deconvolution by projection onto convex sets. 496-499 - Chong-Yung Chi, Jerry M. Mendel:
Performance of minimum-variance deconvolution filter. 500-503 - Guy Demoment, Roger Reynaud, Alain Herment:
Fast minimum variance deconvolution. 504-507 - Joe K. Hammond, R. F. Harrison:
modelling and deconvolution of nonstationary acoustic signals from moving sources using a convariance equivalent formulation. 508-511 - Eric W. Hansen, Phaih-Lan Law:
Abel inversion by Kalman filtering. 512-515 - Samir Akkouche, Gérard Thomas:
Optimal constrained restoration and dynamic programming. 516-518
Image Coding
- Didier Perny:
Compression of underwater images for their transmission on a low bit rate acoustic channel. 519-522 - Petros Maragos, Ronald W. Schafer:
Morphological skeleton representation and coding of binary images. 523-526 - Roland Wilson:
Quad-tree predictive coding: A new class of image data compression algorithms. 527-530 - Alan S. Kwabwe, Robert A. King:
Recovery of gray scale images from approximations of their contours. 531-534 - Petros Maragos, Russell M. Mersereau, Ronald W. Schafer:
Multichannel linear predictive coding of color images. 535-538 - Paul R. Boucher, Morris Goldberg:
Color image compression by adaptive vector quantization. 539-542 - R. Srinivasan, K. R. Rao:
CMT and hybrid coding of the component color TV signal. 543-546 - J. P. Belan, B. Choquet:
Solid state colour camera simulator: Preliminary results. 547-550 - Nasser M. Nasrabadi, Robert A. King:
A new image coding technique using transforms vector quantization. 551-554 - Wai-Kuen Cham, D. Allott, Roger J. Clarke:
Block classification image coding with combined transforms. 555-558 - Priyadarshan Jakatdar, William A. Pearlman:
An optimal transform tree coding method applied to images. 559-562
Quantization Effects
- Pierre Siohan, Acyl Benslimane:
Design of optimal finite wordlength linear phase FIR filters: New applications. 563-566 - Madeleine Bonnet, Odile Macchi:
An echo canceller having reduced word size taps and using the sign algorithm with extra controlled noise. 567-570 - Fuyun Ling, John G. Proakis:
Numerical accuracy and stability: Two problems of adaptive estimation algorithms caused by round-off error. 571-574 - Masaru Sakurai, Junzo Murakami:
Reduction of quantization effects in adaptive filters. 575-578 - Douglas T. Sherwood, Neil J. Bershad:
Accumulator quantization noise in the single-weight complex LMS adaptive algorithm. 579-582 - Mark W. Smith, David C. Farden:
Statistical design of cascade finite wordlength FIR digital filters. 583-585 - Francis Castanie, D. Wan:
Application of the random reference quantization principle to radix-2 FFT computation. 586-589 - James W. Modestino, Nariman Farvardin:
Optimum entropy-coded quantizer design for a class of discrete-time sources. 590-593 - Erik I. Verriest:
On redefining the optimal least squares filter under floating point operations. 594-597 - Varadaraj P. Shenoy, Fred J. Taylor:
Error analysis of LMS adaptive digital filter implemented with logarithmic number system. 598-601 - Rajeev Jain, Joos Vandewalle, Hugo De Man:
Efficient CAD tools for the coefficient optimisation of arbitrary integrated digital filters. 602-605 - Tokahiro Yamaguchi, Norio Arakawa:
Effects of finite Kernel word length in signal processing. 606-609 - D. A. Schwartz, Thomas P. Barnwell III:
Increasing the parallelism of filters through transformation to block state variable form. 610-613 - John Mark, Alison Brown, Tony Matthews:
Quantization reduction for evaluating laser gyro performance using a moving average filter. 614-617
Signal Analysis and Restoration
- David M. Thomas, Monson H. Hayes:
Procedures for signal reconstruction from noisy phase. 618-621 - Thomas J. Brzustowicz, Rui J. P. de Figueiredo:
Iterative restoration by generalized replacement. 622-625 - Barry J. Sullivan, Bede Liu:
Extrapolation of discrete-time band-limited signals using singular value decomposition with decimation. 626-629 - A. A. (Louis) Beex, K. A. Becker:
Spectral estimation and matched filter performance. 630-633 - P. Davies, Joe K. Hammond:
Envelope modelling of impulse response functions of systems having a high modal density. 634-637 - Jorge L. C. Sanz, Thomas S. Huang:
Band-limited signal extrapolation in the presence of noise. 638-641 - Costas E. Goutis, Richard M. Leahy, P. G. Cassidy:
Spectra using data distribution and covariance modelling. 642-645 - Long-bin Ling, Rui Yin, Xinghua Wang:
Nonlinear filters for reducing spiky noise: 2-dimension. 646-649 - M. J. Tsai, D. A. O'Connor:
Experiments with extrapolation of band-limited signal. 650-653 - Manell Zakharia:
Relationship between Paley-Wiener theorem and the stationary phase method? 654-657 - Masuzo Yanagida, Osamu Kakusho:
Difference operation for Pre/Post-emphasis in linear prediction analysis. 658-661
Image Analysis
- Serge Castan, Jun Shen:
A one-source photometric method for N-order specular surfaces. 662-665 - Rangasami L. Kashyap, Paul M. Lapsa:
Two dimensional stochastic fractional AR models with strong periodicities. 666-669 - Michael D. Richard, Charles W. Therrien, Jae S. Lim:
Image segmentation using spatial linear prediction. 670-673 - Jay B. Jordan, Lonnie C. Ludeman:
Image segmentation using maximum entropy techniques. 674-677 - Howard Elliott, Haluk Derin, Roberto Cristi, Donald Geman:
Application of the Gibbs distribution to image segmentation. 678-681 - Haluk Derin, Howard Elliott, Roberto Cristi, Donald Geman:
Bayes smoothing algorithms for segmentation of images modeled by Markov random fields. 682-685 - J. E. Bevington, Russell M. Mersereau:
A maximum-likelihood approach to image segmentation by texture. 686-689 - N. Rajendran, Maher A. Sid-Ahmed, James J. Soltis:
Multi-level thresholding and its application to feature extraction in machine parts. 690-693 - Rama Chellappa, Sangit Chatterjee:
Classification of textures using Markov random field models. 694-697 - Bhaskar Ramamurthi, Allen Gersho:
Image vector quantization with a perceptually-based cell classifier. 698-701 - Richard A. Jones, Yogendra J. Tejwani:
Machine recognition of partial planar shapes using feature vectors. 702-705 - Sally L. Wood:
Segmentation of computerized tomography images for three dimensional analysis. 706-709
Underwater Acoustics: Arrays I
- Bodo Scholz, Wolfgang Kroll:
Bearing estimation in a shallow water environment using the eigenvector decomposition method. 710-713 - Georges Bienvenu, Laurent Kopp:
Decreasing high resolution method sensitivity by conventional beamformer preprocessing. 714-717 - Todd K. Citron, Thomas Kailath:
An improved eigenvector beamformer. 718-721 - Ang-Zhao Di, Li-Sheng Tian:
Matrix decomposition and multiple source location. 722-725 - Tie-Jun Shan, Thomas Kailath:
New adaptive processor for coherent signals and interference. 726-729 - Stanislav B. Kesler, Samson Boodaghians, Jelisaveta Kesler:
Resolution of incoherent and coherent sources by autoregressive beamforming. 730-733 - William S. Hodgkiss, Dimitrios Alexandrou:
Sea surface reverberation rejection. 734-737 - Chi Chung Ko:
Structure of covariance matrix and eigenvalues of broadband tapped delay line adaptive array. 738-741 - Frances A. Reed, Curtis M. Flynn, Paul L. Feintuch:
The effect of interference extent on LMS spatial cancellation. 742-745 - Wolfgang Kroll, Bodo Scholz:
Application of the maximum entropy beamformer to a shallow water line array. 746-749 - Leon H. Sibul:
Application of singular value decomposition to adaptive beamforming. 750-753 - Ezio G. Pusone, Lewis J. Lloyd:
Synthetic aperture sonar: An analysis of beamforming and system design. 754-757 - G. E. Martin:
Degradation of angular resolution for eigenvector-eigenvalue (EVEV) high-resolution processors with inadequate estimation of noise coherence. 758-761
Systolic Arrays
- B. R. Mercy:
Systolic array technique applied to symmetric FIR filters. 762-765 - Ping Ang, Martin Morf:
Concurrent array processor for fast eigenvalue computations. 766-769 - John G. McWhirter, C. R. Ward, Andrew J. Robson, P. J. Hargrave:
The application of a systolic least squares processing array to adaptive beamforming. 770-773 - T. Willey, Tariq S. Durrani, Roy Chapman:
An FFT systolic processor and its applications. 774-777 - Francis Jutand, Nicolas Demassieux, Dominique Vicard, Gérard Chollet:
VLSI architectures for dynamic time warping using systolic arrays. 778-781 - Richard W. Linderman, Walter H. Ku:
A three dimensional systolic array architecture for fast matrix multiplication. 782-785
Speech Processing Hardware Implementations
- Bruce Fette, Chaz Rimpo, Joseph Kish:
A manpack portable LPC 10 vocoder. 786-789 - S. A. Townes, Trieu-Kien Truong:
The VLSI design of a sub-band coder. 790-793 - Jim W. Burgett, John S. Collura:
Real time implimentation of 16kbs APC with Hybrid Quantization. 794-797 - Bertrand Denoix:
A speech input and processing board for a personal computer. 798-801 - B. P. Tao, M. Oijala:
Architecture for a VLSI implementation of an LPC-based, isolated-word recognition system. 802-805
Connected Speech Recognition
- Nobuo Hataoka, Yoshiaki Asakawa, Akio Komatsu, Akira Ichikawa:
Speaker-independent connected digit recognition. 1-4 - Anna Maria Colla, Donatella Sciarra:
Automatic diphone bootstrapping for speaker-adaptive continuous speech recognition. 5-8 - Francine Chen, Victor W. Zue:
Application of allophonic and lexical constraints in continuous digit recognition. 9-12 - Lawrence R. Rabiner, Jay G. Wilpon, Sandra G. Terrace:
A directory listing retrieval system based on connected letter recognition. 13-16 - M. Cravero, Luciano Fissore, Roberto Pieraccini, Carlo Scagliola:
Syntax driven recognition of connected words by Markov models. 17-20 - Richard M. Schwartz, Yen-Lu Chow, Salim E. Roucos, Michael A. Krasner, John I. Makhoul:
Improved hidden Markov modeling of phonemes for continuous speech recognition. 21-24 - Chin-Hui Lee, Kalyan Genesan:
Speech recognition under additive noise. 25-28 - Denis Jouvet, Richard M. Schwartz:
One-pass syntax-directed connected-word recognition in a time-sharing environment. 29-32 - John G. Ackenhusen:
The CDTWP: A programmable processor for connected word recognition. 33-36 - Jean-Luc Gauvain, Joseph Mariani:
Evaluation of time compression for connected word recognition. 37-40
Physiological Models and Speech Signal Representation
- Richard F. Lyon:
Computational models of neural auditory processing. 41-44 - Stephanie Seneff:
Pitch and spectral estimation of speech based on auditory synchrony model. 45-48 - Harald Höge:
A parametric representation of short-time power spectra based on the acoustic properties of the ear. 49-51 - Berj L. Bardakjian:
Modelling of the laryngeal acoustic source by labile nonlinear oscillators. 52-55 - Dale E. Veeneman, Spencer L. BeMent:
Automatic glottal inverse filtering. 56-59 - A. K. Krishnamurthy:
Two channel (speech and egg) analysis for formant and glottal inverse filtering. 60-63 - Mark A. Richards, Ronald W. Schafer:
Acoustic tube analysis of formant bandwidths and frequencies in helium speech. 64-67 - Unto K. Laine:
An all-zero model for higher pole correction. 68-71
Image Restoration
- Zhang Zhao Pu, Howard Kaufman:
Comparative evaluation of a new procedure for adaptive estimation of noisy images. 72-75 - Aggelos K. Katsaggelos, Jan Biemond, Russell M. Mersereau, Ronald W. Schafer:
An iterative method for restoring noisy blurred images. 76-79 - Philip Chan, Jae S. Lim:
One-dimensional processing for adaptive image restoration. 80-83 - Darwin T. Kuan, Alexander A. Sawchuk, Timothy C. Strand, Pierre Chavel:
Nonstationary 2-D recursive restoration of images with signal-dependent noise. 84-87 - Aharon Levi, Henry Stark:
Image restoration by the method of generalized projections with application to restoration from magnitude. 88-91 - Y. Zhou, Craig K. Rushforth, Richard L. Frost:
Singular value decomposition, singular vectors, and the discrete prolate spheroidal sequences. 92-95 - Robin N. Strickland:
Experiments on the use of local statistics for adaptive image processing. 96-99 - Tamar Peli, Thomas F. Quatieri:
Homomorphic restoration of images degraded by light cloud cover. 100-103 - Didier Saint-Félix, Ali Mohammad-Djafari, Guy Demoment:
Iterative generalized inverse image restoration. 104-107 - S. Cooke, Tariq S. Durrani:
A two-dimensional adaptive image deblurring filter. 108-111 - M. S. Ahmed, Khaldoun K. Tahboub:
Recursive Wiener filtering for image restoration. 112-115 - Kai-Bor Yu:
An improved signal restoration method using frequency domain information. 116-119
Topics in Signal Detection and Estimation
- Mysore R. Raghuveer, Chrysostomos L. Nikias:
A parametric approach to bispectrum estimation. 120-123 - T. Srinivasan, David C. Swanson, Frank W. Symons Jr.:
ARMA model order/Data length tradeoff for specified frequency resolution. 124-127 - S. Sjoberg:
Computationally efficient estimation of the mean frequency for real-valued signals. 128-131 - Steven M. Kay, Louis L. Scharf:
Invariant detection of transient ARMA signals with unknown initial conditions. 132-135 - Giuseppe Martinelli, Gianni Orlandi:
Explicit formulas for estimating the frequencies of sine waves in noise with few samples. 136-138 - Ronald E. Boucher, Joseph C. Hassab:
A quantitative comparison of generalized correlator window functions in presence of strong spectral peak for spatially separated sources. 139-142 - Mark Horwedel, Sathyanarsyan S. Rao:
The complete characterization of a multiple sinusoid signal. 143-146 - S. S. Ng:
A technique for spectral component location within a FFT resolution cell. 147-149
Artificial Intelligence and Signal Processing
- Andrew P. Witkin:
Scale-space filtering: A new approach to multi-scale description. 150-153 - Gary E. Kopec:
The signal data base system SDB. 154-157 - H. Penny Nii:
Signal-to-symbol transformation: Reasoning in the HASP/SIAP program. 158-161 - Cory Myers, Alan V. Oppenheim, Randall Davis, Webster P. Dove:
Knowledge based speech analysis and enhancement. 162-165 - Alan S. Gevins, Nelson Morgan:
"Ignorance-based" systems. 166-169
Techniques and Technologies for Automated Signal and Information Processing
- Gene Hostetter:
Spectral estimation using level-crossing data. 170-173 - Herbert E. Rauch:
Expert systems as automated decision aids. 174-177 - Joe A. Presley Jr.:
Automated detection in multiple-target environments using the censored mean-level detector. 178-181 - James M. Alsup:
High-resolution techniques for two-dimensional estimation of angle-of-arrival for planar arrays. 182-185 - Ashok Erramilli, Peter M. Schultheiss:
The effect of an auxiliary source on the performance of a randomly perturbed array. 186-189 - Norman L. Owsley:
Joint source and sensor location estimation. 190-193 - W. Brandenburg:
A point mechanical model for the dynamics of towed arrays. 194-197 - G. W. Johnson, A. O. Cohen, E. J. Modugno, C. W. Shier:
Optimal passive localization from a single sensor using multiple linear hypotheses. 198-201 - J. Bohmann, H. Meyer:
An all-digital realization of a baseband DLL implemented as dynamical state estimator. 202-205 - Richard L. Moose, Mauro J. Caputi:
A convergence analysis of an adaptive underwater passive tracking system. 206-209 - Mati Wax, Thomas Kailath:
A new approach to decentralized array processing. 210-213 - Stuart R. DeGraaf, Don H. Johnson:
Optimal linear arrays for narrow-band beamforming. 214-217 - Masato Abe, Ken'iti Kido:
Composite complex sinusoidal modeling for the estimation of directions and spectra of incident plane waves. 218-221 - Robert S. Walker, A. T. Ashley:
Modal decomposition for detection of plane waves with a line array receiver. 222-225 - Antonio Cantoni, Meng Hwa Er:
A new class of broadband time domain element space antenna array processors. 226-229
VLSI Implementations
- Gregory H. Allen, Peter B. Denyer, David S. Renshaw:
A bit serial linear array DFT. 230-233 - Wan-Chi Siu, Anthony G. Constantinides:
Hardware realization of Mersenne number transforms for fast digital convolution. 234-237 - Laurence E. Turner, Peter B. Denyer, David S. Renshaw:
A bit serial LDI recursive digital filter. 238-241 - M. J. Bell Jr., W. Kenneth Jenkins:
A residue to mixed radix converter and error checker for a five-moduli residue number system. 242-245 - I. Defee:
Parallel processing of digital convolution using finite polynomial rings. 246-249 - Ed F. Deprettere, Patrick M. Dewilde, R. Udo:
Pipelined cordic architectures for fast VLSI filtering and array processing. 250-253
Nonstationary Signal Processing
- Leon Cohen:
Distributions in signal theory. 254-257 - Augustus J. E. M. Janssen:
Gabor representation and Wigner distribution of signals. 258-261 - W. Martin:
Measuring the degree of non-stationarity by using the Wigner-Ville spectrum. 262-265 - Patrick Flandrin:
Some features of time-frequency representations of multicomponent signals. 266-269 - Yves Grenier:
Time-frequency analysis using time-dependent ARMA models. 270-273 - Bernard Escudié, Jean Grea:
Joint representations (JR) in signal theory (ST) and hilbertian analysis: A powerful tool for signal analysis. 274-277 - Theo A. C. M. Claasen, Wolfgang F. G. Mecklenbräuker:
On the time-frequency discrimination of energy distributions: Can they look sharper than Heisenberg ? 278-281 - Richard A. Altes:
Spectrograms and generalized spectrograms for classification of random processes. 282-285 - Ben R. Breed, Theodore E. Posch:
A range and azimuth estimator based on forming the spatial Wigner distribution. 286-287
Phonetic Analysis, Expert Systems, and Data Bases
- Gary E. Kopec:
Voiceless stop consonant identification using LPC spectra. 288-291 - Anshul Kumar, George A. Bekey:
Recognition of consonants using an ARMA model of the speech signal. 292-295 - Lori Faith Lamel, Victor W. Zue:
Properties of consonant sequences within words and across word boundaries. 296-299 - Maria-Gabriella Di Benedetto, Armando Lanaro:
How to avoid vowel normalization in identification of vowels in continuous speech. 300-303 - John M. Lucassen, Robert L. Mercer:
An information theoretic approach to the automatic determination of phonemic baseforms. 304-307 - Edward C. Bronson, Edward J. Coyle, Leah J. Siegel:
Modeling of english speech for the design of a distributed speech understanding system. 308-311 - Wayne A. Lea, Frantz Clermont:
Algorithms for acoustic prosodic analysis. 312-315 - Noëlle Carbonell, Dominique Fohr, Jean Paul Haton, François Lonchamp, Jean-Marie Pierrel:
An expert system for the automatic reading of French spectrograms. 316-319 - Renato De Mori, Michel Gilloux, Guy Mercier, M. Simon, C. Tarridec, Jacqueline Vaissière, D. Gillet, M. Gerard:
Integration of acoustic, phonetic, prosodic and lexical knowledge in an expert system for speech understanding. 320-323 - René Carré, Raymond Descout, Maxine Eskénazi, Joseph-Jean Mariani, M. Rossi:
The French language database: Defining, planning, and recording a large database. 324-327 - R. G. Leonard:
A database for speaker-independent digit recognition. 328-331
Algorithm Complexity
- Hari Krishna, Salvatore D. Morgera:
Linear complexity fast algorithms for a class of linear equations. 332-335 - Ahmed El Sherbini, Yo-Han Pao:
A pipeline architecture for implementing Durbin's recursive procedure. 336-338 - Koji Ogino Tanaka, Pian Totaron:
Application of fast algorithms to spectral analysis of speech signal. 339-342 - Mahmood R. Azimi-Sadjadi:
An efficient algorithm for characteristic polynomial of 2-D state-space structures. 343-346 - George Carayannis, Dimitris Manolakis, Nicholas Kalouptsidis:
Efficient algorithms and structures for lagged least squares (LS) FIR filters in the case of prewindowed signals. 347-350 - A. S. Krishnakumar, Martin Morf:
A divide-and-conquer approach to the evaluation of the characteristic polynomial of a symmetric tridiagonal matrix. 351-354 - Dimitris Manolakis, George Carayannis, Nicholas Kalouptsidis:
On the computational organization of fast sequential algorithms. 355-358 - Pierre Duhamel, Hendrik Hollmann:
A decomposition of the arithmetic for NTT's with 2 as a root of unity. 359-362
Signal Processing Hardware Implementations
- Ray Simar Jr.:
A step toward real-time interactive FIR filter design. 363-366 - John A. Eldon, J. D. Haight:
New CMOS chip facilitates multibit correlation. 367-370 - Steve Sweitzer:
A low cost FFT chip set. 371-373 - Botaro Hirosaki, Y. Tomimitsu, Shinya Ishihara, H. Nakada, Kohei Akiyama, Kazunori Nosaka:
A CMOS-VLSI rate conversion digital filter for digital audio signal processing. 374-377 - K. Okada, T. Ehara, H. Suzuki, K. Yanagida, K. Saito, N. Ichiura:
A digital signal processor module architecture and its implementation using VLSIs. 378-381 - G. A. C. Justice, Farhad Mavaddat:
An IEEE696 compatible signal processor. 382-385 - Walter Ulbrich, Tobias G. Noll, B. Zehner:
MOS-VLSI pipelined digital filters for video applications. 386-389 - W. Don Ashcraft, Hughe M. South:
Architecture and control of a distributed signal processor. 390-393 - Henri Barral, Nicolas Moreau:
Circuits for digital signal processing. 394-397 - Fred J. Taylor:
A logarithmic arithmetic unit for signal processing. 398-401 - Veljko Milutinovic:
One approach to microprocessor implementation of 4800 b/s data modem for telephone channels. 402-405 - Brian Bryden, Hisham Hassanein:
Implementation of a full duplex 2.4 kbps LPC vocoder on a single TMS-320 microprocessor chip. 406-409 - Andrew Holck, Wallace W. Anderson:
A single-processor LPC vocoder. 410-413 - R. Geppert, P. Schwartau:
A DFT-based front-end for word recognition systems. 414-417 - A. M. Chiang, Gary A. Shaw:
A CCD chip for parallel pulse-Doppler radar processing. 418-421 - J. Lim, G. N. Kalanj, Ian G. Cumming:
A programmable signal processing element designed for an efficient data-driven signal processing architecture. 422-425
System Identification and Related Topics
- H. Rauner, Werner Wolf, Ulrich Appel:
System identification techniques for noise reduction in evoked potential processing. 426-429 - Pei-Hwa Lo:
On all-zero modelling of a recursive digital filter. 430-431 - David Mansour:
Efficient nonlinear system identification. 432-435 - David C. Swanson, Frank W. Symons Jr.:
Sources of numerical errors in both the square-root normalized and unnormalized least squares lattice algorithms. 436-439 - Ken C. Sharman, Tariq S. Durrani, Mati Wax, Thomas Kailath:
Asymptotic performance of eigenstructure spectral analysis methods. 440-443 - Colin F. N. Cowan, P. F. Adams:
Non-linear system modelling: Concept and application. 444-447 - James A. Cadzow, Otis M. Soloman:
ARMA system identification: An algebraic approach. 448-451 - Daniel R. Fuhrmann, Bede Liu:
An iterative algorithm for locating the minimal eigenvector of a symmetric matrix. 452-455
Adaptive Filtering III
- Allan O. Steinhardt, William B. Jones:
A qualitative instability theory for lattice filters. 456-458 - Zhen Zhao, Zi-Qiang Hou:
The generalized phase spectrum method for time delay estimation. 459-462 - Keith A. Struckman:
Correlation of interpolated time delayed communication signals. 463-466 - Jae Chon Lee, Chong Kwan Un:
On the convergence behaviour of frequency-domain LMS adaptive digital filters. 467-470 - John A. Rajan:
On the histrograms of the correlation functions of product codes. 471-474 - João C. Serra, Nelson L. Esteves:
A blind equalization algorithm without decision. 475-478 - Michael P. Beddoes, Juhn A. Wada:
Comments on parametric and non-parametric detection of epileptiform spike activity. 479-482 - Charles M. Loeffler, R. E. Leonard Jr.:
Phase unwrapping via median filtering. 483-485 - Ping Xue, Bede Liu:
Adaptive equalizer using finite-bit power-of-two quantizer. 486-489 - Dennis R. Morgan, Athanasios Aridgides:
Adaptive array cancellation of multipath interference. 490-493
Radar and Active Sonar
- V. Nagarajan, M. R. Chidambara:
Tracking a meneuvering target in clutter-a new approach. 494-497 - V. Nagarajan, G. K. Chaturvedi, S. D. Dhage:
Modified distribution free CFAR processor for clutter edges and multi-target situations. 498-501 - T. Bucciarelli, G. Picardi, E. Prestifilippo:
Clutters cancellation using autoregressive techniques. 502-505 - Avedis S. Arslanian, Tariq S. Durrani:
Target-clutter identification by lattice processors. 506-509 - S. P. Nawathe, B. V. Rao:
Distribution free Doppler processor. 510-513 - N. Sridhar Reddy, M. N. S. Swamy:
Resolution of range and Doppler ambiguities in medium PRF radars in multiple-target environment. 514-517 - Takashi Ohmuro, Yasuo Tachibana, Michimasa Kondo:
Side-lobe canceller response for pulse interference. 518-521 - Jeffrey J. Sitterle, Ernest G. Baxa Jr.:
On applications of linear prediction filtering to small wavelength Doppler weather radar signal processing. 522-525 - Ronald A. Mucci:
An efficient procedure for broadband Doppler compensation. 526-529 - Robert H. Seegal:
Modern, active sonar AGC design considerations. 530-533 - Zhen-Biao Lin:
A method for computation of wideband ambiguity function and the numerical analysis of the bat's sonar signal. 534-537
Image Processing II
- David J. Rossi, Alan S. Willsky:
Maximum likelihood estimation of object size and orientation from projection data. 538-541 - K. S. Thyagarajan, Hüseyin Abut, H. Bheda:
A matrix quantizer for image processing. 542-544 - Phillip L. Kelly, V. Ralph Algazi:
Two step color shrinking algorithm for encoding of graphics. 545-548 - S. Thomas Alexander, Sarah A. Rajala:
Bandwidth reduction and image distortion characteristics of the LMS image compression algorithm. 549-552 - Narciso García, Carlos Muñoz, Alberto Sanz:
Universal compression lossless code statistically built. 553-556 - Emmanuel A. Arnould, Jean-Pierre Dugré:
Real time discrete cosine transform an original architecture. 557-560 - Carl Dean Bowling, Richard A. Jones:
A coefficient energy concentration model for motion compensated image coding. 561-564 - Richard A. Jones, Mark K. Cook:
A transformation for clustering noisy shape observations. 565-568 - Robert F. Kubichek, E. A. Quincy, S. B. Smithson:
Segmentation of 2-D seismic data. 569-572 - Bhaskar Ramamurthi, Allen Gersho:
Edge-oriented spatial filtering of images with application to post-processing of vector quantized images. 573-576 - Steven M. Kay, Gerald J. Lemay:
Edge detection using the linear model. 577-580 - Patrick L. Love, Marwan A. Simaan:
Seismic signal character recognition using analysis techniques. 581-584
Late Papers
- Sin-Horng Chen, John Murray, John F. Walkup:
Robust image estimation in signal-dependent noise. 585-588 - Geneviève Jourdain, J. Martin:
Optimal data estimation system over multipath channel. 589-592 - Donald W. Tufts, Ramdas Kumaresan:
Accuracy of frequency estimation and its relation to prediction filter order. 593-596 - Keikichi Hirose, Hiroya Fujisaki, Mikio Yamaguchi:
Synthesis by rule of voice fundamental frequency contours of spoken Japanese from linguistic information. 597-600 - Srdjan S. Stankovic, Marina V. Dragosevic, Miodrag Carapic:
Estimation of frequencies of sinusoids using the extended Kalman filter. 601-604 - Hiroya Fujisaki, Keikichi Hirose, Tomohiro Inoue, Yasuo Sato:
Automatic recognition of spoken words from a large vocabulary using syllable templates. 605-608
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