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ORCIDICASSP 1976: Philadelphia, Pennsylvania, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '76, Philadelphia, Pennsylvania, USA, April 12-14, 1976. IEEE 1976
General Segmentation & Phoneme Recognition
- Richard M. Schwartz:
Acoustic-phonetic experiment facility for the study of continuous speech. 1-4 - Kazuyo Tanaka:
A dynamic processing approach to extraction and categorization of phonemic information. 5-8 - N. R. Dixon, Harvey F. Silverman:
The 1976 modular acoustic processor (MAP) : Signal analysis and phonemic segmentation. 9-14 - Harvey F. Silverman, N. R. Dixon:
The 1976 modular acoustic processor (MAP) : Diadic segment classification and final phonemic string estimation. 15-20 - Richard M. Schwartz, Victor W. Zue:
Acoustic-phonetic recognition in BBN SPEECHLIS. 21-24 - Charles C. Tappert:
A Markov model acoustic phonetic component for automatic speech recognition. 25-28 - Hideki Kasuya, Hisashi Wakita:
Speech segmentation and feature normalization based on area functions. 29-32 - Paul Mermelstein:
The syntax of acoustic segments. 33-36 - Philip Christov:
An algorithm using linguistic information and its application to the analysis of speech in the spectral domain. 37 - Laurent Miclet:
A channel adapted vocoder and its applications to continuous speech recognition. 38
Pitch & Vocal Tract Modeling
- John N. Holmes:
Formant excitation before and after glottal closure. 39-42 - Yuki Kakita, Shizuo Hiki:
A study on laryngeal control for pitch change by use of anatomical structure model. 43-46 - Bernard Guérin, Mohamad Mrayati, René Carré:
A voice source taking account of coupling with the supraglottal cavities. 47-50 - M. Carcaud, J. Courbon, J. Genin, J. Lucas:
A hardware vocal source simulator. 51-54 - Dean R. Kloker:
A technique for the automatic location and description of pitch contours. 55-58 - Alistair D. C. Holden, John Y. Cheung:
Computer modeling and estimation of linguistic stress patterns. 59-62 - D. G. Childers, A. Paige, G. P. Moore, M. Nadal-Suris:
Automation of the measurement of laryngeal vibration patterns from high speed film. 63-66 - James F. McGill:
An experimental investigation of the optimal filter as an area function perdictor. 67-70 - S. Brooks, Frank Fallside:
A technique for converting the linear prediction areas model of speech to a simple articulatory model. 71-74 - Raymond Desccut, Bernard Tousignant, Michel Lecours:
Vocal tract area function measurements: Two time-domain methods. 75-78 - Ismail I. El-Mallawany:
Detection of the closed glottis interval. 79 - Frank Fallside, S. Brooks:
Analysis and areas modelling of Nasalised speech by a multivariable identification technique. 80 - Silvano Rivoira, Angelo Serra:
Research of the pitch contour rules for an Italian language speech understanding system. 81 - Marco Mezzalama, Enrico Rusconi, Pietro Torasso:
Automatic generation of pitch contour for speech synthesis. 82
Narrow Band Speech Communications
- Leland B. Jackson, John Bertrand:
An adaptive inverse digital filter for formant analysis of speech. 84-86 - John Makhoul:
Methods for nonlinear spectral distortion of speech signals. 87-90 - George S. Kang, David C. Coulter:
600 bps Voice digitizer. 91-94 - Jean-Frédéric Zurcher, Patrick Graillot, Michael Cartier, Guy David, Pierre Breant, Jean-Pierre Van Uffelen:
Speech digitalization with channel vocoders. 95-98 - Donald P. Fulghum:
Output spectrum contour scaling for an all digital channel vocoder. 99-102 - John Makhoul, R. Viswanathan, William Russell:
A framework for the objective evaluation of vocoder speech quality. 103-106 - Steven Meister, Richard Wiggins:
Quality comparison measure for linear predictive systems. 107-109
General Digital Signal Processing Concepts
- M. Shaker Sabri, Willem J. D. Steenaart:
Discrete Hilbert transform filtering. 116-119 - V. Umapathi Reddy, M. Sundaramurthy:
New results in fixed-point fast Fourier transform error analysis. 120-125 - John A. Spicer:
A new algorithm for doing the finite discrete Fourier transformation in the frequency domain imposing uniform and Gaussian boundary conditions. 126-129 - Richard O. Rowlands:
The odd discrete Fourier transform. 130-133 - Emanuel Vegh, Lawrence M. Leibowitz:
Discrete convolution of complex integer sequences. 134-135 - Kamisetty Ramamohan Rao
, Nasir Ahmed:
Orthogonal transforms for digital signal processing. 136-140 - Masuzo Yanagida, Osamu Kakusho:
Discrete Fourier transform based on a double-sampling and its applications. 141-144 - M. Ahmadi, Anthony G. Constantinides, Robert A. King:
Design technique for a class of stable two-dimensional recursive digital filters. 145-147 - Sorin Cohn-Sfetcu, J. E. Gibbs:
Harmonic differential calculus and filtering in Galois fields. 148-153
Special Phoneme Recognition
- John Burge, Frederick Hayes-Roth:
A novel pattern learning and classification procedure applied to the learning of vowels. 154-157 - Hiroya Fujisaka, Osamu Kunisaki:
Analysis, recognition and perception of voiceless fricative consonants in Japanese. 158-161 - David J. Broad:
Acoustic discrimination between [f] and [θ] in a single speaker. 162-165 - Iris Kameny:
Automatic acoustic-phonetic analysis of vowels and sonorants. 166-169 - Patrick F. Castelaz, Russell J. Niederjohn:
A comparative study of the use of zero-crossing analysis methods for vowel recognition. 170-173 - Yoshinari Kanamori, Ken'iti Kido:
Recognition of vowels in connected speech by use of the characteristics on perception of vowel. 174-177 - M. Adbul Kader Pramanik, Ken'iti Kido:
Bengali speech: Formant structures of single vowels and initial vowels of words. 178-181 - Lee M. Molho:
Automatic acoustic-phonetic analysis of fricatives and plosives. 182-185 - Hisao Kuwahara, Hisao Sakai:
Fusion and identification of synthetic vowels in dichotic listening. 186-189
Automatic Word Recognition
- M. B. Herscher, R. B. Cox:
Source data entry using voice input. 190-193 - Wen C. Lin, K. Ganesan:
Study of an on-line, adaptive speaker-independent word recognition system based on acoustic-phonetic analysis and statistical pattern recognition techniques. 194-197 - Jean A. Dreyfus-Graf:
Recognition of coded speech (phonocodes). 198-201 - Lawrence R. Rabiner, Marvin R. Sambur:
Speaker independent recognition of connected digits. 202-205 - Michael J. Coker, Steven F. Boll:
An improved isolation word recognition system based upon the linear prediction residual. 206-209 - Phillips B. Scott:
VICI - A speaker independent word recognition system. 210-213 - Ken'iti Kido, Takahide Matsuoka, Jouji Miwa, Shozo Makino, Yoshinari Kanamori:
Spoken word recognition system for unlimited adult male speakers. 214-217 - Moonis Ali:
Computers applied for the recognition of Hindi syllables. 218-221 - Joseph Kalinowski, Joseph C. Brown, Shiraz G. Bhanji, Merle G. Hooten, John W. Preusse:
Application of discrete word recognition and response to multiuser tactical communications: WRS. 222-225 - Miron Derkach, R. Gumetsky, B. Gura:
An attempt of automatic recognition of some Russian words. 226-228 - Shuzo Saito, Masaki Kohda:
Spoken word recognition using the restricted number of learning samples. 229-232
Wideband Speech Communications
- Ronald E. Crochiere, Susan A. Webber, James L. Flanagan:
Digital coding of speech in sub-bands. 233-236 - Marvin R. Sambur, Nikil S. Jayant:
LPC Synthesis starting from white noise corrupted or differentially quantized speech. 237-240 - John R. Welch, Charles F. Teacher:
A split-band predictive coding system at 16 kb/s. 241-243 - Aaron J. Goldberg, Richard L. Freudberg, Ronald S. Cheung:
High quality 16 kb/s voice transmission. 244-246 - Peter Cummiskey:
Single-integration, adaptive delta modulation. 247-250 - Ronald H. Frazier, Siamak Samsam, Louis D. Braida, Alan V. Oppenheim:
Enhancement of speech by adaptive filtering. 251-253 - Dieter Langle:
Digital encoding of variable-length vectors with application to pitch extraction and pitch-synchronous speech analysis and synthesis. 254-257 - David L. Cohn, James L. Melsa:
A pitch compensating quantizer. 258-261 - G. Pirani, Carlo Scagliola:
Performance analysis of DPCM speech-transmission systems using Kalman predictors. 262-265 - David J. Goodman, Barbara J. McDermott, Lloyd H. Nakatani:
Subjective evaluation of PCM coded speech. 266-270 - Joseph S. Golab, Robert G. Bland:
Advanced voice modems for aero communications. 271-274 - L. S. Moye:
A method for filling gaps in a speech signal left by the excision of impulsive noise. 275
Digital Filter Theory
- Yrjö Neuvo, Olli Simula:
Digital lattice filters with reduced number of multipliers. 276-279 - Michael J. Carey, G. D. Tattersall, D. Goodman, A. R. Potter:
Filtering for code conversion in digital telephone exchanges. 280-283 - Benjamin J. Leon, Michael T. McCallig:
Design of nonrecursive digital filters to meet maximum and minimum frequency response constraints. 284-287 - Bhagwati Prasad Agrawal:
An algorithm for designing constrained least-squares filters. 288-291 - Ronald E. Crochiere, Lawrence R. Rabiner:
Recent developments in the design and implementation of digital decimators, interpolators, and narrow band filters. 292-295 - P. F. Scott:
The autocorrelation function and spectra of a signal that has been randomly sampled. 296-299 - Louis L. Scharf, Joseph Perl:
Covariance-invariant signal processing. 300-304 - M. Shaker Sabri, A. E. Mostafa:
Graphical determination of the group delay characteristics of digital filters. 305
Speech Analysis
- Norman Green:
Analysis-synthesis using pole-zero approximations to speech spectra. 306-309 - Shuichi Itahashi, Shoichi Yokoyama:
Automatic formant extraction utilizing mel scale and equal loudness contour. 310-313 - Donald W. Tufts, Stephen E. Levinson, R. Rao:
Measuring pitch and formant frequencies for a speech understanding system. 314-317 - Kazuhiro Fuchi, Shuichi Itahashi:
Direct linear prediction for fundamental frequency analysis. 318-321 - Wolfgang J. Hess:
An algorithm for digital time-domain pitch period determination of speech signals and its application to detect F0dynamics in VCV utterances. 322-325 - Leah J. Siegel, Kenneth Steiglitz:
A pattern classification algorithm for the voiced/Unvoiced decision. 326-329 - Gilbert M. Kaufman:
Dynamic energy tracking for responsive voicing. 330-331 - Michael J. Cheng, Lawrence R. Rabiner, Aaron E. Rosenberg, Carol A. McGonegal:
Some comparisons among several pitch detection algorithms. 332-335 - B. Yegnanarayana:
Effect of noise and distortion in speech on parametric extraction. 336-339 - David J. Anderson, John R. Deller Jr., Robert E. Stone Jr.:
Computer analysis of time jitter in vowel sounds. 340-342 - Günther Ruske:
Real-time information reduction in digital sound spectograms of speech. 343-346 - Hirokazu Sato:
Perceptual and acoustic cues of female voice. 347 - Hiroyoshi Morikawa, Hiroya Fujisaki:
An adaptive speech analysis system. 348 - Wolfgang J. Hess:
Non-prosodic pitch variations in continouos speech. 349
Spectral Analysis & Deconvolution
- L. T. Quick, L. P. Bolgiano:
Deconvolution by poisson transformation. 350-353 - Bobby R. Hunt, H. Joel Trussell:
Sectioned digital filtering for nonlinear Bayesian signal deconvolution. 354-356 - G. Clifford Carter, Charles H. Knapp:
Time delay estimation. 357-360 - Eugene I. Plotkin:
Signal processing by function elimination filters. 361-364 - Thomas E. Eger:
Sectioned spectrum processing for wideband signals. 365-368 - Bruce A. Eisenstein, Louis R. Cerrato:
Crossfrequency estimation for deconvolution. 369-372 - Michael P. Ekstrom:
A Monte Carlo approach to numerical deconvolution. 373-375 - Madihally J. Narasimha, Kishan Shenoi, Allen M. Peterson:
Quadratic residues: Application to chirp filters and discrete Fourier transforms. 376-378
Underwater Acoustics-Algorithms and Hardware
- Thomas J. Curry, Donald W. Tufts:
Comparison of transit detectors in ocean-like environments. 379-382 - William Barry:
Localized variation in the ocean's transmission properties: Its drastic effect on a sonar display. 383-385 - Lyles C. Adair:
A novel method for measuring phase and sensitivity of long focused acoustic array. 386-388 - Robert C. Trider:
A fast Fourier transform (FFT) based sonar signal processor. 389-393 - D. V. Gupta, J. F. Vetelino, Thomas J. Curry, J. T. Francis:
An adaptive thresholding system for functioning in nonstationary noise backgrounds. 394-397 - Dwight O. Monteith Jr.:
Modeling hydrophone array directivity effects for sonar system performance prediction. 398-401
Loudspeaker Systems
- A. Neville Thiele:
Applications of network synthesis to loudspeaker system theory. 402-405 - Richard H. Small:
Synthesis of loudspeaker driver parameters. 406-408 - J. Robert Ashley:
Non-linear effects in direct radiator loudspeaker systems. 409 - D. B. Keele Jr., Raymond Newman:
Application of recent Australian loudspeaker research to producible loudspeaker systems. 410-412 - J. E. Benson:
Theory and applications of electrically tapered electro-acoustic arrays. 413-415 - Frederick Hayes-Roth, Victor R. Lesser:
Focus of attention in a distributed-logic speech understanding system. 416-420 - Frederick Hayes-Roth, David J. Mostow:
Syntax and semantics in a distributed speech understanding system. 421-424 - Lalit R. Bahl, James K. Baker, Paul S. Cohen, N. R. Dixon, Frederick Jelinek, Robert L. Mercer, Harvey F. Silverman:
Preliminary results on the performance of a system for the automatic recognition of continuous speech. 425-429 - Jean Paul Haton, Jean-Marie Pierrel:
Organization and operation of a connected speech understanding system at lexical, syntactic and semantic levels. 430-433 - Toby E. Skinner, Dean R. Kloker, Mark F. Medress:
A speech recognition system for connected word sequences. 434-437 - William A. Woods, Madeleine Bates, Geoffrey Brown, Bertram C. Bruce, John W. Klovstad, Bonnie Nash-Webber:
Uses of higher level knowledge in a speech understanding system: A progress report. 438-441 - Masaki Kohda, Ryohei Nakatsu, Kiyohiro Shikano:
Speech recognition in the question-answering system operated by conversational speech. 442-445 - Taras K. Vintsiuk:
Generative grammars and dynamic programming in speech recognition with learning. 446-449 - Taras K. Vintsiuk, O. N. Gavrilyuk, A. G. Shinkazh:
Phoneme-by-phoneme recognition of speech composed of the words of given vocabulary. 450-452 - Patrice Quinton:
A syntactic analyzer adapted to speech recognition. 453-456 - L. Buisson, G. Mercier, Patrice Quinton, R. Vives:
A multi-purpose speech recognition system. 457
Application of Linear Predictive Coding to Speech Communications
- Alistair D. C. Holden, Y. K. Gulut:
A new method for accurate analysis of voiced speech. 458-461 - John Makhoul:
New lattice methods for linear prediction. 462-465 - John Makhoul, Lynn Cosell:
LPCW: An LPC vocoder with linear predictive spectral warping. 466-469 - John E. Roberts, Richard Wiggins:
Piecewise linear predictive coding (PLPC). 470-473 - Thomas E. Tremain:
Linear predictive coding systems. 474-478 - Charles Schmid:
A direct method for sequentially updating linear predictor coefficients for the covariance method. 479-480 - John D. Markel, Augustine H. Gray Jr.:
A comparison of three methods for coefficient quantization and bit allocation. 481-484 - R. Viswanathan, John Makhoul, William Russell:
Towards perceptually consistent measures of spectral distance. 485-488 - George Carayannis, Claude Guéguen:
The factorial linear modelling: A Karhumen-Loeve approach to speech analysis. 489-492 - D. L. Helsey, Lloyd J. Griffiths:
Linear estimation filters in spectral analysis. 493-496 - Caldwell P. Smith:
Evaluation of the performance of piecewise-linear-prediction coding (PLPC) of speech signals. 497
Structure and Quantization
- S. Y. Hwang:
Roundoff noise minimization in state-space digital filtering. 498-500 - Andrew C. Callahan:
Random rounding: Some principles and applications. 501-504 - Clifford T. Mullis, Richard A. Roberts:
Filter structures which minimize roundoff noise in fixed point digital filters. 505-508 - Dean J. Schmidlin:
An equivalent network theory for a class of discrete-time networks. 509-512 - Kotaro Hirano, Hisashi Sakaguchi, Bede Liu:
Optimization of recursive cascade filters. 513-516 - Theodore R. Lapp, Robert A. Gabel:
An algorithm for optimally ordering the sections of a cascade digital filter. 517-520 - C. L. Chao, B. C. Chi:
A general realization method for wave digital filters. 521-524 - A. L. Moyer:
An efficient parallel algorithm for digital IIR filters. 525-528 - L. P. Mulcahy:
Digital fixed-point multiplication error structure and some consequences. 529-532 - Arild Lacroix:
Error estimation of digital filters with arbitrary structure and arithmetic by simulation. 533-536
Distortion in Music Entertainment Systems
- Svetislav V. Djuric:
Distortion in microphones. 537-539 - Marvin Camras:
Distortion in tape recording systems. 540 - James E. White:
Distortion in disc recording systems. 541 - Matti Otala:
Distortion in audio amplifiers. 542 - Paul W. Klipsch:
Loudspeaker distortion. 543-546 - J. Robert Ashley:
On the audibility of distortion. 547
Automatic Word Spotting
- William E. Cooper:
Speech timing of coreference. 548 - A. Richard Smith:
Word hypothesization in the hearsay II speech system. 549-552 - Craig Cook:
Word verification in a speech understanding system. 553-556 - Richard W. Christiansen, Craig K. Rushforth:
Word spotting in continuous speech using linear predictive coding. 557-560 - William A. Woods, Victor W. Zue:
Dictionary expansion via phonological rules for a speech understanding system. 561-564 - Renato De Mori, Pietro Torasso:
Lexical classification in a speech understanding system using fuzzy relations. 565-568 - Nina H. MacDonald:
Duration as a syntactic boundary cue in ambiguous sentences. 569-572 - Dennis H. Klatt:
A digital filter bank for spectral matching. 573-576 - Beatrice T. Oshika:
Phonological rule testing of conversational speech. 577 - John W. Klovstad:
Probabilistic lexical retrieval component with embedded phonological word boundary rules. 578
Aids for Handicapped
- Howard C. Schweitzer, G. Donald Causey:
Intermodulation distortion in hearing aids: The need for measurement standards and inherent complications. 579-582 - Harry Levitt, R. E. C. White, S. B. Resnick:
Protocol for prescriptive fitting of a wearable master hearing aid. 583-585 - D. A. MacKinnon, H. C. Lee:
Realtime recognition of unvoiced fricatives in continuous speech to aid the deaf. 586-589 - L. C. Stewart, Wilbur D. Larkin, Robert A. Houde:
A real time spectrograph with implications for speech training for the deaf. 590-593 - Frank A. Saunders, William A. Hill, Carol A. Simpson:
Speech perception via the tactile mode: Progress report. 594-597 - Moise H. Goldstein Jr., Rachel E. Stark, Grace H. Yeni-Komshian, David G. Grant:
Tactile stimulation as an aid for the deaf in production and reception of speech: Preliminary studies. 598-601 - Barbara Franklin:
Analysis of consonant recognition scores of congenital sensorineural hearing impaired. 602-605 - William De l'Aune, Chester Lewis, Mary Dolan, Thomas Grimmelsman, Walter Needham:
Two sensory aids having profound effects on the blind. 606-610 - Joseph G. Agnello, Ernest M. Weiler, Max F. Farley:
An instrumentation for facilitating easy onset patterns for stutterers. 611-612 - Shizuo Hiki, Satoshi Imaizumi, Minoru Hirano, Hideaki Matsushita, Yuki Kakita:
Acoustical analysis for voice disorders. 613-616 - Shizuo Hiki, Ryuzaemon Kagami:
Some properties of formant frequencies of vowels uttered by deaf and hard of hearing children. 617
Hardware and Architecture for Signal Processors
- J. P. Agrawal, Jacob Ninan:
Hardware considerations in FFT processors. 618-621 - L. Robert Morris:
Automatic generation of time efficient digital signal processing software. 622-625 - Charlton M. Walter:
Programmable communications terminal optimization using systematic digital signal compression procedures. 626-629 - S. T. Scott, Howard A. Stromberg:
Simple hybrid systems for accurate synthesis and analysis of harmonic spectra. 630-632 - F. C. Pirz:
A digital loop communication system applied to signal processing and speech research. 633-635 - Abraham Peled:
A digital signal processing system. 636-639 - J. Robinson, John R. Welch, Charles F. Teacher:
The programmable array processor. 640-643 - Eric Hanson:
A new portable stand alone digital processor. 644-646 - Edmund K. Cheng, Carver A. Mead:
A two's complement pipeline multiplier. 647-650 - John A. V. Rogers:
GASP: A programmable signal processor. 651
Underwater Acoustics-Performance Measures and Prediction
- Robert J. Urick:
Sonar design in the real ocean: Multipath limitations on sonar performance. 652-655 - Stanley L. Adams, William John Jobst:
Statistical properties of underwater acoustic ambient noise fields. 656-659 - Albert A. Gerlach:
Motion induced coherence degradation in passive systems. 660-663 - Stanley L. Adams, J. Doubek:
Dispersive properties of the underwater acoustic channel. 664-667 - R. D. Graves, Anton Nagl, Herbert Überall, G. L. Zarur:
Normal mode theory of underwater sound propagation in a range dependent environment. 668-670 - T. C. Slotwinski:
Sonar design in the real ocean: Target, background, and own ship limitations on sonar performance. 671-674 - R. F. Tiel:
The passive sonar equation - effects of additive interference. 675-678 - M. A. Cosgrove:
Wideband target strength measurements. 679-681 - D. E. Nelson:
An automated solution for the wide band sonar equation. 682-685
Speech Synthesis
- Rolf Carlson, Björn Granström:
A text-to-speech system based entirely on rules. 686-688 - Cecil H. Coker, S. A. Webber:
Speech synthesis from english text: A progress report. 689 - Peter M. Seeviour, John N. Holmes, Michael W. Judd:
Automatic generation of control signals for a parallel formant speech synthesizer. 690-693 - Danielle Larreur, Françoise Emerard:
Speech synthesis by dyads and automatic intonation processing. 694-697 - Ching V. Suen:
Computer synthesis of Mandarin. 698-700 - Jonathan Allen, Douglas D. O'Shaughnessy:
A comprehensive model for fundamental frequency generation. 701-704 - Steven F. Boll, Ercolino Ferretti, Tracy Petersen:
Improving synthetic speech quality using binaural reverberation. 705-708 - Uwe Dibbern:
Multiplex-vocoder for voice response. 709-712 - Jouji Suzuki:
Speech processing by splicing of autocorrelation function. 713-716 - James F. McGill:
A multi-channel digital audio laboratory facility. 717-720 - Lei F. Willems:
Speech resynthesis with a hardware synthesizer. 721 - Gian Antonio Mian, F. Morgantini, Carlo Offelli:
An application of the linear prediction technique to efficient coding of speech segments. 722
Speaker Recognition
- V. V. S. Sarma, B. Yegnanarayana:
Cascade realization of digital inverse filter for extracting speaker dependent features. 723-726 - Marvin R. Sambur:
Text independent speaker recognition using orthogonal linear prediction. 727-729 - G. S. Ramishvili, M. A. Tushishvili:
On the connection of some time characteristics of speech signal with the individuality of voice. 730-733 - Wen C. Lin, Sasi K. Pillay:
Feature evaluation and selection for an on-line, adaptive speaker verification system. 734-737 - Ernst Bunge:
Automatic speaker recognition by computers. 738 - Frank Fallside:
Speaker identification by multivariable linear prediction analysis. 739 - H. S. Hayre:
Speech - a possible indicator of physical stress. 740
Applications of Signal Processing
- Thomas C. Cantwell, Richard D. Wilmot:
Analysis of techniques for processing parallel signal outputs. 741-744 - Ronald C. Houts, Donald W. Burlage:
Design procedure for improving the usable bandwidth of an MTI radar signal processor. 745-748 - Bruce A. Eisenstein, John Fehlauer:
Signal processing for feature extraction and pattern recognition. 749-752 - William T. Rhodes, James M. Florence:
Frequency-variant optical systems for signal analysis and processing. 753-755 - James K. Hsiao:
Performance of a range-ambiguous MTI and doppler filter system. 756-759 - Allen C. Eberhardt, W. F. Reiter:
Digital signal processing techniques in truck tire vibration and sound analysis. 760-763 - Chiou-Shiun Chen, Louis E. Roemer:
The application of cepstrum technique in power cable fault detection. 764-767 - Takeru Irabu, Yuichi Tomita, Toshihiko Hagisawa, Eiichi Kiuchi:
Plane averaging signal processing on radar using DFT technology. 768-771
Electroacoustics Potpourri
- J. Robert Ashley:
Is four channel a quadrifizzle? 772-776 - C. E. Wilson:
Noise measurement to ensure compatible land use. 777-778 - Herbert H. Ernyei:
Theoretical aspect of electromechanical transducers. 779-785 - Michael D. Watson, Brian A. McIntosh, Douglas O. Revelle:
A meteor infrasound recording system. 786-789 - Lester L. Boyer:
The visual component of speech reception in conference centers. 790-793 - Hiroshi Ono, Shigeji Saito, Shinsaku Mori:
Vibration pick-up type ear microphone. 794
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