Table of Contents

• Acoustic-phonetic experiment facility for the study of continuous speechRichard M. Schwartz. 1-4 [doi]
• A dynamic processing approach to extraction and categorization of phonemic informationKazuyo Tanaka. 5-8 [doi]
• The 1976 modular acoustic processor (MAP) : Signal analysis and phonemic segmentationN. R. Dixon, Harvey F. Silverman. 9-14 [doi]
• The 1976 modular acoustic processor (MAP) : Diadic segment classification and final phonemic string estimationHarvey F. Silverman, N. R. Dixon. 15-20 [doi]
• Acoustic-phonetic recognition in BBN SPEECHLISRichard M. Schwartz, Victor W. Zue. 21-24 [doi]
• A Markov model acoustic phonetic component for automatic speech recognitionCharles C. Tappert. 25-28 [doi]
• Speech segmentation and feature normalization based on area functionsHideki Kasuya, Hisashi Wakita. 29-32 [doi]
• The syntax of acoustic segmentsPaul Mermelstein. 33-36 [doi]
• An algorithm using linguistic information and its application to the analysis of speech in the spectral domainPhilip Christov. 37 [doi]
• A channel adapted vocoder and its applications to continuous speech recognitionLaurent Miclet. 38 [doi]
• Formant excitation before and after glottal closureJohn N. Holmes. 39-42 [doi]
• A study on laryngeal control for pitch change by use of anatomical structure modelYuki Kakita, Shizuo Hiki. 43-46 [doi]
• A voice source taking account of coupling with the supraglottal cavitiesBernard Guérin, Mohamad Mrayati, René Carré. 47-50 [doi]
• A hardware vocal source simulatorM. Carcaud, J. Courbon, J. Genin, J. Lucas. 51-54 [doi]
• A technique for the automatic location and description of pitch contoursDean R. Kloker. 55-58 [doi]
• Computer modeling and estimation of linguistic stress patternsAlistair D. C. Holden, John Y. Cheung. 59-62 [doi]
• Automation of the measurement of laryngeal vibration patterns from high speed filmD. G. Childers, A. Paige, G. P. Moore, M. Nadal-Suris. 63-66 [doi]
• An experimental investigation of the optimal filter as an area function perdictorJames F. McGill. 67-70 [doi]
• A technique for converting the linear prediction areas model of speech to a simple articulatory modelS. Brooks, Frank Fallside. 71-74 [doi]
• Vocal tract area function measurements: Two time-domain methodsRaymond Desccut, Bernard Tousignant, Michel Lecours. 75-78 [doi]
• Detection of the closed glottis intervalIsmail I. El-Mallawany. 79 [doi]
• Analysis and areas modelling of Nasalised speech by a multivariable identification techniqueFrank Fallside, S. Brooks. 80 [doi]
• Research of the pitch contour rules for an Italian language speech understanding systemSilvano Rivoira, Angelo Serra. 81 [doi]
• Automatic generation of pitch contour for speech synthesisMarco Mezzalama, Enrico Rusconi, Pietro Torasso. 82 [doi]
• An adaptive inverse digital filter for formant analysis of speechLeland B. Jackson, John Bertrand. 84-86 [doi]
• Methods for nonlinear spectral distortion of speech signalsJohn Makhoul. 87-90 [doi]
• 600 bps Voice digitizerGeorge S. Kang, David C. Coulter. 91-94 [doi]
• Speech digitalization with channel vocodersJean-Frederic Zurcher, Patrick Graillot, Michael Cartier, Guy David, Pierre Breant, Jean-Pierre Van Uffelen. 95-98 [doi]
• Output spectrum contour scaling for an all digital channel vocoderDonald P. Fulghum. 99-102 [doi]
• A framework for the objective evaluation of vocoder speech qualityJohn Makhoul, R. Viswanathan, William Russell. 103-106 [doi]
• Quality comparison measure for linear predictive systemsSteven Meister, Richard Wiggins. 107-109 [doi]
• Discrete Hilbert transform filteringM. Shaker Sabri, Willem J. D. Steenaart. 116-119 [doi]
• New results in fixed-point fast Fourier transform error analysisV. Umapathi Reddy, M. Sundaramurthy. 120-125 [doi]
• A new algorithm for doing the finite discrete Fourier transformation in the frequency domain imposing uniform and Gaussian boundary conditionsJohn A. Spicer. 126-129 [doi]
• The odd discrete Fourier transformRichard O. Rowlands. 130-133 [doi]
• Discrete convolution of complex integer sequencesEmanuel Vegh, Lawrence M. Leibowitz. 134-135 [doi]
• Orthogonal transforms for digital signal processingKamisetty Ramamohan Rao, Nasir Ahmed. 136-140 [doi]
• Discrete Fourier transform based on a double-sampling and its applicationsMasuzo Yanagida, Osamu Kakusho. 141-144 [doi]
• Design technique for a class of stable two-dimensional recursive digital filtersM. Ahmadi, Anthony G. Constantinides, R. A. King. 145-147 [doi]
• Harmonic differential calculus and filtering in Galois fieldsSorin Cohn-Sfetcu, J. E. Gibbs. 148-153 [doi]
• A novel pattern learning and classification procedure applied to the learning of vowelsJohn Burge, Frederick Hayes-Roth. 154-157 [doi]
• Analysis, recognition and perception of voiceless fricative consonants in JapaneseHiroya Fujisaka, Osamu Kunisaki. 158-161 [doi]
• Acoustic discrimination between [f] and [θ] in a single speakerDavid J. Broad. 162-165 [doi]
• Automatic acoustic-phonetic analysis of vowels and sonorantsIris Kameny. 166-169 [doi]
• A comparative study of the use of zero-crossing analysis methods for vowel recognitionPatrick F. Castelaz, Russel J. Niederjohn. 170-173 [doi]
• Recognition of vowels in connected speech by use of the characteristics on perception of vowelYoshinari Kanamori, Ken'iti Kido. 174-177 [doi]
• Bengali speech: Formant structures of single vowels and initial vowels of wordsM. Adbul Kader Pramanik, Ken'iti Kido. 178-181 [doi]
• Automatic acoustic-phonetic analysis of fricatives and plosivesLee Molho. 182-185 [doi]
• Fusion and identification of synthetic vowels in dichotic listeningHisao Kuwahara, Hisao Sakai. 186-189 [doi]
• Source data entry using voice inputM. B. Herscher, R. B. Cox. 190-193 [doi]
• Study of an on-line, adaptive speaker-independent word recognition system based on acoustic-phonetic analysis and statistical pattern recognition techniquesWen C. Lin, K. Ganesan. 194-197 [doi]
• Recognition of coded speech (phonocodes)Jean A. Dreyfus-Graf. 198-201 [doi]
• Speaker independent recognition of connected digitsLawrence R. Rabiner, Marvin R. Sambur. 202-205 [doi]
• An improved isolation word recognition system based upon the linear prediction residualMichael J. Coker, Steven F. Boll. 206-209 [doi]
• VICI - A speaker independent word recognition systemPhillips B. Scott. 210-213 [doi]
• Spoken word recognition system for unlimited adult male speakersKen'iti Kido, Takahide Matsuoka, Joji Miwa, Shozo Makino, Yoshinari Kanamori. 214-217 [doi]
• Computers applied for the recognition of Hindi syllablesMoonis Ali. 218-221 [doi]
• Application of discrete word recognition and response to multiuser tactical communications: WRSJoseph Kalinowski, Joseph C. Brown, Shiraz G. Bhanji, Merle G. Hooten, John W. Preusse. 222-225 [doi]
• An attempt of automatic recognition of some Russian wordsM. Derkach, R. Gumetsky, B. Gura. 226-228 [doi]
• Spoken word recognition using the restricted number of learning samplesShuzo Saito, Masaki Kohda. 229-232 [doi]
• Digital coding of speech in sub-bandsRonald E. Crochiere, Susan A. Webber, James L. Flanagan. 233-236 [doi]
• LPC Synthesis starting from white noise corrupted or differentially quantized speechM. R. Sambur, Nikil S. Jayant. 237-240 [doi]
• A split-band predictive coding system at 16 kb/sJohn R. Welch, Charles F. Teacher. 241-243 [doi]
• High quality 16 kb/s voice transmissionAaron J. Goldberg, Richard L. Freudberg, Ronald S. Cheung. 244-246 [doi]
• Single-integration, adaptive delta modulationP. Cummiskey. 247-250 [doi]
• Enhancement of speech by adaptive filteringRonald H. Frazier, Siamak Samsam, Louis D. Braida, Alan V. Oppenheim. 251-253 [doi]
• Digital encoding of variable-length vectors with application to pitch extraction and pitch-synchronous speech analysis and synthesisDieter Langle. 254-257 [doi]
• A pitch compensating quantizerDavid L. Cohn, James L. Melsa. 258-261 [doi]
• Performance analysis of DPCM speech-transmission systems using Kalman predictorsG. Pirani, C. Scagliola. 262-265 [doi]
• Subjective evaluation of PCM coded speechD. J. Goodman, B. J. McDermott, Lloyd H. Nakatani. 266-270 [doi]
• Advanced voice modems for aero communicationsJoseph S. Golab, Robert G. Bland. 271-274 [doi]
• A method for filling gaps in a speech signal left by the excision of impulsive noiseL. S. Moye. 275 [doi]
• Digital lattice filters with reduced number of multipliersYrjö Neuvo, Olli Simula. 276-279 [doi]
• Filtering for code conversion in digital telephone exchangesMichael J. Carey 0002, G. D. Tattersall, D. Goodman, A. R. Potter. 280-283 [doi]
• Design of nonrecursive digital filters to meet maximum and minimum frequency response constraintsBenjamin J. Leon, Michael T. McCallig. 284-287 [doi]
• An algorithm for designing constrained least-squares filtersB. P. Agrawal. 288-291 [doi]
• Recent developments in the design and implementation of digital decimators, interpolators, and narrow band filtersRonald E. Crochiere, Lawrence R. Rabiner. 292-295 [doi]
• The autocorrelation function and spectra of a signal that has been randomly sampledP. F. Scott. 296-299 [doi]
• Covariance-invariant signal processingLouis L. Scharf, Joseph Perl. 300-304 [doi]
• Graphical determination of the group delay characteristics of digital filtersM. S. Sabri, A. E. Mostafa. 305 [doi]
• Analysis-synthesis using pole-zero approximations to speech spectraNorman Green. 306-309 [doi]
• Automatic formant extraction utilizing mel scale and equal loudness contourShuichi Itahashi, Shoichi Yokoyama. 310-313 [doi]
• Measuring pitch and formant frequencies for a speech understanding systemDonald W. Tufts, Stephen E. Levinson, R. Rao. 314-317 [doi]
• Direct linear prediction for fundamental frequency analysisKazuhiro Fuchi, Shuichi Itahashi. 318-321 [doi]
• An algorithm for digital time-domain pitch period determination of speech signals and its application to detect F0dynamics in VCV utterancesWolfgang J. Hess. 322-325 [doi]
• A pattern classification algorithm for the voiced/Unvoiced decisionLeah J. Siegel, Kenneth Steiglitz. 326-329 [doi]
• Dynamic energy tracking for responsive voicingGilbert M. Kaufman. 330-331 [doi]
• Some comparisons among several pitch detection algorithmsMichael J. Cheng, Lawrence R. Rabiner, Aaron E. Rosenberg, Carol A. McGonegal. 332-335 [doi]
• Effect of noise and distortion in speech on parametric extractionB. Yegnanarayana. 336-339 [doi]
• Computer analysis of time jitter in vowel soundsDavid J. Anderson, John R. Deller Jr., Robert E. Stone Jr.. 340-342 [doi]
• Real-time information reduction in digital sound spectograms of speechGünther Ruske. 343-346 [doi]
• Perceptual and acoustic cues of female voiceHirokazu Sato. 347 [doi]
• An adaptive speech analysis systemHiroyoshi Morikawa, Hiroya Fujisaki. 348 [doi]
• Non-prosodic pitch variations in continouos speechWolfgang J. Hess. 349 [doi]
• Deconvolution by poisson transformationL. T. Quick, L. P. Bolgiano. 350-353 [doi]
• Sectioned digital filtering for nonlinear Bayesian signal deconvolutionBobby R. Hunt, H. Joel Trussell. 354-356 [doi]
• Time delay estimationG. Clifford Carter, Charles H. Knapp. 357-360 [doi]
• Signal processing by function elimination filtersEugene I. Plotkin. 361-364 [doi]
• Sectioned spectrum processing for wideband signalsThomas E. Eger. 365-368 [doi]
• Crossfrequency estimation for deconvolutionBruce A. Eisenstein, Louis R. Cerrato. 369-372 [doi]
• A Monte Carlo approach to numerical deconvolutionM. P. Ekstrom. 373-375 [doi]
• Quadratic residues: Application to chirp filters and discrete Fourier transformsM. J. Narasimha, K. Shenoi, A. M. Peterson. 376-378 [doi]
• Comparison of transit detectors in ocean-like environmentsThomas J. Curry, Donald W. Tufts. 379-382 [doi]
• Localized variation in the ocean's transmission properties: Its drastic effect on a sonar displayWilliam Barry. 383-385 [doi]
• A novel method for measuring phase and sensitivity of long focused acoustic arrayLyles C. Adair. 386-388 [doi]
• A fast Fourier transform (FFT) based sonar signal processorRobert C. Trider. 389-393 [doi]
• An adaptive thresholding system for functioning in nonstationary noise backgroundsD. V. Gupta, J. F. Vetelino, Thomas J. Curry, J. T. Francis. 394-397 [doi]
• Modeling hydrophone array directivity effects for sonar system performance predictionDwight O. Monteith Jr.. 398-401 [doi]
• Applications of network synthesis to loudspeaker system theoryA. N. Thiele. 402-405 [doi]
• Synthesis of loudspeaker driver parametersR. H. Small. 406-408 [doi]
• Non-linear effects in direct radiator loudspeaker systemsJ. Robert Ashley. 409 [doi]
• Application of recent Australian loudspeaker research to producible loudspeaker systemsD. B. Keele Jr., Raymond Newman. 410-412 [doi]
• Theory and applications of electrically tapered electro-acoustic arraysJ. E. Benson. 413-415 [doi]
• Focus of attention in a distributed-logic speech understanding systemFrederick Hayes-Roth, Victor R. Lesser. 416-420 [doi]
• Syntax and semantics in a distributed speech understanding systemFrederick Hayes-Roth, David J. Mostow. 421-424 [doi]
• Preliminary results on the performance of a system for the automatic recognition of continuous speechLalit R. Bahl, James K. Baker, Paul S. Cohen, N. R. Dixon, Frederick Jelinek, Robert L. Mercer, Harvey F. Silverman. 425-429 [doi]
• Organization and operation of a connected speech understanding system at lexical, syntactic and semantic levelsJean-Paul Haton, Jean-Marie Pierrel. 430-433 [doi]
• A speech recognition system for connected word sequencesToby E. Skinner, Dean R. Kloker, Mark F. Medress. 434-437 [doi]
• Uses of higher level knowledge in a speech understanding system: A progress reportW. A. Woods, M. Bates, G. Brown, B. Bruce, J. W. Klovstad, B. Nash-Webber. 438-441 [doi]
• Speech recognition in the question-answering system operated by conversational speechMasaki Kohda, Ryohei Nakatsu, Kiyohiro Shikano. 442-445 [doi]
• Generative grammars and dynamic programming in speech recognition with learningTaras K. Vintsiuk. 446-449 [doi]
• Phoneme-by-phoneme recognition of speech composed of the words of given vocabularyTaras K. Vintsiuk, O. N. Gavrilyuk, A. G. Shinkazh. 450-452 [doi]
• A syntactic analyzer adapted to speech recognitionP. Quinton. 453-456 [doi]
• A multi-purpose speech recognition systemL. Buisson, G. Mercier, P. Quinton, R. Vives. 457 [doi]
• A new method for accurate analysis of voiced speechAlistair D. C. Holden, Y. K. Gulut. 458-461 [doi]
• New lattice methods for linear predictionJohn Makhoul. 462-465 [doi]
• LPCW: An LPC vocoder with linear predictive spectral warpingJohn Makhoul, Lynn Cosell. 466-469 [doi]
• Piecewise linear predictive coding (PLPC)John E. Roberts, Richard Wiggins. 470-473 [doi]
• Linear predictive coding systemsThomas Tremain. 474-478 [doi]
• A direct method for sequentially updating linear predictor coefficients for the covariance methodCharles Schmid. 479-480 [doi]
• A comparison of three methods for coefficient quantization and bit allocationJohn D. Markel, Augustine H. Gray Jr.. 481-484 [doi]
• Towards perceptually consistent measures of spectral distanceR. Viswanathan, John Makhoul, William Russell. 485-488 [doi]
• The factorial linear modelling: A Karhumen-Loeve approach to speech analysisGeorge Carayannis, C. Gueguen. 489-492 [doi]
• Linear estimation filters in spectral analysisD. L. Helsey, L. J. Griffiths. 493-496 [doi]
• Evaluation of the performance of piecewise-linear-prediction coding (PLPC) of speech signalsCaldwell P. Smith. 497 [doi]
• Roundoff noise minimization in state-space digital filteringS. Y. Hwang. 498-500 [doi]
• Random rounding: Some principles and applicationsA. C. Callahan. 501-504 [doi]
• Filter structures which minimize roundoff noise in fixed point digital filtersClifford T. Mullis, Richard A. Roberts. 505-508 [doi]
• An equivalent network theory for a class of discrete-time networksDean J. Schmidlin. 509-512 [doi]
• Optimization of recursive cascade filtersKotaro Hirano, Hisashi Sakaguchi, Bede Liu. 513-516 [doi]
• An algorithm for optimally ordering the sections of a cascade digital filterT. R. Lapp, R. A. Gabel. 517-520 [doi]
• A general realization method for wave digital filtersC. L. Chao, B. C. Chi. 521-524 [doi]
• An efficient parallel algorithm for digital IIR filtersA. L. Moyer. 525-528 [doi]
• Digital fixed-point multiplication error structure and some consequencesL. P. Mulcahy. 529-532 [doi]
• Error estimation of digital filters with arbitrary structure and arithmetic by simulationArild Lacroix. 533-536 [doi]
• Distortion in microphonesSvetislav V. Djuric. 537-539 [doi]
• Distortion in tape recording systemsMarvin Camras. 540 [doi]
• Distortion in disc recording systemsJames White. 541 [doi]
• Distortion in audio amplifiersMatti Otala. 542 [doi]
• Loudspeaker distortionPaul W. Klipsch. 543-546 [doi]
• On the audibility of distortionJ. Robert Ashley. 547 [doi]
• Speech timing of coreferenceWilliam E. Cooper. 548 [doi]
• Word hypothesization in the hearsay II speech systemA. Richard Smith. 549-552 [doi]
• Word verification in a speech understanding systemCraig Cook. 553-556 [doi]
• Word spotting in continuous speech using linear predictive codingRichard W. Christiansen, Craig K. Rushforth. 557-560 [doi]
• Dictionary expansion via phonological rules for a speech understanding systemWilliam A. Woods, Victor W. Zue. 561-564 [doi]
• Lexical classification in a speech understanding system using fuzzy relationsRenato de Mori, P. Torasso. 565-568 [doi]
• Duration as a syntactic boundary cue in ambiguous sentencesNina H. MacDonald. 569-572 [doi]
• A digital filter bank for spectral matchingDennis H. Klatt. 573-576 [doi]
• Phonological rule testing of conversational speechBeatrice T. Oshika. 577 [doi]
• Probabilistic lexical retrieval component with embedded phonological word boundary rulesJohn W. Klovstad. 578 [doi]
• Intermodulation distortion in hearing aids: The need for measurement standards and inherent complicationsHoward C. Schweitzer, G. Donald Causey. 579-582 [doi]
• Protocol for prescriptive fitting of a wearable master hearing aidH. Levitt, R. E. C. White, S. B. Resnick. 583-585 [doi]
• Realtime recognition of unvoiced fricatives in continuous speech to aid the deafD. A. MacKinnon, H. C. Lee. 586-589 [doi]
• A real time spectrograph with implications for speech training for the deafL. C. Stewart, Wilbur D. Larkin, Robert A. Houde. 590-593 [doi]
• Speech perception via the tactile mode: Progress reportFrank A. Saunders, William A. Hill, Carol A. Simpson. 594-597 [doi]
• Tactile stimulation as an aid for the deaf in production and reception of speech: Preliminary studiesMoise H. Goldstein Jr., Rachel E. Stark, Grace H. Yeni-Komshian, D. G. Grant. 598-601 [doi]
• Analysis of consonant recognition scores of congenital sensorineural hearing impairedBarbara Franklin. 602-605 [doi]
• Two sensory aids having profound effects on the blindWilliam De l'Aune, Chester Lewis, Mary Dolan, Thomas Grimmelsman, Walter Needham. 606-610 [doi]
• An instrumentation for facilitating easy onset patterns for stutterersJoseph G. Agnello, Ernest M. Weiler, Max F. Farley. 611-612 [doi]
• Acoustical analysis for voice disordersShizuo Hiki, Satoshi Imaizumi, Minoru Hirano, Hideaki Matsushita, Yuki Kakita. 613-616 [doi]
• Some properties of formant frequencies of vowels uttered by deaf and hard of hearing childrenShizuo Hiki, Ryuzaemon Kagami. 617 [doi]
• Hardware considerations in FFT processorsJ. P. Agrawal, Jacob Ninan. 618-621 [doi]
• Automatic generation of time efficient digital signal processing softwareL. Robert Morris. 622-625 [doi]
• Programmable communications terminal optimization using systematic digital signal compression proceduresC. M. Walter. 626-629 [doi]
• Simple hybrid systems for accurate synthesis and analysis of harmonic spectraS. T. Scott, Howard A. Stromberg. 630-632 [doi]
• A digital loop communication system applied to signal processing and speech researchF. C. Pirz. 633-635 [doi]
• A digital signal processing systemAbraham Peled. 636-639 [doi]
• The programmable array processorJ. Robinson, J. Welch, C. Teacher. 640-643 [doi]
• A new portable stand alone digital processorE. Hanson. 644-646 [doi]
• A two's complement pipeline multiplierEdmund K. Cheng, Carver A. Mead. 647-650 [doi]
• GASP: A programmable signal processorJohn A. V. Rogers. 651 [doi]
• Sonar design in the real ocean: Multipath limitations on sonar performanceR. J. Urick. 652-655 [doi]
• Statistical properties of underwater acoustic ambient noise fieldsStanley L. Adams, William John Jobst. 656-659 [doi]
• Motion induced coherence degradation in passive systemsAlbert A. Gerlach. 660-663 [doi]
• Dispersive properties of the underwater acoustic channelStanley L. Adams, J. Doubek. 664-667 [doi]
• Normal mode theory of underwater sound propagation in a range dependent environmentR. D. Graves, Anton Nagl, Herbert Überall, G. L. Zarur. 668-670 [doi]
• Sonar design in the real ocean: Target, background, and own ship limitations on sonar performanceT. C. Slotwinski. 671-674 [doi]
• The passive sonar equation - effects of additive interferenceR. F. Tiel. 675-678 [doi]
• Wideband target strength measurementsM. A. Cosgrove. 679-681 [doi]
• An automated solution for the wide band sonar equationD. E. Nelson. 682-685 [doi]
• A text-to-speech system based entirely on rulesRolf Carlson, Björn Granström. 686-688 [doi]
• Speech synthesis from english text: A progress reportC. H. Coker, S. A. Webber. 689 [doi]
• Automatic generation of control signals for a parallel formant speech synthesizerPeter M. Seeviour, John N. Holmes, Michael W. Judd. 690-693 [doi]
• Speech synthesis by dyads and automatic intonation processingDanielle Larreur, Françoise Emerard. 694-697 [doi]
• Computer synthesis of MandarinChing V. Suen. 698-700 [doi]
• A comprehensive model for fundamental frequency generationJonathan Allen, Douglas D. O'Shaughnessy. 701-704 [doi]
• Improving synthetic speech quality using binaural reverberationSteven F. Boll, Ercolino Ferretti, Tracy Petersen. 705-708 [doi]
• Multiplex-vocoder for voice responseUwe Dibbern. 709-712 [doi]
• Speech processing by splicing of autocorrelation functionJouji Suzuki. 713-716 [doi]
• A multi-channel digital audio laboratory facilityJames F. McGill. 717-720 [doi]
• Speech resynthesis with a hardware synthesizerLei F. Willems. 721 [doi]
• An application of the linear prediction technique to efficient coding of speech segmentsGian Antonio Mian, F. Morgantini, Carlo Offelli. 722 [doi]
• Cascade realization of digital inverse filter for extracting speaker dependent featuresV. V. S. Sarma, B. Yegnanarayana. 723-726 [doi]
• Text independent speaker recognition using orthogonal linear predictionMarvin R. Sambur. 727-729 [doi]
• On the connection of some time characteristics of speech signal with the individuality of voiceG. S. Ramishvili, M. A. Tushishvili. 730-733 [doi]
• Feature evaluation and selection for an on-line, adaptive speaker verification systemWen C. Lin, Sasi K. Pillay. 734-737 [doi]
• Automatic speaker recognition by computersErnst Bunge. 738 [doi]
• Speaker identification by multivariable linear prediction analysisFrank Fallside. 739 [doi]
• Speech - a possible indicator of physical stressH. S. Hayre. 740 [doi]
• Analysis of techniques for processing parallel signal outputsThomas C. Cantwell, Richard D. Wilmot. 741-744 [doi]
• Design procedure for improving the usable bandwidth of an MTI radar signal processorRonald C. Houts, Donald W. Burlage. 745-748 [doi]
• Signal processing for feature extraction and pattern recognitionBruce A. Eisenstein, John Fehlauer. 749-752 [doi]
• Frequency-variant optical systems for signal analysis and processingWilliam T. Rhodes, James M. Florence. 753-755 [doi]
• Performance of a range-ambiguous MTI and doppler filter systemJames K. Hsiao. 756-759 [doi]
• Digital signal processing techniques in truck tire vibration and sound analysisAllen C. Eberhardt, W. F. Reiter. 760-763 [doi]
• The application of cepstrum technique in power cable fault detectionChiou-Shiun Chen, Louis E. Roemer. 764-767 [doi]
• Plane averaging signal processing on radar using DFT technologyTakeru Irabu, Yuichi Tomita, Toshihiko Hagisawa, Eiichi Kiuchi. 768-771 [doi]
• Is four channel a quadrifizzle?J. Robert Ashley. 772-776 [doi]
• Noise measurement to ensure compatible land useC. E. Wilson. 777-778 [doi]
• Theoretical aspect of electromechanical transducersHerbert H. Ernyei. 779-785 [doi]
• A meteor infrasound recording systemMichael D. Watson, Brian A. McIntosh, Douglas O. Revelle. 786-789 [doi]
• The visual component of speech reception in conference centersLester L. Boyer. 790-793 [doi]
• Vibration pick-up type ear microphoneH. Ono, S. Saito, S. Mori. 794 [doi]