Table of Contents

• Studies on pattern recognition approach to voiced-unvoiced-silence classificationV. V. S. Sarma, D. Venugopal. 1-4 [doi]
• Improvement of voicing decisions by use of contextEdward P. Neuburg. 5-7 [doi]
• Epoch extraction from linear prediction residualT. V. Ananthapadmanabha, B. Yegnanarayana. 8-11 [doi]
• Maximum likelihood pitch estimation using state-variable techniquesRobert J. McAulay. 12-14 [doi]
• Interactive digital inverse filtering and its relation to linear prediction methodsMelvyn J. Hunt, John S. Bridle, John N. Holmes. 15-18 [doi]
• A new method of Cepstrum analysis by using comb lifterGen Ooyama, Sigeru Katagiri, Ken'iti Kido. 19-22 [doi]
• Arma modeling applied to linear prediction of speechThomas Cairns, William A. Coberly, David F. Findley. 23-26 [doi]
• Study of an adaptive lattice structure for linear prediction analysis of speechThomas E. Carter. 27-30 [doi]
• Real-zeros in pitch detectionM. Habib, D. Robinson, W. David Sincoskie. 31-34 [doi]
• Identification of nonstationary components by complex spectral zero analysisJ. Bee Bednar, William A. Coberly. 35-38 [doi]
• Two-dimensional processing of spectrogram dataJohn W. Woods, Vinay K. Ingle. 39-42 [doi]
• A high-quality all-digital sound spectrograph developed for speech signal analysisJohn N. Holmes, Michael W. Judd, David H. Walesby. 43-46 [doi]
• Realization of digital transfer functions using cascaded normalized two-pairsKalyan Mondal, Sanjit K. Mitra. 47-50 [doi]
• Calculation of all equivalent and canonic 2nd order digital filter structuresE. Lueder, K. Haug. 51-54 [doi]
• State-structures and minimal state-structures for arbitrary digital filtersKalyan Mondal, S. Chakrabarti, Sanjit K. Mitra. 55-57 [doi]
• Realization of a class of two-dimensional analog ladders with applications to wave digital filtersM. Omair Ahmad, C. H. Reddy, Venkatanarayana Ramachandran, M. N. Shanmukha Swamy. 58-61 [doi]
• A figure of merit for digital filtersL. J. Fruit. 62-66 [doi]
• Formats in audio bandwidth processing applicationsE. Cohler, J. Storer, R. Borgioli. 67-70 [doi]
• Digital filter realizations without overflow oscillationsWilliam L. Mills, Clifford T. Mullis, Richard A. Roberts. 71-74 [doi]
• An approach to eliminate roundoff errors in digital filtersAhmad I. Abu-El-Haija, Allen M. Peterson. 75-78 [doi]
• Selection of data windows for digital signal processingK. M. M. Prabhu, J. P. Agrawal. 79-82 [doi]
• Adaptive lattice methods for linear predictionJohn Makhoul, R. Viswanathan. 83-86 [doi]
• An adaptive lattice structure for noise-cancelling applicationsLloyd J. Griffiths. 87-90 [doi]
• Non-optimal convergence of adaptive LMS with uncorrelated dataR. Jeffrey Keeler. 91-95 [doi]
• Selecting the length of an adaptive transversal filterWilliam S. Hodgkiss. 96-99 [doi]
• An adaptive equalizer with significantly reduced number of operationsDietrich Maiwald, Hans-Peter Kaeser, Felix Closs. 100-104 [doi]
• Real-time signal processing for unbiased system identificationJudith G. Claassen, Wiley E. Thompson. 105-108 [doi]
• Adaptive data orthogonalizationNorman L. Owsley. 109-112 [doi]
• Some mathematical results on the effects on digital adaptive filters of implementation errors and noiseAlan Weiss, Debasis Mitra. 113-117 [doi]
• Simple adaptive IIR filteringJohn R. Treichler, Michael G. Larimore, C. Richard Johnson Jr.. 118-122 [doi]
• A short-term sequential regression algorithmNasir Ahmed, Donald R. Hummels, Michael L. Uhl, David L. Soldan. 123-126 [doi]
• On the maximum achievable conversion efficiency on a parametric acoustic arrayFrancis Hugh Fenlon. 127-129 [doi]
• Vibration sensitivity of the parametric acoustic receiving arrayC. Richard Reeves, Voldi E. Maki, Tommy G. Goldsberry, David F. Rohde. 130-133 [doi]
• Errors in array response calculations due to incorrectly folding the vertical acoustic arrival structure into beam patternJohn J. Cornyn. 134-136 [doi]
• Control of the directional response of acoustic transducer array elements on curved surfaces by modification of the diffracted fieldFrancis Hugh Fenlon, Geoffrey L. Wilson. 137-140 [doi]
• Array processing using the frequency-wavenumber approachMelvin J. Hinich. 141-142 [doi]
• Effect of sampling errors on array gainCharles R. Baker, Louis R. Chow. 143-147 [doi]
• Synthetic array processing for underwater mapping applicationsHenry E. Lee. 148-151 [doi]
• A flexible towed sonar for ocean acoustic measurementsWilliam Barry, Darrell R. Jackson, Jim O. P. Schultz. 152-154 [doi]
• An audio response method for CAI servicesNaoki Ishii, Ken'ya Murakami. 155-158 [doi]
• Speech synthesis using real-time softwareDavid J. Quarmby, D. H. Midgeley, J. Ruby. 159-162 [doi]
• A mixed-source model for speech compression and synthesisJohn Makhoul, R. Viswanathan, Richard M. Schwartz, A. W. F. Huggins. 163-166 [doi]
• Implementation of a channel vocoder synthesizer using a fast, time-multiplexed digital filterDenis L. Baggi. 167-170 [doi]
• An excitation function for LPC synthesis which retains the human Glottal phase characteristicsDavid Y. Wong, John D. Markel. 171-174 [doi]
• Votrax real time hardware for phoneme synthesis of speechRichard T. Gagnon. 175-178 [doi]
• Speech generation through waveform synthesisM. Baumwolspiner. 179-182 [doi]
• Comparisons of system identification methods in the presence of high noise levels and bandlimited inputsLawrence R. Rabiner, Ronald E. Crochiere, Jont B. Allen. 183-187 [doi]
• Linear prediction and maximum entropy spectral analysis of finite bandwidth signals in noiseS. Thomas Alexander, Edgar H. Satorius, James R. Zeidler. 188-191 [doi]
• Comparative study of iterative deconvolution algorithmsRussell M. Mersereau, Ronald W. Schafer. 192-195 [doi]
• Image restoration of space variant blurs by sectioned methodsH. Joel Trussell, Bobby R. Hunt. 196-198 [doi]
• Nonlinear filter for inversion of channel distortionSubhash C. Kwatra, Vijay K. Jain. 199-202 [doi]
• Observations on linear estimationLeland B. Jackson, Frank K. Soong. 203-207 [doi]
• New stochastic realization algorithms for identification of ARMA modelsGérard Alengrin, Gérard Favier. 208-213 [doi]
• Analysis and representation of composite signals by cepstral inverse filteringJeff H. Derby. 214-217 [doi]
• Multilevel crossing rates for automated signal classificationR. J. Mitchell, R. C. Gonzalez. 218-222 [doi]
• Target identification from radar signaturesRobert D. Strattan. 223-227 [doi]
• Multiband FIR digital filter design algorithm for radar clutter suppressionBrian P. Holt, Ronald C. Houts. 228-231 [doi]
• Typical performance characteristics of a two-dimensional CFARNeal B. Lawrence, Jerry D. Moore. 232-235 [doi]
• A technique for pole-zero modeling of complex-valued autocorrelationsO. L. Godwin, Vijay K. Jain. 236 [doi]
• A programmable sonar signal processorC. R. Carter, Simon Haykin, H. C. Chan. 237-240 [doi]
• Reconnaissance sonar for deep ocean seamount detectionMartin G. Fagot. 241-244 [doi]
• An at-sea system for the prediction of underwater sound propagationDuane C. Tate. 245-247 [doi]
• Inherent errors in sonar range predictionHarold R. Hall. 248-251 [doi]
• Design considerations for feedback amplifiersW. Marshall Leach Jr.. 252-254 [doi]
• Audibility of transient intermodulation distortionMargit Petri-Larmi, Matti Otala, Eero Leinonen, Jorma Lammasniemi. 255-262 [doi]
• Omitted factors in audio circuit designJohn Curl. 263-266 [doi]
• Computer analysis of transient distortion and low transient distortion amplifier designKenneth F. Leonard. 267-269 [doi]
• Electroacoustic distortions: Multidimensional analysis of hearing aid transduced speech and musicA. Yonovitz, Barbara Jill Bickford, Joseph Lozar, Dianne R. Ferrell. 270-274 [doi]
• A comparison between two approaches to automatic speaker recognitionL. Fasolo, Gian Antonio Mian. 275-278 [doi]
• Extraction of speaker-specific features from spoken code sentencesPeter Jesorsky, Ulrich Höfker, Maati Talmi. 279-282 [doi]
• New techniques for text-independent speaker identificationLarry Pfeifer. 283-286 [doi]
• Text-independent speaker identification from a large linguistically unconstrained time-spaced data baseJohn D. Markel, Steven B. Davis. 287-290 [doi]
• Text-independent speaker identification based on piecewise canonical discriminant analysisHiroshi Matsumoto, Tadamoto Nimura. 291-294 [doi]
• Automatic speaker identification for a large number of speakersH. M. Dante, V. V. S. Sarma. 295-298 [doi]
• 2400/16, 000 Bps Multirate voice processorAaron J. Goldberg. 299-302 [doi]
• High quality adaptive predictive coding of speechMichael G. Berouti, John Makhoul. 303-306 [doi]
• 9.6/7.2 Kbps Voice excited predictive coder (VEPC)Daniel Esteban, Claude Galand, Daniel Mauduit, Jean E. Menez. 307-311 [doi]
• An optimal adaptation logic for delta modulationSamar K. Chakravarty, Pradip K. Srimani. 312-315 [doi]
• Forward-adaptive delta modulator without explicit transmission of step sizePredrag M. Petrovic. 316-319 [doi]
• 32 Kbps CCITT Compatible split band coding schemeDaniel Esteban, Claude Galand. 320-325 [doi]
• Evaluation of LPC/CVSD tandem connectionsThomas P. Barnwell III, Ronald W. Schafer, Aubrey M. Bush. 326-329 [doi]
• A spectral enhancement procedure for the wideband/Narrowband tandemL. E. Bergeron. 330-333 [doi]
• Towards a variable frame rate speech transmission system with frame selection by time-domain segmentation - a status reportWolfgang J. Hess, Josef Heiler. 334-337 [doi]
• Two dimensional data compression of speechAnil K. Jain, Demitri M. Maroulis. 338-340 [doi]
• Two-dimensional speech compressionJames M. Alsup, Harper J. Whitehouse. 341-344 [doi]
• High resolution autoregressive spectrum analysis using noise power cancellationLarry Marple. 345-348 [doi]
• High resolution spectral analysis of sinusoids in correlated noiseEdgar H. Satorius, S. Thomas Alexander. 349-351 [doi]
• Frequency estimation by linear predictionLeland B. Jackson, Donald W. Tufts, Frank K. Soong, Rahul M. Rao. 352-356 [doi]
• Improvement of autoregressive spectral estimates in the presence of noiseSteven M. Kay. 357-360 [doi]
• The accuracy of center frequency estimators using linear predictive methodsGervasio Prado, Paul Moroney. 361-364 [doi]
• Complex covariance/Maximum entropy doppler estimates for pulsed CO2lidarR. Jeffrey Keeler, Robert W. Lee. 365-368 [doi]
• An error formula for iterative prefiltering frequency estimatesFrancis Landolf. 369-371 [doi]
• Extrapolation and spectral estimation for bandlimited signalsDean P. Kolba, Thomas W. Parks. 372-374 [doi]
• Constant-Q signal analysis and synthesisJames E. Youngberg, Steven F. Boll. 375-378 [doi]
• Spectrum analysis using frequency-domain adaptive windowingJohn E. Timm. 379-382 [doi]
• Spectrum estimation of non-uniform sampled dataN. C. Mohanty, L. O. Krause. 383-386 [doi]
• Input signals for sampling and identification of ecological systemsCarolyne M. Gowdy. 387-390 [doi]
• Input, signals and control in ecosystemsJames Hill, Susan L. Durham. 391-397 [doi]
• Extension of Levins loop analysis to transient and periodic disturbancesRobert H. Flake. 398-401 [doi]
• Estimating Oxygen demand in aquatic ecosystemsT. C. Vorce, R. J. Mulholland. 402-404 [doi]
• Gaps in the technology of speech understandingWayne A. Lea, June E. Shoup. 405-408 [doi]
• Computing relative redundancy to measure grammatical constraint in speech recognition tasksM. Mohan Sondhi, Stephen E. Levinson. 409-412 [doi]
• Dynamic programming, the viterbi algorithm, and low cost speech recognitionGeorge M. White. 413-417 [doi]
• Automatic recognition of continuously spoken sentences from a finite state grammerLalit R. Bahl, James K. Baker, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer. 418-421 [doi]
• Recognition of continuously read natural corpusLalit R. Bahl, James K. Baker, Paul S. Cohen, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer. 422-424 [doi]
• A voice-input programming system using basic-like languageYasuhisa Niimi, Yutaka Kobayashi. 425-428 [doi]
• The Dawid speech recognition systemR. D. Glave, G. van der Giet. 429-432 [doi]
• Fuzzy semantic network for a speech understanding system - an experimental studyLorenza Saitta. 433-436 [doi]
• A simple vocoderLuis F. Rocha. 437-440 [doi]
• Low bit rate cepstral vocoder using the log magnitude approximation filterSatoshi Imai. 441-444 [doi]
• CCD CZT Spectral analysis applied to real time homomorphics speech analysis-synthesisThomas F. Quatieri. 445-449 [doi]
• Implementation of a pole-zero analysis - synthesis system for speechRichard L. Cann, Kenneth Steiglitz. 450-453 [doi]
• A variable frame length linear predictive coderJohn M. Turner, Bradley W. Dickinson. 454-457 [doi]
• A method for reducing the transmission rate of a channel vocoder by using frame interpolationEdward McLarnon. 458-461 [doi]
• Providing channel error protection for a 2400 bps linear predictive coded voice systemJesse W. Fussell, Barry M. Abzug, Paul W. Boudra Jr., Michael C. Cowing. 462-465 [doi]
• Channel coding considerations for digital speech encoded by linear predictionG. Robert Redinbo. 466-471 [doi]
• Linear predictive coding of speech signals in a high ambient noise environmentHidefumi Kobatake, Junta Inari, Shin-ichi Kakuta. 472-475 [doi]
• Linear prediction techniques in the Walsh spectral domain for speech analysis and synthesisM. R. Ashouri, Anthony G. Constantinides. 476-479 [doi]
• Application of canonical coordinate methods to the characterization of a family of error minimizing signal compression techniquesCharlton M. Walter. 480-482 [doi]
• An estimate of the order of an optimal FIR band-pass digital filterFred Mintzer, Bede Liu. 483-486 [doi]
• Digital filters with prescribed zerosYrjö Neuvo, Tapio Saramäki, Robert A. Gabel. 487-490 [doi]
• Synthesis of digital filters with very low sensitivity of the frequency response to the change of the cofficientsPierre Duhamel. 491-494 [doi]
• Design of IIR filters using Pseudo-Boolean methodsRakesh K. Patney, S. C. Dutta Roy. 495-498 [doi]
• Design and applications of uniform digital bandpass filter banksMadihally J. Narasimha, Allen M. Peterson. 499-503 [doi]
• A multiplexing scheme for multirate digital filtering with half-band filtersMaurice G. Bellanger. 504-507 [doi]
• New class of recursive digital filters for decimationHoracio G. Martinez, Thomas W. Parks. 508-511 [doi]
• An optimal filter design for variable sampling ratesJames E. Heller. 512-515 [doi]
• Digital lowpass filtering using the discrete Hilbert transformS. C. Dutta Roy, Anurag Agrwal. 516-519 [doi]
• Nonstationary signal processing and model validationJames R. Rowland, Willard M. Holmes. 520-523 [doi]
• Recursive derivation of reflection coefficients from noisy seismic dataN. E. Nahl, J. M. Mendel, L. M. Silverman. 524-528 [doi]
• On digital signal modelling and classification with the teleseismic dataC.-H. Chen. 529-531 [doi]
• Acoustic wave propagation by finite-difference techniquesR. M. Alford, K. R. Kelly, N. D. Whitmore. 532 [doi]
• Ultrasonic measurements of defects in metals using cepstral processingRamesh Shankar, Robert J. McDonough. 533-537 [doi]
• Digital recording of ultrasonic signalsA. L. Frisillo, C. F. Hadley. 538-540 [doi]
• A comparison of linear prediction, FFT, and zero-crossing analysis techniques for vowel recognitionPatrick F. Castelaz, Russel J. Niederjohn. 541-545 [doi]
• Speaker-independent vowel indetification in continuous speechVishwa Gupta, J. Kent Bryan, John N. Gowdy. 546-548 [doi]
• A probabalistic vector model for identification of intervocalic stop consonantsT. J. Edwards. 549-552 [doi]
• A comparative study of phonemic recognition by discrete orthogonal transformsH. A. Barger, K. R. Rao. 553-556 [doi]
• A phoneme recognition system based on human auditionC. L. Searle, J. Zachary Jacobson, S. G. Rayment. 557-560 [doi]
• Application of the subspace method to speech recognitionMatti Jalanko, Teuvo Kohonen. 561-564 [doi]
• Application of novelty filter to segmentation of speechSeppo Haltsonen, Matti Jalanko, Kalle-J. Bry, Teuvo Kohonen. 565-568 [doi]
• Signal/Quantizing-distortion ratio measurements of fast-adaptive delta modulation systemsL. D. J. Eggermont, E. C. Dijkmans. 569-572 [doi]
• Predictive coding of speech signals and subjective error criteriaBishnu S. Atal, Manfred R. Schroeder. 573-576 [doi]
• Speech synthesis by linear interpolation of spectral parameters between dyad boundariesChristine H. Shadle, Bishnu S. Atal. 577-580 [doi]
• Perceptual and objective evaluation of speech processed by adaptive differential PCMBarbara J. McDermott, Carlo Scagliola, David J. Goodman. 581-585 [doi]
• A study of complexity and quality of speech waveform codersJosé M. Tribolet, Peter Noll, Barbara J. McDermott, Ronald E. Crochiere. 586-590 [doi]
• Objective speech quality evaluation of narrowband LPC vocodersR. Viswanathan, William Russell, John Makhoul. 591-594 [doi]
• Statistical correlation between objective and subjective measures for speech qualityThomas P. Barnwell III, Aubrey M. Bush. 595-598 [doi]
• Estimation of LPC coefficients from speech waveforms degraded by additive random noiseJae S. Lim. 599-601 [doi]
• An investigation of several frequency-domain processing methods for enhancing the intelligibility of speech in wideband random noiseRobert A. Curtis, Russel J. Niederjohn. 602-605 [doi]
• Suppression of noise in speech using the saber methodSteven F. Boll. 606-609 [doi]
• LMS Adaptive filtering for enhancing the quality of noisy speechMarvin R. Sambur. 610-613 [doi]
• Two dimensional filtering using fermat number transformsR. H. VanderKraats, Anastasios N. Venetsanopoulos. 614-618 [doi]
• Signal processing with number theoretic transforms and limited word lengthsPierre R. Chevillat, Felix Closs. 619-623 [doi]
• q+ 1Eric Dubois 0002, Anastasios N. Venetsanopoulos. 624-627 [doi]
• Simulation of large length filters using fermat number transformsN. Sridhar Reddy, V. Umapathi Reddy. 628-631 [doi]
• An inplace self recordering FFTJames K. Beard. 632-633 [doi]
• Real-factor FFT algorithmsK. M. Cho, Gabor C. Temes. 634-637 [doi]
• New algorithms for convolution and DFT based on polynomial transformsHenri J. Nussbaumer. 638-641 [doi]
• M-Adic invariant filtersB. P. Agrawal, Kishan Shenoi. 642-645 [doi]
• Reconstruction of signals from their linear mapping imageJames A. Cadzow. 646-650 [doi]
• Linear detection filtering for the context of a least-squares estimator for signal processing applicationsWilliam H. Haas, Claude S. Lindquist. 651-654 [doi]
• Dealiasing of the spectra of sampled noiseH. R. Bilger, J. P. Nougier. 655-658 [doi]
• Underwater sound arrival angle estimation by multiple cross-correlation measurementsH. J. Young. 659-664 [doi]
• The least squares estimation of time delay and its use in signal detectionY. T. Chan, R. V. Hattin, J. B. Plant. 665-669 [doi]
• Confidence bounds for magnitude-squared coherence estimatesEverett H. Scannell Jr., G. Clifford Carter. 670-673 [doi]
• An all-digital phase-measurement techniques using clipped-quadrature correlationJames L. Roberts. 674-677 [doi]
• Detection performance of the smooth coherence transform (SCOT)J. P. Kuhn. 678-683 [doi]
• Object classification using sonar dataDavid J. Quarmby, Geoffrey M. Duck. 684-687 [doi]
• A study of stochastic processes associated with sonar detectionJude Franklin. 688-691 [doi]
• Acoustic transfer function of the ocean for a motional sourceAlbert A. Gerlach. 692-695 [doi]
• Display and interpretation of a time-spread underwater acoustic channel's bandlimited impulse responseRaymond L. Veenkant. 696-699 [doi]
• Some experiments with a syntax directed speech recognition systemStephen E. Levinson, Aaron E. Rosenberg. 700-703 [doi]
• Performance analysis of syntactic-recognizer of isolated wordsSilvano Rivoira, Pietro Torasso. 704-707 [doi]
• Recognition of monosyllabic words in continuous sentences using composite word templatesPaul Mermelstein. 708-711 [doi]
• An automatic word spotting system for conversational speechMark F. Medress, Timothy Diller, Dean R. Kloker, Larry L. Lutton, Henry N. Oredson, Toby E. Skinner. 712-717 [doi]
• Directory assistance by means of automatic recognition of spoken spelled namesAaron E. Rosenberg, C. E. Schmidt. 718-721 [doi]
• An approach to speech recognition using syllabic decision unitsGünther Ruske, Thomas Schotola. 722-725 [doi]
• A word recognition method from a classified phoneme string in the Lithan speech understanding systemSei-Ichi Nakagawa, Toshiyuki Sakai. 726-730 [doi]
• Nearest neighbour decision rule for vowel and digit recognitionT. K. Raja, B. Yegnanarayana. 731-734 [doi]
• Spoken word recognition system for unlimited speakersKen'iti Kido, Jouji Miwa, Shozo Makino, Yoshihiro Niitsu. 735-738 [doi]
• Two-dimensional signal processing from hexagonal rastersRussell M. Mersereau. 739-742 [doi]
• Digital decomposition and representation of video signals using projection theoryGerald M. Flachs, Wiley E. Thompson, Yee Hsun U, Steve Szymanski. 743-746 [doi]
• On generation of two-dimensional dataN. C. Mohanty. 747-750 [doi]
• Inversion of block - Toeplitz matrices using bivariate szego polynomialsJames H. Justice. 751-752 [doi]
• Two-dimensional zoom FFTRonald F. Stork, Elmer A. Hoyer. 753-756 [doi]
• Recursive digital filters in image processingA. Chottera, Graham A. Jullien. 757-760 [doi]
• Design of stable 2-D half-plane recursive filters using spectral factorizationMichael P. Ekstrom, Richard E. Twogood, John W. Woods. 761-764 [doi]
• Design of inherently stable two-dimensional recursive filters imitating the behaviour of one-dimensional analog filtersAmar M. Ali, Anthony G. Constantinides. 765-768 [doi]
• An algorithm for testing stability of two-dimensional digital recursive filtersGary A. Shaw. 769-772 [doi]
• Computer aided generation of two dimensional transfer functions from one dimensional transfer functionsP. Karivaratharajan, M. N. Shanmukha Swamy. 773-776 [doi]
• Design of semicausal two-dimensional recursive filtersHyokang Chang, Jake K. Aggarwal. 777-781 [doi]
• A well suited two-dimensional linear recursive filter for image processingTzeng-Tung Hwang. 782-787 [doi]
• A simple hardware implementation of digital notch filtersAhmad I. Abu-El-Haija, Madihally J. Narasimha, Allen M. Peterson. 788-791 [doi]
• Digital phase locked loopC. P. Reddy, Erik Fountain. 792-795 [doi]
• Parallel counter design using four-valued threshold logicK. Wayne Current, Douglas A. Mow. 796-799 [doi]
• An error anaylsis of a FFT implementation using the residue number systemBen-Dau Tseng, William C. Miller, Graham A. Jullien, J. J. Soltis, A. Baraniecka. 800-803 [doi]
• Techniques for residue-to-analog conversion for high data rate digital filteringW. Kenneth Jenkins. 804-807 [doi]
• 20Channel 300 MHz bandwidth digital spectrum analyzerGeorge A. Morris Jr., Helmut C. Wilck. 808-811 [doi]
• A 16-bit microprocessor-based digital filter architectureJohn D. Mackay, Harvey F. Silverman. 812-815 [doi]
• The compass block-diagram compiler: A new development in signal-processor programmingJoseph R. Fisher, Martin E. Kaliski. 816-819 [doi]
• Impulse response testing of acoustic spacesRichard C. Cabot. 820-823 [doi]
• Super directive spectrum analyzer by use of moving microphonesKen'iti Kido, Masaaki Ishigame, Masato Abe. 824-827 [doi]
• Free-field measurements for a loudspeaker system in a normal room-Using digital signal processing techniquesTsuneo Nitta, Masatoshi Tanaka. 828-831 [doi]
• Correction of near-field acoustic measurements made with arbitrary measuring transducersW. Marshall Leach Jr.. 832-835 [doi]
• Reflection of spherical sound wave from a rigid sphereYoshiro Miida. 836-839 [doi]
• Statistical properties of the poisson reverberation envelopeAli Zolfaghari. 840-845 [doi]
• On an extension of A. N. Krylov's numerical method for determining the frequencies of small vibrations of systems with dampingRobert Kalaba, Elena Zagustin. 846-847 [doi]