Abstract is missing.
- 2-D FIR filter design from independent "Small" generating kernels using a mean square and Tchebyshev error criterionOlivier D. Faugeras, Jean-François Abramatic. 1-4 [doi]
- A comparison of different window formulations for two-dimensional FIR filter designTheresa C. Speake, Russell M. Mersereau. 5-8 [doi]
- Stabilization of two-dimensional recursive filtersHyokang Chang, Jake K. Aggarwal. 9-12 [doi]
- Generation of two-dimensional digital functions without non-essential singularities of the second kindC. H. Reddy, P. Karivaratharajan, M. N. S. Swamy, Venkatanarayana Ramachandran. 13-19 [doi]
- Why filter recursively in two dimensions?Richard E. Twogood, Michael P. Ekstrom. 20-23 [doi]
- Design of 2-D recursive filters with separable denominator transfer functionsJean-François Abramatic, François Germain, Emmanuel Rosencher. 24-27 [doi]
- Space-domain design of two-dimensional recursive digital filtersGary A. Shaw, Russell M. Mersereau. 28-31 [doi]
- The design of 2-D recursive filters in the 2-D reflection coefficient domainThomas L. Marzetta. 32-35 [doi]
- Two-dimensional half-plane recursive filter designGiovanni Garibotto. 36-39 [doi]
- Design of stable symmetric and non-symmetric half-plane digital recursive filtersP. A. Ramamoorthy, Len T. Bruton. 40-43 [doi]
- On synthesizing natural-sounding speech by linear predictionBishnu S. Atal, Nancy David. 44-47 [doi]
- Linear prediction of formants for low bit rate digital speech transmissionSassan Ahmadi, Anthony G. Constantinides. 48-51 [doi]
- A two-step speech compression system with vector quantizingAndres Buzo, Augustine H. Gray Jr., Robert M. Gray, John D. Markel. 52-55 [doi]
- A mixed-phase homomorphic vocoderThomas F. Quatieri. 56-59 [doi]
- Narrowband LPC speech transmission over noisy channelsE. Blackman, R. Viswanathan, William Russell, John Makhoul. 60-63 [doi]
- A robust vocoder with pitch-adaptive spectral envelope estimation and an integrated maximum-likelihood pitch estimatorDouglas B. Paul. 64-68 [doi]
- A linear predictive vocoder with new pitch extraction and exciting sourceAkira Kurematsu, Hikoichi Ishigami, Seishi Kitayama, Fumihiro Yato, Junso Tamura. 69-72 [doi]
- A novel vocoder concept based on discrete time acoustic tubesArild Lacroix, Bela Makai. 73-76 [doi]
- Linear prediction of transformed speechM. R. Ashouri. 77-80 [doi]
- An analysis/Synthesis framework for transform coding of speechJosé M. Tribolet, Ronald E. Crochiere. 81-84 [doi]
- Computer recognition of stop consonantsPiero Demichelis, Renato de Mori, Pietro Laface, Mary O'Kane. 85-88 [doi]
- Automatic discrimination of fricative consonants based on human auditionB. P. Kimberley, C. L. Searle. 89-92 [doi]
- On sensitivity of vocal tract area functionsT. V. Sreenivas, T. V. Ananthapadmanabha. 93-96 [doi]
- Spectral classification of phonemes by learning subspacesTeuvo Kohonen, Gábor Németh, Kalle-J. Bry, Matti Jalanko, Heikki Riittinen. 97-100 [doi]
- Speech segmentation and recognition using syntactic methods on the direct signalMichel Baudry, B. Dupeyrat. 101-104 [doi]
- An experimental system for acoustic-phonetic decoding of continuous speechJean-Paul Haton, Claude Sanchez. 105-107 [doi]
- Automatic selection of phonemes from an equally spaced quasi-phoneme string by the entropy principleSeppo Haltsonen, Kalle-J. Bry. 108-111 [doi]
- A unique real-time speech decoder that operates from new perspectivesDonald C. Lokerson. 112-115 [doi]
- Experiments on spectrogram readingVictor W. Zue, Ronald A. Cole. 116-119 [doi]
- Testing sonar data for multivariate normalityCharles R. Baker. 120-123 [doi]
- A quantitative study of optimum and sub-optimum filters in the generalized correlatorJoseph C. Hassab, Ronald E. Boucher. 124-127 [doi]
- A parameter estimation approach to time delay estimationY. T. Chan, J. M. Riley, J. B. Plant. 128-131 [doi]
- Local estimation of delay parameter following robust detectionA. R. Pratt. 132-135 [doi]
- Detection performance of an adaptive processor in non-stationary noiseTerry Rickard, Mauro J. Dentino, James R. Zeidler. 136-139 [doi]
- Robust sequential detection of narrowband acoustic signals in noiseRoger F. Dwyer. 140-143 [doi]
- A phase-coherence detector/EstimatorRoberto Berezdivin, Robert Perl, Robert Braunstein. 144-147 [doi]
- Sources of and remedies for spectral line splitting in autoregressive spectrum analysisSteven M. Kay, Larry Marple. 151-154 [doi]
- Spectral analysis using prediction methodsW. Brandenburg. 155-158 [doi]
- Spectral line analysis by Pisarenko and Prony methodsLarry Marple. 159-161 [doi]
- Fourier-autoregressive spectral estimationSteven M. Kay. 162-165 [doi]
- Maximum entropy (adaptive) filtering applied to radar clutterCarey Gibson, Simon Haykin, Stanislav Kesler. 166-169 [doi]
- Digital signal processing for doppler radar signalsR. T. Schaefer, Ronald W. Schafer, Russell M. Mersereau. 170-173 [doi]
- Fast estimation of narrowband spectraUlrich Steimel. 174-177 [doi]
- A family of phase complementary filtersR. Linggard, B. D. V. Smith. 178-181 [doi]
- Spectral analysis using the Karhunen-Loeve transformJ. Ziegenbein. 182-185 [doi]
- Magnitude-phase relationships for short-time Fourier transforms based on Gaussian analysis windowsMichael R. Portnoff. 186-189 [doi]
- Extrapolation and spectral estimation for bandlimited, time-concentrated signalsDean P. Kolba, Thomas W. Parks. 190-193 [doi]
- Optimal estimation of essentially and strictly bandlimited signals and their spectrum by generalized splinesRui J. P. de Figueiredo. 194-199 [doi]
- A spectral subtraction algorithm for suppression of acoustic noise in speechSteven F. Boll. 200-203 [doi]
- Reduction of nonstationary acoustic noise in speech using LMS adaptive noise cancellingDennis Pulsipher, Steven F. Boll, Craig K. Rushforth, LaMar Timothy. 204-207 [doi]
- Enhancement of speech corrupted by acoustic noiseMichael G. Berouti, Richard M. Schwartz, John Makhoul. 208-211 [doi]
- A frequency domain noise cancelling preprocessor for narrowband speech communications systemsRobert D. Preuss. 212-215 [doi]
- Performance of LPC vocoders in a noisy environmentCharles F. Teacher, David C. Coulter. 216-219 [doi]
- LPC voice digitizer with background noise suppressionDonald P. Fulghum, J. E. Gunn III. 220-223 [doi]
- Maximum likelihood parameter estimation of noisy dataBruce R. Musicus, Jae S. Lim. 224-227 [doi]
- Estimating the parameters of a noisy all-pole process using pole-zero modelingWilliam J. Done, Craig K. Rushforth. 228-231 [doi]
- CCD adaptive filtering for robust LPC speech processingG. M. Borsuk, M. H. White. 232-234 [doi]
- An approach to speaker normalization in an automatic speech recognition systemJohannes Jaschul. 235-238 [doi]
- A new system for continuous speech recognition - preliminary resultsStephen E. Levinson, Aaron E. Rosenberg. 239-244 [doi]
- Automatic recognition of continuous digits sequences by means of segmentation and dynamic programmingJean-Paul Haton, Olivier Morel. 245-248 [doi]
- Recognition results for several experimental acoustic processorsLalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer. 249-251 [doi]
- The structure of a lexicon for a speech understanding systemRenato de Mori, Leonardo Lesmo, Marisa Poncini. 252-255 [doi]
- Phonetic word verificationTimothy Diller. 256-261 [doi]
- Performance statistics of the HEAR acoustic processorJanet M. Baker. 262-265 [doi]
- Techniques for recognition of spectrogram patterns based on dynamic modelingRui J. P. de Figueiredo, Thomas J. Brzustowicz. 266-268 [doi]
- Speech recognition in the context of two-way immediate person-machine interactionJoseph J. Mariani, Jean-Sylvain Liénard, G. Renard. 269-272 [doi]
- A review of adaptive antennasBernard Widrow. 273-278 [doi]
- An array optimization techniqueLouis A. Mole, Frank A. Andrews. 279-281 [doi]
- Adaptive array processing: Time vs. frequency domainWilliam S. Hodgkiss. 282-285 [doi]
- Minimax optimization of two-dimensional focused nonuniformly spaced arraysRussell P. Kraft, John F. McDonald, F. Ahlgren. 286-289 [doi]
- Effects of random shading, phasing errors and element failures on the beampatterns of line and planar arraysAzizul H. Quazi, Albert H. Nuttall. 290-293 [doi]
- A computer model for the analysis of source motion, the ocean environment, and interference effects on acoustic signal coherenceKenneth A. Faucher, James J. Foster. 294-297 [doi]
- The roles of integration time and acoustic multipaths in determining the structure of CW line spectraKenneth E. Hawker, Jack A. Shooter. 298-301 [doi]
- Impact of the ocean acoustic transfer function on the coherence of undersea propagationsAlbert A. Gerlach. 302-305 [doi]
- Influence of the spatial coherence of the background noise on high resolution passive methodsGeorges Bienvenu. 306-309 [doi]
- Audio signal processing with transversal filtersDouglas Preis. 310-313 [doi]
- Constant-Q analysis using the chirp z-transformJames M. Kates. 314-317 [doi]
- Design of a wideband, constant beamwidth, array microphone for use in the near-fieldF. C. Pirz. 318-321 [doi]
- Time domain measurement of loudspeaker driver parametersW. Marshall Leach Jr., Ronald W. Schafer, Thomas P. Barnwell III. 322-325 [doi]
- A new computer aided method for the complete acoustical design of broadcasting and recording studiosOscar J. Bonello. 326-329 [doi]
- Auditory backward inhibition can ruin a concert hallJ. Robert Ashley. 330-334 [doi]
- Constrained minimization of roundoff noise in fixed-point digital filtersDavid S. K. Chan. 335-339 [doi]
- Normal realizations of IIR digital filtersWilliam L. Mills, Clifford T. Mullis, Richard A. Roberts. 340-343 [doi]
- Reduced multiplier, low roundoff noise digital filtersMasud Arjmand, Richard A. Roberts. 344-346 [doi]
- Linear transformations for the design of digital and active filtersAmar M. Ali. 347-350 [doi]
- Coefficient quantization effects on pole locations for state model digital filtersJames D. Ledbetter, Rao Yarlagadda. 351-354 [doi]
- A state-space approach for elimination of limit cycles in digital filters with arbitrary structuresTatsuo Higuchi, Hiroski Takeo. 355-358 [doi]
- Passive cascaded lattice digital filtersAugustine H. Gray Jr.. 359-362 [doi]
- Simplified error models for digital filtersRolf Block, Arild Lacroix. 363-366 [doi]
- Narrowband recursive filters with error spectrum shapingDavid C. Munson Jr., Bede Liu. 367-370 [doi]
- Some considerations on coefficient sensitivity and noise in direct form IIR interpolators and decimatorsTor A. Ramstad. 371-374 [doi]
- Coefficient inaccuracy in FIR filtersAllen Gersho, B. Gopinath, Andrew M. Odlyzko. 375-377 [doi]
- Effects of finite coefficient precision on FIR filter spectraV. B. Lawrence, A. C. Salazar. 378-379 [doi]
- Accumulation of distortion in signal processing systemsDavid G. Messerschmitt. 380-383 [doi]
- Identification of complex autoregressive processesJ. A. Ponnusamy, Mandyam D. Srinath, Periagaram K. Rajasekaran. 384-387 [doi]
- Geometrical characterization of the canonical coordinate basis underlying a family of error minimizing signal compression techniquesCharlton M. Walter. 388-391 [doi]
- Inversion of signal operationsJames A. Cadzow. 392-397 [doi]
- The solution of discrete convolutions with a bounded error constraintAlbert Arcese. 398-400 [doi]
- An experimental study of the effects of noise on a class of iterative deconvolution algorithmsMark A. Richards, Ronald W. Schafer, Russell M. Mersereau. 401-404 [doi]
- System identification using a maximum-likelihood spectral matching techniqueGervasio Prado. 405-408 [doi]
- Spectral root homomorphic deconvolution systemJae S. Lim. 409-414 [doi]
- Evaluation of the Steiglitz algorithm for estimating the parameters of an ARMA processWilliam J. Done, Craig K. Rushforth. 415-418 [doi]
- Modulo-PCM: A new source coding schemeThomas Ericson, V. Ramamoorthy. 419-422 [doi]
- A generalized adaptive quantization system with a new reconstruction method, for noisy transmissionDebasis Mitra. 423-427 [doi]
- High-frequency regeneration in speech coding systemsJohn Makhoul, Michael G. Berouti. 428-431 [doi]
- Bit-rate-halving algorithm for PCM-encoded speech using a new bidimensional data compression schemeJean-Pierre Adoul, Sarto Morissette, Michel Rudko. 432-435 [doi]
- A high-gain DSI-ADPCM systemYohtaro Yatsuzuka. 436-441 [doi]
- Digital transmission of commentary-grade (7 kHz) audio at 56 or 64 kb/sJames D. Johnston, David J. Goodman. 442-444 [doi]
- Variable rate codingJohn J. Dubnowski, Ronald E. Crochiere. 445-448 [doi]
- A methodology for studying telephone amplitude distortion effects on narrowband speech processorsJohn D. Markel, Steven B. Davis, Ted H. Applebaum. 449-452 [doi]
- Optimizing predictive coders for minimum audible noiseBishnu S. Atal, Manfred R. Schroeder. 453-455 [doi]
- Talker variance and phonetic feature variance in diagnostic intelligibility scores for digital voice communications processorsCaldwell P. Smith. 456-459 [doi]
- Intelligibility of intervocalic consonants in noiseLouis C. W. Pols. 460-463 [doi]
- Optimum linear filter for speech communicationCraig R. Allen. 464-466 [doi]
- Subjective evaluation of SPAC in improving the quality of noisy speechMamoru Nakatsui. 467-470 [doi]
- The presentation of continuous speech with synchronous printed textDouglas C. Sargent, Andrew Malcolm. 471-474 [doi]
- The effects of Teflon injection on laryngeal dynamicsG. L. Bull, Michael M. E. Johns, W. E. McDonald, R. C. Bralley. 475-478 [doi]
- A conceptually unique speech training aid systemDonald C. Lokerson. 479-481 [doi]
- SIRENE, a system for speech training of deaf peopleMarie-Christine Haton, Jean-Paul Haton. 482-485 [doi]
- New algorithms for fast convolution based on convolution preserving spline signalsDietmar Achilles. 486-489 [doi]
- Hardware implementation of convolution using number theoretic transformsA. Baraniecka, Graham A. Jullien. 490-493 [doi]
- Computation of Fourier integral using polynomial interpolationL. P. Bolgiano, K. L. Kabir. 494-497 [doi]
- Round-off error propagation in Durbin's, Levinson's, and Trench's algorithmsGeorge Cybenko. 498-501 [doi]
- Resolution of superposed signals with envelope-constrained filtersBengt Mandersson. 502-505 [doi]
- On complexity of fast convolution algorithmsH. Gethöffer. 506-509 [doi]
- New polynomial transform algorithms for fast DFT computationHenri J. Nussbaumer, Philippe Quandalle. 510-513 [doi]
- Parallelism in the computation of the FFT and the WFTAHamid Nawab, James H. McClellan. 514-517 [doi]
- Complex rectangular transformsV. Umapathi Reddy, N. Sridhar Reddy. 518-521 [doi]
- Multidimensional DFT processing in subspaces whose dimensions are relatively primeDouglas F. Elliott, D. A. Orton. 522-525 [doi]
- A novel approach for implementing pitch prediction in sub-band codingRonald E. Crochiere. 526-529 [doi]
- Sub-band coder design incorporating quadrature filters and pitch predictionArthur Jay Barabell, Ronald E. Crochiere. 530-533 [doi]
- Speech coding based on a composite - Gaussian source modelV. Ramamoorthy, Thomas Ericson. 534-537 [doi]
- An efficient coding of the prediction residualLegand L. Burge, Rao Yarlagadda. 538-541 [doi]
- An improved residual encoder for speech compressionChi S. Chang. 542-545 [doi]
- Description of a hybrid 7.2 kbps vocoderHarold E. Watkins. 546-549 [doi]
- A new configuration for speech digitization at 9600 bits per secondDavid L. Cohn, James L. Melsa. 550-553 [doi]
- Development of a 4.8-9.6 kbps RELP vocoderMark Dankberg, David Y. Wong. 554-557 [doi]
- Voice-excited LPC coders for 9.6 kbps speech transmissionR. Viswanathan, William Russell, John Makhoul. 558-561 [doi]
- 8 kbps voice transmission by SPACJouji Suzuki, Kiyosumi Yoshiya. 562-565 [doi]
- Frequency warping for nonuniform talker normalizationHiroshi Matsumoto, Hisashi Wakita. 566-569 [doi]
- Order dependence in templates for monosyllabic word identificationSteven B. Davis. 570-573 [doi]
- Speaker independent recognition of isolated words using clustering techniquesLawrence R. Rabiner, Stephen E. Levinson, Aaron E. Rosenberg, Jay G. Wilpon. 574-577 [doi]
- Considerations in applying clustering techniques to speaker independent word recognitionLawrence R. Rabiner, Jay G. Wilpon. 578-581 [doi]
- A voice-controlled mechanical arm for immobilized patientsJohn M. Campbell, George N. Saridis. 582-585 [doi]
- Syntax and semantics in a word-sequence recognition systemSilvano Rivoira, Pietro Torasso. 586-590 [doi]
- A redundant hash addressing method adapted for the postprocessing and error-correction of computer recognized speechErkki Reuhkala, Matti Jalanko, Teuvo Kohonen. 591-594 [doi]
- Speech characterization from a rough spectral analysisJean-Sylvain Liénard. 595-598 [doi]
- Word spotting in conversational speechMark F. Medress, Marcia A. Derr, Timothy Diller, Dean R. Kloker, Larry L. Lutton, Henry N. Oredson, John F. Siebenand, Toby E. Skinner. 599-602 [doi]
- Adaptive linear prediction filtering for airborne underwater acoustic signal processorsAnthony W. Robertson. 603-607 [doi]
- A sonar target recognition experimentPaul C. Chestnut, Helen Landsman. 608-611 [doi]
- A broadband echo ranging system for measuring the frequency characteristics of fish schoolsD. E. Nelson, R. A. Johnson. 612-615 [doi]
- Experiments with a large aperture parametric acoustic receiving arrayC. Richard Reeves, Tommy G. Goldsberry, David F. Rohde. 616-619 [doi]
- A passive detection system for a wide class of illuminator signalsG. Retzer. 620-623 [doi]
- Filter design for estimating human blood-flow velocityJames Griffith, James Peterson. 624-627 [doi]
- A phase plane method for the analysis of seismic signalsDriss Aboutajdine, Zine El Abidine Amri, Mohamed Najim, Jack-Gérard Postaire. 628-631 [doi]
- Phase in speech and picturesAlan V. Oppenheim, Jae S. Lim, Gary E. Kopec, Stephen C. Pohlig. 632-637 [doi]
- Two-dimensional signal filters under modeling uncertaintiesSaleem A. Kassam, Tong Leong Lim, Leonard J. Cimini Jr.. 638-641 [doi]
- Multiple model recursive estimation of imagesVinay K. Ingle, John W. Woods. 642-645 [doi]
- On state-space signal processing with application to image enhancementC.-H. Chen. 646-649 [doi]
- Image enhancement by stochastic homomorphic filteringRobert W. Fries, James W. Modestino. 650-655 [doi]
- A two-dimensional filter design for isotropic reconstruction of track type airborne geophysical surveysPaul E. Anuta, Clare D. McGillem. 656-660 [doi]
- Fast least-squares phase estimation in speckle imagingRichard L. Frost, Craig K. Rushforth, Brent S. Baxter. 661-664 [doi]
- Frequency and bearing estimation by two-dimensional linear predictionLeland B. Jackson, H. C. Chien. 665-668 [doi]
- A two-dimensional maximum entropy spectral estimatorSalim E. Roucos, Donald G. Childers. 669-672 [doi]
- High-resolution two-dimensional spectral analysisOtis L. Frost, Thomas M. Sullivan. 673-676 [doi]
- A separable 2-D autoregressive spectral estimation algorithmLawrence S. Joyce. 677-680 [doi]
- Recursive digital filters with low coefficient sensitivityMichael T. McCallig, Richard R. Kurth, R. C. Steel. 681-683 [doi]
- A high-level block-diagram signal processing languageGary E. Kopec. 684-687 [doi]
- A design and computing system for signal processing applicationsH. Gethöffer, K. Hoffmann, A. Lenzer, Nicholas Roethe, H. Waldschmidt. 688-691 [doi]
- Design of real-time signal processing software for efficient use of high-speed array processors in multitasking environmentsRobert P. Dutton. 692-697 [doi]
- Efficient implementation of one and two dimensional digital signal processing algorithms on a multi-processor architectureThomas P. Barnwell III, S. Gaglio, C. J. M. Hodges. 698-701 [doi]
- A programmable signal processing architectureJ. M. Glass. 702-705 [doi]
- A real time analysis/Display system for nonstationary coastal processesL. A. Gerhardt, et al.. 706-709 [doi]
- Digital processing of Seasat SAR dataIan G. Cumming, John R. Bennett. 710-718 [doi]
- Optimal estimation and speech analysisJerry D. Gibson, Andrew C. Goris. 719-722 [doi]
- Sequential gradient estimation predictor for speech signalsCumhur Cengiz Evci, Raymond Steele, Costas S. Xydeas. 723-726 [doi]
- Reflection coefficient estimates based on a Markov chain modelBradley W. Dickinson, John M. Turner. 727-730 [doi]
- LPC analysis using a variable acoustic tube modelPanos Papamichalis, Thomas P. Barnwell III. 731-734 [doi]
- Reduction of computation in pole-zero modeling of speech signalsDacfey Dzung. 735-738 [doi]
- Statistical properties of an LPC distance measureJosé M. Tribolet, Lawrence R. Rabiner, Man Mohan Sondhi. 739-743 [doi]
- A distance measure based on the derivative of linear prediction phase spectrumB. Yegnanarayana, D. Raj Reddy. 744-747 [doi]
- Rate/Pitch modification of speech using the constant-Q transformJames E. Youngberg. 748-751 [doi]
- Features for the identification of mixed excitation in speech analysisLeah J. Siegel. 752-755 [doi]
- Automatic thresholding for voicing detection algorithmsEdward P. Neuburg. 756-758 [doi]
- Rank-order speech classification algorithm (RASCAL)Benjamin V. Cox, Lamar K. Timothy. 759-763 [doi]
- Multichannel zero-crossing-interval pitch estimationDavid Friedman. 764-767 [doi]
- Pitch extraction using MOS-LSI circuitryM. Dalrymple, D. Senderowicz, Robert W. Brodersen. 768-772 [doi]
- Time-domain pitch period extraction of speech signals using three nonlinear digital filtersWolfgang J. Hess. 773-776 [doi]
- A pattern recognition approach to compare natural and synthesized speechS. Sheshadri, M. B. Waldron. 777-780 [doi]
- Some factors influencing the performances of a speaker recognition system based on LPCGian Antonio Mian. 781-784 [doi]
- Accuracy of speaker verification via orthogonal parameters for noisy speechMalayappan Shridhar, M. Baraniecki. 785-788 [doi]
- Structure and performance of an on-line speaker verification systemUlrich Höfker, Peter Jesorsky, Bernhard Kriener, Maati Talmi, Dieter Wesseling. 789-792 [doi]
- Toward the development of practical methods of evaluating speaker recognizabilityWilliam D. Voiers. 793-796 [doi]
- Multistage decision schemes for speaker recognitionH. M. Dante, V. V. S. Sarma, G. R. Dattatreya. 797-800 [doi]
- New transformed variables for designing recursive digital filtersTapio Saramäki, Yrjö Neuvo. 801-804 [doi]
- On the design of complementary filtersRobert A. Gabel. 805-808 [doi]
- A method for the design of phase equalizersHoracio G. Martinez. 809-812 [doi]
- Design of stable all-pass filtersDavid B. Harris. 813-817 [doi]
- Statistical design of ARMA filtersLouis L. Scharf, James C. Luby. 818-821 [doi]
- Linear prediction in the design of Hilbert transformersWilliam R. Bauer. 822-823 [doi]
- Optimal design of digital Hilbert transformers with a concavity constraintKenneth Steiglitz. 824-827 [doi]
- A maximally flat filter design algorithm for quadrature mirror filters (QMF)L. E. Bergeron. 828-831 [doi]
- Application of transposition to decimation and interpolation in digital signal processing systemsTheo A. C. M. Claasen, Wolfgang F. G. Mecklenbräuker. 832-835 [doi]
- Anti-alias filters for tuner applicationsFrank Cornett, Elwood L. Seifert. 836-839 [doi]
- On the design of recursive lowpass digital filtersKishan Shenoi, B. P. Agrawal. 840-843 [doi]
- On the stability of a 1-bit-quantized feedback systemMartijn H. H. Hofelt. 844-848 [doi]
- FIR filter hardware reduction with adaptive delta-modulationWilliam F. Lawrence, Robert W. Newcomb. 849-852 [doi]
- Correlation detection of an FTH - signal using a CCD/CZT implementationSultan Mahmood, Rodger E. Ziemer. 853-856 [doi]
- Optimization of digital signal processors using array processor and CCD technologyJames K. Beard. 857-858 [doi]
- A high data rate, low power all-digital correlation circuit designK. Wayne Current, Douglas A. Mow, S. Youssef-Digaleh. 859-862 [doi]
- New concept of digital multi-frequency receiverAkira Ichikawa, Kazuo Nakata. 863-867 [doi]
- A parallel processor for real-time speech signal processingA. V. Ashajayanthi, S. Rajaram, N. Viswanadham. 868-871 [doi]
- Microprocessor based real-time speech processorBeat Pfister, S. Horvath Jr.. 872-875 [doi]
- A microprocessor-based digital filter laboratory station for an undergraduate signal processing laboratoryY. Gal, J. A. Howard, Sanjit K. Mitra. 876-879 [doi]
- Real-time text processing for Italian speech synthesisEnrico Vivalda, Stefano Sandri, Claudio Miotti. 880-883 [doi]
- Eight-channel digital speech synthesizer based on LPC techniquesLuciano Nebbia, Paolo Lucchini. 884-886 [doi]
- A microprocessor based audio response systemMarco Mezzalama, Angelo Serra. 887-890 [doi]
- Diphone synthesis for phonetic vocodingRichard M. Schwartz, John W. Klovstad, John Makhoul, Dennis H. Klatt, Victor Zue. 891-894 [doi]
- Time domain speech synthesis-by-rules using a flexible and fast signal management systemXavier Rodet, Jean-Luc Delatre. 895-898 [doi]
- The MISS speech synthesis systemArvin Levine, William R. Sanders. 899-902 [doi]
- An LPC k-parameter software speech synthesizer via dynamic microprogramming a general purpose computerL. Robert Morris, David L. Allan. 903-906 [doi]
- A fast FORTRAN implementation of the U. S. naval research laboratory algorithm for automatic translation of english text to VOTRAX parametersL. Robert Morris. 907-913 [doi]
- Analysis of formant and pitch information for Spanish phonemesGerardo Murillo, Fernando Berdichevsky, Christopher Cutler. 914-916 [doi]
- A microcomputer-based speech synthesizer which speaks SpanishFernando Berdichevsky, Gerardo Murillo, Christopher Cutler. 917-920 [doi]
- Speech synthesis from vocal tract area function acoustical measurementsBernard Tousignant, J.-P. Lefevre, Michel Lecours, J. C. Soumagne. 921-924 [doi]
- Adaptive structures for multiple-input noise cancelling applicationsLloyd L. Griffiths. 925-928 [doi]
- Adaptive noise cancelling of a sinusoidal interference using a lattice structureEdgar H. Satorius, J. D. Smith, P. M. Reeves. 929-932 [doi]
- γ-LMS and its use in a noise-compensating adaptive spectral analysis techniqueJohn R. Treichler. 933-936 [doi]
- Noise cancellation via linear prediction filteringEdgar H. Satorius, James R. Zeidler, S. Thomas Alexander. 937-940 [doi]
- On the "Desired behavior" of adaptive signal processing algorithmsDavid C. Farden, Justin Goding Jr., Khalid Sayood. 941-944 [doi]
- Detection of sinusoids by linear prediction filter frequency responseP. M. Reeves. 945-949 [doi]
- The spectral dynamics of evolving LMS adaptive filtersM. J. Shensa. 950-953 [doi]
- Filtering of narrowband signals using analytic sianal propertiesLouis F. Rocha. 954-957 [doi]
- A fixed-point iteration algorithm for adaptive linear estimation applied to spectral line enhancementStephen D. Huffman, Loren W. Nolte. 958-961 [doi]
- A multidimensional Modelling approach to texture classification and segmentationJean-Pierre Gambotto, Claude Guéguen. 962-966 [doi]
- Locating a passive source with array measurements a summary of resultsPeter M. Schultheiss. 967-970 [doi]
- A preprocessing filter for enhancing LPC analysis/Synthesis of noisy speechMarvin R. Sambur. 971-974 [doi]
- A 4800 bps voice excited predictive coder (VEPC) based on improved baseband/Sub-bands filtersDaniel Esteban, Claude Galand, Daniel Mauduit, Jean E. Menez. 975-979 [doi]
- Optimum quantizer algorithm for real-time block quantizingJean E. Menez, Fernand Boéri, Daniel Esteban. 980-984 [doi]
- Real time signal processing software for multiplierless microprocessorsDaniel Esteban, Daniel Mauduit, O. Maurel. 985-989 [doi]
- Evaluation of two narrowband speech algorithmsDavid P. Kemp, Jesse W. Fussell. 990-994 [doi]