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ICASSP 1979: Washington, D. C., USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '79, Washington, D. C., USA, April 2-4, 1979. IEEE 1979
2-D Digital Signal Processing I
- Olivier D. Faugeras, Jean-François Abramatic:
2-D FIR filter design from independent "Small" generating kernels using a mean square and Tchebyshev error criterion. 1-4 - Theresa C. Speake, Russell M. Mersereau:
A comparison of different window formulations for two-dimensional FIR filter design. 5-8 - Hyokang Chang, Jake K. Aggarwal:
Stabilization of two-dimensional recursive filters. 9-12 - C. H. Reddy, P. Karivaratharajan, M. N. S. Swamy, Venkatanarayana Ramachandran:
Generation of two-dimensional digital functions without non-essential singularities of the second kind. 13-19 - Richard E. Twogood, Michael P. Ekstrom:
Why filter recursively in two dimensions? 20-23 - Jean-François Abramatic, François Germain, Emmanuel Rosencher:
Design of 2-D recursive filters with separable denominator transfer functions. 24-27 - Gary A. Shaw, Russell M. Mersereau:
Space-domain design of two-dimensional recursive digital filters. 28-31 - Thomas L. Marzetta:
The design of 2-D recursive filters in the 2-D reflection coefficient domain. 32-35 - Giovanni Garibotto:
Two-dimensional half-plane recursive filter design. 36-39 - P. A. Ramamoorthy, Len T. Bruton:
Design of stable symmetric and non-symmetric half-plane digital recursive filters. 40-43
Narrowband Speech Communication
- Bishnu S. Atal, Nancy David:
On synthesizing natural-sounding speech by linear prediction. 44-47 - Sassan Ahmadi, Anthony G. Constantinides:
Linear prediction of formants for low bit rate digital speech transmission. 48-51 - Andres Buzo, Augustine H. Gray Jr., Robert M. Gray, John D. Markel:
A two-step speech compression system with vector quantizing. 52-55 - Thomas F. Quatieri:
A mixed-phase homomorphic vocoder. 56-59 - E. Blackman, R. Viswanathan, William Russell, John Makhoul:
Narrowband LPC speech transmission over noisy channels. 60-63 - Douglas B. Paul:
A robust vocoder with pitch-adaptive spectral envelope estimation and an integrated maximum-likelihood pitch estimator. 64-68 - Akira Kurematsu, Hikoichi Ishigami, Seishi Kitayama, Fumihiro Yato, Junso Tamura:
A linear predictive vocoder with new pitch extraction and exciting source. 69-72 - Arild Lacroix, Bela Makai:
A novel vocoder concept based on discrete time acoustic tubes. 73-76 - M. R. Ashouri:
Linear prediction of transformed speech. 77-80 - José M. Tribolet, Ronald E. Crochiere:
An analysis/Synthesis framework for transform coding of speech. 81-84
Automatic Phoneme Recognition
- Piero Demichelis, Renato De Mori, Pietro Laface, Mary O'Kane:
Computer recognition of stop consonants. 85-88 - Barry P. Kimberley, Campbell L. Searle:
Automatic discrimination of fricative consonants based on human audition. 89-92 - T. V. Sreenivas, T. V. Ananthapadmanabha:
On sensitivity of vocal tract area functions. 93-96 - Teuvo Kohonen, Gábor Németh, Kalle-J. Bry, Matti Jalanko, Heikki Riittinen:
Spectral classification of phonemes by learning subspaces. 97-100 - Marc Baudry, Benoit Dupeyrat:
Speech segmentation and recognition using syntactic methods on the direct signal. 101-104 - Jean Paul Haton, Claude Sanchez:
An experimental system for acoustic-phonetic decoding of continuous speech. 105-107 - Seppo Haltsonen, Kalle-J. Bry:
Automatic selection of phonemes from an equally spaced quasi-phoneme string by the entropy principle. 108-111 - Donald C. Lokerson:
A unique real-time speech decoder that operates from new perspectives. 112-115 - Victor W. Zue, Ronald A. Cole:
Experiments on spectrogram reading. 116-119
Underwater Signal Processing
- Charles R. Baker:
Testing sonar data for multivariate normality. 120-123 - Joseph C. Hassab, Ronald E. Boucher:
A quantitative study of optimum and sub-optimum filters in the generalized correlator. 124-127 - Yiu Tong Chan, J. M. Riley, J. B. Plant:
A parameter estimation approach to time delay estimation. 128-131 - A. R. Pratt:
Local estimation of delay parameter following robust detection. 132-135 - Terry Rickard, Mauro J. Dentino, James R. Zeidler:
Detection performance of an adaptive processor in non-stationary noise. 136-139 - Roger F. Dwyer:
Robust sequential detection of narrowband acoustic signals in noise. 140-143 - Roberto Berezdivin, Robert Perl, Robert Braunstein:
A phase-coherence detector/Estimator. 144-147
Spectral Analysis
- Steven M. Kay, Larry Marple:
Sources of and remedies for spectral line splitting in autoregressive spectrum analysis. 151-154 - W. Brandenburg:
Spectral analysis using prediction methods. 155-158 - Larry Marple:
Spectral line analysis by Pisarenko and Prony methods. 159-161 - Steven M. Kay:
Fourier-autoregressive spectral estimation. 162-165 - Carey Gibson, Simon Haykin, Stanislav B. Kesler:
Maximum entropy (adaptive) filtering applied to radar clutter. 166-169 - R. T. Schaefer, Ronald W. Schafer, Russell M. Mersereau:
Digital signal processing for doppler radar signals. 170-173 - Ulrich Steimel:
Fast estimation of narrowband spectra. 174-177 - R. J. Linggard, B. D. V. Smith:
A family of phase complementary filters. 178-181 - J. Ziegenbein:
Spectral analysis using the Karhunen-Loeve transform. 182-185 - Michael R. Portnoff:
Magnitude-phase relationships for short-time Fourier transforms based on Gaussian analysis windows. 186-189 - Dean P. Kolba, Thomas W. Parks:
Extrapolation and spectral estimation for bandlimited, time-concentrated signals. 190-193 - Rui J. P. de Figueiredo:
Optimal estimation of essentially and strictly bandlimited signals and their spectrum by generalized splines. 194-199
Noise Reduction in Speech Processing
- Steven F. Boll:
A spectral subtraction algorithm for suppression of acoustic noise in speech. 200-203 - Dennis Pulsipher, Steven F. Boll, Craig K. Rushforth, LaMar Timothy:
Reduction of nonstationary acoustic noise in speech using LMS adaptive noise cancelling. 204-207 - Michael G. Berouti, Richard M. Schwartz, John Makhoul:
Enhancement of speech corrupted by acoustic noise. 208-211 - Robert D. Preuss:
A frequency domain noise cancelling preprocessor for narrowband speech communications systems. 212-215 - Charles F. Teacher, David C. Coulter:
Performance of LPC vocoders in a noisy environment. 216-219 - Donald P. Fulghum, J. E. Gunn III:
LPC voice digitizer with background noise suppression. 220-223 - Bruce R. Musicus, Jae S. Lim:
Maximum likelihood parameter estimation of noisy data. 224-227 - William J. Done, Craig K. Rushforth:
Estimating the parameters of a noisy all-pole process using pole-zero modeling. 228-231 - Gerald M. Borsuk, Marvin H. White:
CCD adaptive filtering for robust LPC speech processing. 232-234
Automatic Recognition of Continuous Speech
- Johannes Jaschul:
An approach to speaker normalization in an automatic speech recognition system. 235-238 - Stephen E. Levinson, Aaron E. Rosenberg:
A new system for continuous speech recognition - preliminary results. 239-244 - Jean Paul Haton, Olivier Morel:
Automatic recognition of continuous digits sequences by means of segmentation and dynamic programming. 245-248 - Lalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer:
Recognition results for several experimental acoustic processors. 249-251 - Renato De Mori, Leonardo Lesmo, Marisa Poncini:
The structure of a lexicon for a speech understanding system. 252-255 - Timothy Diller:
Phonetic word verification. 256-261 - Janet M. Baker:
Performance statistics of the HEAR acoustic processor. 262-265 - Rui J. P. de Figueiredo, Thomas J. Brzustowicz:
Techniques for recognition of spectrogram patterns based on dynamic modeling. 266-268 - Joseph-Jean Mariani, Jean-Sylvain Liénard, G. Renard:
Speech recognition in the context of two-way immediate person-machine interaction. 269-272
Underwater Arrays and Medium Effects
- Bernard Widrow:
A review of adaptive antennas. 273-278 - Louis A. Mole, Frank A. Andrews:
An array optimization technique. 279-281 - William S. Hodgkiss:
Adaptive array processing: Time vs. frequency domain. 282-285 - Russell P. Kraft, John F. McDonald, F. Ahlgren:
Minimax optimization of two-dimensional focused nonuniformly spaced arrays. 286-289 - Azizul H. Quazi, Albert H. Nuttall:
Effects of random shading, phasing errors and element failures on the beampatterns of line and planar arrays. 290-293 - Kenneth A. Faucher, James J. Foster:
A computer model for the analysis of source motion, the ocean environment, and interference effects on acoustic signal coherence. 294-297 - Kenneth E. Hawker, Jack A. Shooter:
The roles of integration time and acoustic multipaths in determining the structure of CW line spectra. 298-301 - Albert A. Gerlach:
Impact of the ocean acoustic transfer function on the coherence of undersea propagations. 302-305 - Georges Bienvenu:
Influence of the spatial coherence of the background noise on high resolution passive methods. 306-309
Audio and Acoustical Systems
- Douglas Preis:
Audio signal processing with transversal filters. 310-313 - James M. Kates:
Constant-Q analysis using the chirp z-transform. 314-317 - F. C. Pirz:
Design of a wideband, constant beamwidth, array microphone for use in the near-field. 318-321 - W. Marshall Leach Jr., Ronald W. Schafer, Thomas P. Barnwell III:
Time domain measurement of loudspeaker driver parameters. 322-325 - Oscar J. Bonello:
A new computer aided method for the complete acoustical design of broadcasting and recording studios. 326-329 - J. Robert Ashley:
Auditory backward inhibition can ruin a concert hall. 330-334
Roundoff Noise and Coefficent Sensitivity in Digital Filters
- David S. K. Chan:
Constrained minimization of roundoff noise in fixed-point digital filters. 335-339 - William L. Mills, Clifford T. Mullis, Richard A. Roberts:
Normal realizations of IIR digital filters. 340-343 - Masud Arjmand, Richard A. Roberts:
Reduced multiplier, low roundoff noise digital filters. 344-346 - Amar M. Ali:
Linear transformations for the design of digital and active filters. 347-350 - James D. Ledbetter, Rao K. Yarlagadda:
Coefficient quantization effects on pole locations for state model digital filters. 351-354 - Tatsuo Higuchi, Hiroski Takeo:
A state-space approach for elimination of limit cycles in digital filters with arbitrary structures. 355-358 - Augustine H. Gray Jr.:
Passive cascaded lattice digital filters. 359-362 - Rolf Block, Arild Lacroix:
Simplified error models for digital filters. 363-366 - David C. Munson Jr., Bede Liu:
Narrowband recursive filters with error spectrum shaping. 367-370 - Tor A. Ramstad:
Some considerations on coefficient sensitivity and noise in direct form IIR interpolators and decimators. 371-374 - Allen Gersho, B. Gopinath, Andrew M. Odlyzko:
Coefficient inaccuracy in FIR filters. 375-377 - Victor B. Lawrence, Andres C. Salazar:
Effects of finite coefficient precision on FIR filter spectra. 378-379 - David G. Messerschmitt:
Accumulation of distortion in signal processing systems. 380-383
System Identification and Modeling
- J. A. Ponnusamy, Mandyam D. Srinath, Periagaram K. Rajasekaran:
Identification of complex autoregressive processes. 384-387 - Charlton M. Walter:
Geometrical characterization of the canonical coordinate basis underlying a family of error minimizing signal compression techniques. 388-391 - James A. Cadzow:
Inversion of signal operations. 392-397 - Albert Arcese:
The solution of discrete convolutions with a bounded error constraint. 398-400 - Mark A. Richards, Ronald W. Schafer, Russell M. Mersereau:
An experimental study of the effects of noise on a class of iterative deconvolution algorithms. 401-404 - Gervasio Prado:
System identification using a maximum-likelihood spectral matching technique. 405-408 - Jae S. Lim:
Spectral root homomorphic deconvolution system. 409-414 - William J. Done, Craig K. Rushforth:
Evaluation of the Steiglitz algorithm for estimating the parameters of an ARMA process. 415-418
Wideband Speech Communication
- Thomas Ericson, V. Ramamoorthy:
Modulo-PCM: A new source coding scheme. 419-422 - Debasis Mitra:
A generalized adaptive quantization system with a new reconstruction method, for noisy transmission. 423-427 - John Makhoul, Michael G. Berouti:
High-frequency regeneration in speech coding systems. 428-431 - Jean-Pierre Adoul, Sarto Morissette, Michel Rudko:
Bit-rate-halving algorithm for PCM-encoded speech using a new bidimensional data compression scheme. 432-435 - Yohtaro Yatsuzuka:
A high-gain DSI-ADPCM system. 436-441 - James D. Johnston, David J. Goodman:
Digital transmission of commentary-grade (7 kHz) audio at 56 or 64 kb/s. 442-444 - John J. Dubnowski, Ronald E. Crochiere:
Variable rate coding. 445-448
Speech Quality Evaluation and Enhancement
- John D. Markel, Steven B. Davis, Ted H. Applebaum:
A methodology for studying telephone amplitude distortion effects on narrowband speech processors. 449-452 - Bishnu S. Atal, Manfred R. Schroeder:
Optimizing predictive coders for minimum audible noise. 453-455 - Caldwell P. Smith:
Talker variance and phonetic feature variance in diagnostic intelligibility scores for digital voice communications processors. 456-459 - Louis C. W. Pols:
Intelligibility of intervocalic consonants in noise. 460-463 - Craig R. Allen:
Optimum linear filter for speech communication. 464-466 - Mamoru Nakatsui:
Subjective evaluation of SPAC in improving the quality of noisy speech. 467-470
Speech Aids for the Handicapped
- Douglas C. Sargent, Andrew Malcolm:
The presentation of continuous speech with synchronous printed text. 471-474 - G. L. Bull, Michael M. E. Johns, W. E. McDonald, R. C. Bralley:
The effects of Teflon injection on laryngeal dynamics. 475-478 - Donald C. Lokerson:
A conceptually unique speech training aid system. 479-481 - Marie-Christine Haton, Jean-Paul Haton:
SIRENE, a system for speech training of deaf people. 482-485
Transforms and Algorithms
- Dietmar Achilles:
New algorithms for fast convolution based on convolution preserving spline signals. 486-489 - A. Baraniecka, Graham A. Jullien:
Hardware implementation of convolution using number theoretic transforms. 490-493 - L. P. Bolgiano, K. L. Kabir:
Computation of Fourier integral using polynomial interpolation. 494-497 - George Cybenko:
Round-off error propagation in Durbin's, Levinson's, and Trench's algorithms. 498-501 - Bengt Mandersson:
Resolution of superposed signals with envelope-constrained filters. 502-505 - H. Gethöffer:
On complexity of fast convolution algorithms. 506-509 - Henri J. Nussbaumer, Philippe Quandalle:
New polynomial transform algorithms for fast DFT computation. 510-513 - Hamid Nawab, James H. McClellan:
Parallelism in the computation of the FFT and the WFTA. 514-517 - V. Umapathi Reddy, N. Sridhar Reddy:
Complex rectangular transforms. 518-521 - Douglas F. Elliott, D. A. Orton:
Multidimensional DFT processing in subspaces whose dimensions are relatively prime. 522-525
Mediumband Speech Communication
- Ronald E. Crochiere:
A novel approach for implementing pitch prediction in sub-band coding. 526-529 - Arthur Jay Barabell, Ronald E. Crochiere:
Sub-band coder design incorporating quadrature filters and pitch prediction. 530-533 - V. Ramamoorthy, Thomas Ericson:
Speech coding based on a composite - Gaussian source model. 534-537 - Legand L. Burge, Rao K. Yarlagadda:
An efficient coding of the prediction residual. 538-541 - Chi S. Chang:
An improved residual encoder for speech compression. 542-545 - Harold E. Watkins:
Description of a hybrid 7.2 kbps vocoder. 546-549 - David L. Cohn, James L. Melsa:
A new configuration for speech digitization at 9600 bits per second. 550-553 - Mark Dankberg, David Y. Wong:
Development of a 4.8-9.6 kbps RELP vocoder. 554-557 - R. Viswanathan, William Russell, John Makhoul:
Voice-excited LPC coders for 9.6 kbps speech transmission. 558-561 - Jouji Suzuki, Kiyosumi Yoshiya:
8 kbps voice transmission by SPAC. 562-565
Discrete Word Recognition
- Hiroshi Matsumoto, Hisashi Wakita:
Frequency warping for nonuniform talker normalization. 566-569 - Steven B. Davis:
Order dependence in templates for monosyllabic word identification. 570-573 - Lawrence R. Rabiner, Stephen E. Levinson, Aaron E. Rosenberg, Jay G. Wilpon:
Speaker independent recognition of isolated words using clustering techniques. 574-577 - Lawrence R. Rabiner, Jay G. Wilpon:
Considerations in applying clustering techniques to speaker independent word recognition. 578-581 - John M. Campbell, George N. Saridis:
A voice-controlled mechanical arm for immobilized patients. 582-585 - Silvano Rivoira, Pietro Torasso:
Syntax and semantics in a word-sequence recognition system. 586-590 - Erkki Reuhkala, Matti Jalanko, Teuvo Kohonen:
A redundant hash addressing method adapted for the postprocessing and error-correction of computer recognized speech. 591-594 - Jean-Sylvain Liénard:
Speech characterization from a rough spectral analysis. 595-598 - Mark F. Medress, Marcia A. Derr, Timothy Diller, Dean R. Kloker, Larry L. Lutton, Henry N. Oredson, John F. Siebenand, Toby E. Skinner:
Word spotting in conversational speech. 599-602
Systems and Processing
- Anthony W. Robertson:
Adaptive linear prediction filtering for airborne underwater acoustic signal processors. 603-607 - Paul C. Chestnut, Helen Landsman:
A sonar target recognition experiment. 608-611 - D. E. Nelson, R. A. Johnson:
A broadband echo ranging system for measuring the frequency characteristics of fish schools. 612-615 - C. Richard Reeves, Tommy G. Goldsberry, David F. Rohde:
Experiments with a large aperture parametric acoustic receiving array. 616-619 - G. Retzer:
A passive detection system for a wide class of illuminator signals. 620-623 - James Griffith, James Peterson:
Filter design for estimating human blood-flow velocity. 624-627 - Driss Aboutajdine, Zine El Abidine Amri, Mohamed Najim, Jack-Gérard Postaire:
A phase plane method for the analysis of seismic signals. 628-631
2-D Digital Signal Processing II
- Alan V. Oppenheim, Jae S. Lim, Gary E. Kopec, Stephen C. Pohlig:
Phase in speech and pictures. 632-637 - Saleem A. Kassam, Tong Leong Lim, Leonard J. Cimini Jr.:
Two-dimensional signal filters under modeling uncertainties. 638-641 - Vinay K. Ingle, John W. Woods:
Multiple model recursive estimation of images. 642-645 - Chi Hau Chen:
On state-space signal processing with application to image enhancement. 646-649 - Robert W. Fries, James W. Modestino:
Image enhancement by stochastic homomorphic filtering. 650-655 - Paul E. Anuta, Clare D. McGillem:
A two-dimensional filter design for isotropic reconstruction of track type airborne geophysical surveys. 656-660 - Richard L. Frost, Craig K. Rushforth, Brent S. Baxter:
Fast least-squares phase estimation in speckle imaging. 661-664 - Leland B. Jackson, H. C. Chien:
Frequency and bearing estimation by two-dimensional linear prediction. 665-668 - Salim E. Roucos, Donald G. Childers:
A two-dimensional maximum entropy spectral estimator. 669-672 - Otis L. Frost, Thomas M. Sullivan:
High-resolution two-dimensional spectral analysis. 673-676 - Lawrence S. Joyce:
A separable 2-D autoregressive spectral estimation algorithm. 677-680
Hardware-Software Structures
- Michael T. McCallig, Richard R. Kurth, R. C. Steel:
Recursive digital filters with low coefficient sensitivity. 681-683 - Gary E. Kopec:
A high-level block-diagram signal processing language. 684-687 - H. Gethöffer, K. Hoffmann, A. Lenzer, Nicholas Roethe, H. Waldschmidt:
A design and computing system for signal processing applications. 688-691 - Robert P. Dutton:
Design of real-time signal processing software for efficient use of high-speed array processors in multitasking environments. 692-697 - Thomas P. Barnwell III, S. Gaglio, C. J. M. Hodges:
Efficient implementation of one and two dimensional digital signal processing algorithms on a multi-processor architecture. 698-701 - J. M. Glass:
A programmable signal processing architecture. 702-705 - Lester A. Gerhardt:
A real time analysis/Display system for nonstationary coastal processes. 706-709 - Ian G. Cumming, John R. Bennett:
Digital processing of Seasat SAR data. 710-718
Speech Analysis
- Jerry D. Gibson, Andrew C. Goris:
Optimal estimation and speech analysis. 719-722 - Cumhur Cengiz Evci, Raymond Steele, Costas S. Xydeas:
Sequential gradient estimation predictor for speech signals. 723-726 - Bradley W. Dickinson, John M. Turner:
Reflection coefficient estimates based on a Markov chain model. 727-730 - Panos Papamichalis, Thomas P. Barnwell III:
LPC analysis using a variable acoustic tube model. 731-734 - Dacfey Dzung:
Reduction of computation in pole-zero modeling of speech signals. 735-738 - José M. Tribolet, Lawrence R. Rabiner, Man Mohan Sondhi:
Statistical properties of an LPC distance measure. 739-743 - B. Yegnanarayana, Raj Reddy:
A distance measure based on the derivative of linear prediction phase spectrum. 744-747 - James E. Youngberg:
Rate/Pitch modification of speech using the constant-Q transform. 748-751
Pitch Detection
- Leah J. Siegel:
Features for the identification of mixed excitation in speech analysis. 752-755 - Edward P. Neuburg:
Automatic thresholding for voicing detection algorithms. 756-758 - Benjamin V. Cox, Lamar K. Timothy:
Rank-order speech classification algorithm (RASCAL). 759-763 - David H. Friedman:
Multichannel zero-crossing-interval pitch estimation. 764-767 - M. Dalrymple, D. Senderowicz, Robert W. Brodersen:
Pitch extraction using MOS-LSI circuitry. 768-772 - Wolfgang J. Hess:
Time-domain pitch period extraction of speech signals using three nonlinear digital filters. 773-776 - S. Sheshadri, Manjula B. Waldron:
A pattern recognition approach to compare natural and synthesized speech. 777-780
Talker Verification/Recognition
- Gian Antonio Mian:
Some factors influencing the performances of a speaker recognition system based on LPC. 781-784 - Malayappan Shridhar, M. R. Baraniecki:
Accuracy of speaker verification via orthogonal parameters for noisy speech. 785-788 - Ulrich Höfker, Peter Jesorsky, Bernhard Kriener, Maati Talmi, Dieter Wesseling:
Structure and performance of an on-line speaker verification system. 789-792 - William D. Voiers:
Toward the development of practical methods of evaluating speaker recognizability. 793-796 - H. M. Dante, V. V. S. Sarma, G. R. Dattatreya:
Multistage decision schemes for speaker recognition. 797-800
Digital Filter Design
- Tapio Saramäki, Yrjö Neuvo:
New transformed variables for designing recursive digital filters. 801-804 - Robert A. Gabel:
On the design of complementary filters. 805-808 - Horacio G. Martinez:
A method for the design of phase equalizers. 809-812 - David B. Harris:
Design of stable all-pass filters. 813-817 - Louis L. Scharf, James C. Luby:
Statistical design of ARMA filters. 818-821 - William R. Bauer:
Linear prediction in the design of Hilbert transformers. 822-823 - Kenneth Steiglitz:
Optimal design of digital Hilbert transformers with a concavity constraint. 824-827 - L. E. Bergeron:
A maximally flat filter design algorithm for quadrature mirror filters (QMF). 828-831 - Theo A. C. M. Claasen, Wolfgang F. G. Mecklenbräuker:
Application of transposition to decimation and interpolation in digital signal processing systems. 832-835 - Frank Cornett, Elwood L. Seifert:
Anti-alias filters for tuner applications. 836-839
Devices and Applications
- Kishan Shenoi, Bhagwati Prasad Agrawal:
On the design of recursive lowpass digital filters. 840-843 - Martijn H. H. Hofelt:
On the stability of a 1-bit-quantized feedback system. 844-848 - William F. Lawrence, Robert Newcomb:
FIR filter hardware reduction with adaptive delta-modulation. 849-852 - Sultan Mahmood, Rodger E. Ziemer:
Correlation detection of an FTH - signal using a CCD/CZT implementation. 853-856 - James K. Beard
:
Optimization of digital signal processors using array processor and CCD technology. 857-858 - K. Wayne Current, Douglas A. Mow, S. Youssef-Digaleh:
A high data rate, low power all-digital correlation circuit design. 859-862 - Akira Ichikawa, Kazuo Nakata:
New concept of digital multi-frequency receiver. 863-867 - A. V. Ashajayanthi, S. Rajaram, N. Viswanadham:
A parallel processor for real-time speech signal processing. 868-871 - Beat Pfister, S. Horvath Jr.:
Microprocessor based real-time speech processor. 872-875 - Y. Gal, J. A. Howard, Sanjit K. Mitra:
A microprocessor-based digital filter laboratory station for an undergraduate signal processing laboratory. 876-879
Speech Synthesis
- Enrico Vivalda, Stefano Sandri, Claudio Miotti:
Real-time text processing for Italian speech synthesis. 880-883 - Luciano Nebbia, Paolo Lucchini:
Eight-channel digital speech synthesizer based on LPC techniques. 884-886 - Marco Mezzalama, Angelo Serra:
A microprocessor based audio response system. 887-890 - Richard M. Schwartz, John W. Klovstad, John Makhoul, Dennis H. Klatt, Victor Zue:
Diphone synthesis for phonetic vocoding. 891-894 - Xavier Rodet, Jean-Luc Delatre:
Time domain speech synthesis-by-rules using a flexible and fast signal management system. 895-898 - Arvin Levine, William R. Sanders:
The MISS speech synthesis system. 899-902 - L. Robert Morris, David L. Allan:
An LPC k-parameter software speech synthesizer via dynamic microprogramming a general purpose computer. 903-906 - L. Robert Morris:
A fast FORTRAN implementation of the U. S. naval research laboratory algorithm for automatic translation of english text to VOTRAX parameters. 907-913 - Gerardo Murillo, Fernando Berdichevsky, Christopher Cutler:
Analysis of formant and pitch information for Spanish phonemes. 914-916 - Fernando Berdichevsky, Gerardo Murillo, Christopher Cutler:
A microcomputer-based speech synthesizer which speaks Spanish. 917-920 - Bernard Tousignant, J.-P. Lefevre, Michel Lecours, J. C. Soumagne:
Speech synthesis from vocal tract area function acoustical measurements. 921-924
Adaptive Filtering
- Lloyd J. Griffiths:
Adaptive structures for multiple-input noise cancelling applications. 925-928 - Edgar H. Satorius, J. D. Smith, P. M. Reeves:
Adaptive noise cancelling of a sinusoidal interference using a lattice structure. 929-932 - John R. Treichler:
γ-LMS and its use in a noise-compensating adaptive spectral analysis technique. 933-936 - Edgar H. Satorius, James R. Zeidler, S. Thomas Alexander:
Noise cancellation via linear prediction filtering. 937-940 - David C. Farden, Justin Goding Jr., Khalid Sayood:
On the "Desired behavior" of adaptive signal processing algorithms. 941-944 - P. M. Reeves:
Detection of sinusoids by linear prediction filter frequency response. 945-949 - M. J. Shensa:
The spectral dynamics of evolving LMS adaptive filters. 950-953 - Louis F. Rocha:
Filtering of narrowband signals using analytic sianal properties. 954-957 - Stephen D. Huffman, Loren W. Nolte:
A fixed-point iteration algorithm for adaptive linear estimation applied to spectral line enhancement. 958-961 - Jean-Pierre Gambotto, Claude Guéguen:
A multidimensional Modelling approach to texture classification and segmentation. 962-966 - Peter M. Schultheiss:
Locating a passive source with array measurements a summary of results. 967-970 - Marvin R. Sambur:
A preprocessing filter for enhancing LPC analysis/Synthesis of noisy speech. 971-974 - Daniel J. Esteban, Claude R. Galand, Daniel Mauduit, Jean E. Menez:
A 4800 bps voice excited predictive coder (VEPC) based on improved baseband/Sub-bands filters. 975-979 - Jean E. Menez, Fernand Boéri, Daniel J. Esteban:
Optimum quantizer algorithm for real-time block quantizing. 980-984 - Daniel J. Esteban, Daniel Mauduit, O. Maurel:
Real time signal processing software for multiplierless microprocessors. 985-989 - David P. Kemp, Jesse W. Fussell:
Evaluation of two narrowband speech algorithms. 990-994
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