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Semantic ScholarICASSP 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 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 - G. M. Borsuk, M. 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 Yarlagadda:
An efficient coding of the prediction residual. 538-541 - Chi S. Chang:
An improved residual encoder for speech compression. 542-545