default search action
ICASSP 1978: Tulsa, Oklahoma, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '78, Tulsa, Oklahoma, USA, April 10-12, 1978. IEEE 1978
Speech Analysis
- V. V. S. Sarma, D. Venugopal:
Studies on pattern recognition approach to voiced-unvoiced-silence classification. 1-4 - Edward P. Neuburg:
Improvement of voicing decisions by use of context. 5-7 - T. V. Ananthapadmanabha, B. Yegnanarayana:
Epoch extraction from linear prediction residual. 8-11 - Robert J. McAulay:
Maximum likelihood pitch estimation using state-variable techniques. 12-14 - Melvyn J. Hunt, John S. Bridle, John N. Holmes:
Interactive digital inverse filtering and its relation to linear prediction methods. 15-18 - Gen Ooyama, Sigeru Katagiri, Ken'iti Kido:
A new method of Cepstrum analysis by using comb lifter. 19-22 - Thomas W. Cairns, William A. Coberly, David F. Findley:
Arma modeling applied to linear prediction of speech. 23-26 - Thomas E. Carter:
Study of an adaptive lattice structure for linear prediction analysis of speech. 27-30 - M. Habib, D. Robinson, W. David Sincoskie:
Real-zeros in pitch detection. 31-34 - J. Bee Bednar, William A. Coberly:
Identification of nonstationary components by complex spectral zero analysis. 35-38 - John W. Woods, Vinay K. Ingle:
Two-dimensional processing of spectrogram data. 39-42 - John N. Holmes, Michael W. Judd, David H. Walesby:
A high-quality all-digital sound spectrograph developed for speech signal analysis. 43-46
Structures and Quantization
- Kalyan Mondal, Sanjit K. Mitra:
Realization of digital transfer functions using cascaded normalized two-pairs. 47-50 - Ernst Lueder, K. Haug:
Calculation of all equivalent and canonic 2nd order digital filter structures. 51-54 - Kalyan Mondal, S. Chakrabarti, Sanjit K. Mitra:
State-structures and minimal state-structures for arbitrary digital filters. 55-57 - M. Omair Ahmad, C. H. Reddy, Venkatanarayana Ramachandran, M. N. Shanmukha Swamy:
Realization of a class of two-dimensional analog ladders with applications to wave digital filters. 58-61 - Larry J. Fruit:
A figure of merit for digital filters. 62-66 - E. Cohler, James A. Storer, R. Borgioli:
Formats in audio bandwidth processing applications. 67-70 - William L. Mills, Clifford T. Mullis, Richard A. Roberts:
Digital filter realizations without overflow oscillations. 71-74 - Ahmad I. Abu-El-Haija, Allen M. Peterson:
An approach to eliminate roundoff errors in digital filters. 75-78 - K. M. M. Prabhu, J. P. Agrawal:
Selection of data windows for digital signal processing. 79-82
Adaptive Filtering
- John Makhoul, R. Viswanathan:
Adaptive lattice methods for linear prediction. 83-86 - Lloyd J. Griffiths:
An adaptive lattice structure for noise-cancelling applications. 87-90 - R. Jeffrey Keeler:
Non-optimal convergence of adaptive LMS with uncorrelated data. 91-95 - William S. Hodgkiss:
Selecting the length of an adaptive transversal filter. 96-99 - Dietrich Maiwald, Hans-Peter Kaeser, Felix Closs:
An adaptive equalizer with significantly reduced number of operations. 100-104 - Judith G. Claassen, Wiley E. Thompson:
Real-time signal processing for unbiased system identification. 105-108 - Norman L. Owsley:
Adaptive data orthogonalization. 109-112 - Alan Weiss, Debasis Mitra:
Some mathematical results on the effects on digital adaptive filters of implementation errors and noise. 113-117 - John R. Treichler, Michael G. Larimore, C. Richard Johnson Jr.:
Simple adaptive IIR filtering. 118-122 - Nasir Ahmed, Donald R. Hummels, Michael L. Uhl, David L. Soldan:
A short-term sequential regression algorithm. 123-126
Underwater Acoustic Sensors
- Francis Hugh Fenlon:
On the maximum achievable conversion efficiency on a parametric acoustic array. 127-129 - C. Richard Reeves, Voldi E. Maki, Tommy G. Goldsberry, David F. Rohde:
Vibration sensitivity of the parametric acoustic receiving array. 130-133 - John J. Cornyn:
Errors in array response calculations due to incorrectly folding the vertical acoustic arrival structure into beam pattern. 134-136 - Francis Hugh Fenlon, Geoffrey L. Wilson:
Control of the directional response of acoustic transducer array elements on curved surfaces by modification of the diffracted field. 137-140 - Melvin J. Hinich:
Array processing using the frequency-wavenumber approach. 141-142 - Charles R. Baker, Louis R. Chow:
Effect of sampling errors on array gain. 143-147 - Henry E. Lee:
Synthetic array processing for underwater mapping applications. 148-151 - William Barry, Darrell R. Jackson, Jim O. P. Schultz:
A flexible towed sonar for ocean acoustic measurements. 152-154
Speech Synthesis
- Naoki Ishii, Ken'ya Murakami:
An audio response method for CAI services. 155-158 - David J. Quarmby, D. H. Midgeley, J. Ruby:
Speech synthesis using real-time software. 159-162 - John Makhoul, R. Viswanathan, Richard M. Schwartz, A. W. F. Huggins:
A mixed-source model for speech compression and synthesis. 163-166 - Denis L. Baggi:
Implementation of a channel vocoder synthesizer using a fast, time-multiplexed digital filter. 167-170 - David Y. Wong, John D. Markel:
An excitation function for LPC synthesis which retains the human Glottal phase characteristics. 171-174 - Richard T. Gagnon:
Votrax real time hardware for phoneme synthesis of speech. 175-178 - M. Baumwolspiner:
Speech generation through waveform synthesis. 179-182
System Identification and Modeling
- Lawrence R. Rabiner, Ronald E. Crochiere, Jont B. Allen:
Comparisons of system identification methods in the presence of high noise levels and bandlimited inputs. 183-187 - S. Thomas Alexander, Edgar H. Satorius, James R. Zeidler:
Linear prediction and maximum entropy spectral analysis of finite bandwidth signals in noise. 188-191 - Russell M. Mersereau, Ronald W. Schafer:
Comparative study of iterative deconvolution algorithms. 192-195 - H. Joel Trussell, Bobby R. Hunt:
Image restoration of space variant blurs by sectioned methods. 196-198 - Subhash C. Kwatra, Vijay K. Jain:
Nonlinear filter for inversion of channel distortion. 199-202 - Leland B. Jackson, Frank K. Soong:
Observations on linear estimation. 203-207 - Gérard Alengrin, Gérard Favier:
New stochastic realization algorithms for identification of ARMA models. 208-213 - Jeff H. Derby:
Analysis and representation of composite signals by cepstral inverse filtering. 214-217 - R. J. Mitchell, R. C. Gonzalez:
Multilevel crossing rates for automated signal classification. 218-222
Sonar and Radar Systems
- Robert D. Strattan:
Target identification from radar signatures. 223-227 - Brian P. Holt, Ronald C. Houts:
Multiband FIR digital filter design algorithm for radar clutter suppression. 228-231 - Neal B. Lawrence, Jerry D. Moore:
Typical performance characteristics of a two-dimensional CFAR. 232-235 - O. L. Godwin, Vijay K. Jain:
A technique for pole-zero modeling of complex-valued autocorrelations. 236 - Charles R. Carter, Simon S. Haykin, Hing C. Chan:
A programmable sonar signal processor. 237-240 - Martin G. Fagot:
Reconnaissance sonar for deep ocean seamount detection. 241-244 - Duane C. Tate:
An at-sea system for the prediction of underwater sound propagation. 245-247 - Harold R. Hall:
Inherent errors in sonar range prediction. 248-251
Audio Amplifier Design, Distortion Perception and Measurement
- W. Marshall Leach Jr.:
Design considerations for feedback amplifiers. 252-254 - Margit Petri-Larmi, Matti Otala, Eero Leinonen, Jorma Lammasniemi:
Audibility of transient intermodulation distortion. 255-262 - John Curl:
Omitted factors in audio circuit design. 263-266 - Kenneth F. Leonard:
Computer analysis of transient distortion and low transient distortion amplifier design. 267-269 - A. Yonovitz, Barbara Jill Bickford, Joseph Lozar, Dianne R. Ferrell:
Electroacoustic distortions: Multidimensional analysis of hearing aid transduced speech and music. 270-274
Talker Recognition
- L. Fasolo, Gian Antonio Mian:
A comparison between two approaches to automatic speaker recognition. 275-278 - Peter Jesorsky, Ulrich Höfker, Maati Talmi:
Extraction of speaker-specific features from spoken code sentences. 279-282 - Larry Pfeifer:
New techniques for text-independent speaker identification. 283-286 - John D. Markel, Steven B. Davis:
Text-independent speaker identification from a large linguistically unconstrained time-spaced data base. 287-290 - Hiroshi Matsumoto, Tadamoto Nimura:
Text-independent speaker identification based on piecewise canonical discriminant analysis. 291-294 - H. M. Dante, V. V. S. Sarma:
Automatic speaker identification for a large number of speakers. 295-298
Wideband Speech Communication
- Aaron J. Goldberg:
2400/16, 000 Bps Multirate voice processor. 299-302 - Michael G. Berouti, John Makhoul:
High quality adaptive predictive coding of speech. 303-306 - Daniel J. Esteban, Claude R. Galand, Daniel Mauduit, Jean E. Menez:
9.6/7.2 Kbps Voice excited predictive coder (VEPC). 307-311 - Samar K. Chakravarty, Pradip K. Srimani:
An optimal adaptation logic for delta modulation. 312-315 - Predrag M. Petrovic:
Forward-adaptive delta modulator without explicit transmission of step size. 316-319 - Daniel J. Esteban, Claude R. Galand:
32 Kbps CCITT Compatible split band coding scheme. 320-325 - Thomas P. Barnwell III, Ronald W. Schafer, Aubrey M. Bush:
Evaluation of LPC/CVSD tandem connections. 326-329 - L. E. Bergeron:
A spectral enhancement procedure for the wideband/Narrowband tandem. 330-333 - Wolfgang J. Hess, Josef Heiler:
Towards a variable frame rate speech transmission system with frame selection by time-domain segmentation - a status report. 334-337 - Anil K. Jain, Demitri M. Maroulis:
Two dimensional data compression of speech. 338-340 - James M. Alsup, Harper J. Whitehouse:
Two-dimensional speech compression. 341-344
Spectral Analysis
- Larry Marple:
High resolution autoregressive spectrum analysis using noise power cancellation. 345-348 - Edgar H. Satorius, S. Thomas Alexander:
High resolution spectral analysis of sinusoids in correlated noise. 349-351 - Leland B. Jackson, Donald W. Tufts, Frank K. Soong, Rahul M. Rao:
Frequency estimation by linear prediction. 352-356 - Steven M. Kay:
Improvement of autoregressive spectral estimates in the presence of noise. 357-360 - Gervasio Prado, Paul Moroney:
The accuracy of center frequency estimators using linear predictive methods. 361-364 - R. Jeffrey Keeler, Robert W. Lee:
Complex covariance/Maximum entropy doppler estimates for pulsed CO2lidar. 365-368 - Francis Landolf:
An error formula for iterative prefiltering frequency estimates. 369-371 - Dean P. Kolba, Thomas W. Parks:
Extrapolation and spectral estimation for bandlimited signals. 372-374 - James E. Youngberg, Steven F. Boll:
Constant-Q signal analysis and synthesis. 375-378 - John E. Timm:
Spectrum analysis using frequency-domain adaptive windowing. 379-382 - Nirode C. Mohanty, L. O. Krause:
Spectrum estimation of non-uniform sampled data. 383-386
Signal Processing Methods for Ecological Systems
- Carolyne M. Gowdy:
Input signals for sampling and identification of ecological systems. 387-390 - James Hill, Susan L. Durham:
Input, signals and control in ecosystems. 391-397 - Robert H. Flake:
Extension of Levins loop analysis to transient and periodic disturbances. 398-401 - T. C. Vorce, Robert J. Mulholland:
Estimating Oxygen demand in aquatic ecosystems. 402-404
Automatic Recognition of Continuous Speech
- Wayne A. Lea, June E. Shoup:
Gaps in the technology of speech understanding. 405-408 - M. Mohan Sondhi, Stephen E. Levinson:
Computing relative redundancy to measure grammatical constraint in speech recognition tasks. 409-412 - George M. White:
Dynamic programming, the viterbi algorithm, and low cost speech recognition. 413-417 - Lalit R. Bahl, James K. Baker, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer:
Automatic recognition of continuously spoken sentences from a finite state grammer. 418-421 - Lalit R. Bahl, James K. Baker, Paul S. Cohen, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer:
Recognition of continuously read natural corpus. 422-424 - Yasuhisa Niimi, Yutaka Kobayashi:
A voice-input programming system using basic-like language. 425-428 - R. D. Glave, G. van der Giet:
The Dawid speech recognition system. 429-432 - Lorenza Saitta:
Fuzzy semantic network for a speech understanding system - an experimental study. 433-436
Narrowband Speech Communication
- Luís F. Rocha:
A simple vocoder. 437-440 - Satoshi Imai:
Low bit rate cepstral vocoder using the log magnitude approximation filter. 441-444 - Thomas F. Quatieri:
CCD CZT Spectral analysis applied to real time homomorphics speech analysis-synthesis. 445-449 - Richard L. Cann, Kenneth Steiglitz:
Implementation of a pole-zero analysis - synthesis system for speech. 450-453 - John M. Turner, Bradley W. Dickinson:
A variable frame length linear predictive coder. 454-457 - Edward McLarnon:
A method for reducing the transmission rate of a channel vocoder by using frame interpolation. 458-461 - Jesse W. Fussell, Barry M. Abzug, Paul W. Boudra Jr., Michael D. Cowing:
Providing channel error protection for a 2400 bps linear predictive coded voice system. 462-465 - G. Robert Redinbo:
Channel coding considerations for digital speech encoded by linear prediction. 466-471 - Hidefumi Kobatake, Junta Inari, Shin-ichi Kakuta:
Linear predictive coding of speech signals in a high ambient noise environment. 472-475 - M. R. Ashouri, Anthony G. Constantinides:
Linear prediction techniques in the Walsh spectral domain for speech analysis and synthesis. 476-479 - Charlton M. Walter:
Application of canonical coordinate methods to the characterization of a family of error minimizing signal compression techniques. 480-482
Digital Filters
- Fred Mintzer, Bede Liu:
An estimate of the order of an optimal FIR band-pass digital filter. 483-486 - Yrjö Neuvo, Tapio Saramäki, Robert A. Gabel:
Digital filters with prescribed zeros. 487-490 - Pierre Duhamel:
Synthesis of digital filters with very low sensitivity of the frequency response to the change of the cofficients. 491-494 - Rakesh K. Patney, S. C. Dutta Roy:
Design of IIR filters using Pseudo-Boolean methods. 495-498 - Madihally J. Narasimha, Allen M. Peterson:
Design and applications of uniform digital bandpass filter banks. 499-503 - Maurice G. Bellanger:
A multiplexing scheme for multirate digital filtering with half-band filters. 504-507 - Horacio G. Martinez, Thomas W. Parks:
New class of recursive digital filters for decimation. 508-511 - James E. Heller:
An optimal filter design for variable sampling rates. 512-515 - S. C. Dutta Roy, Anurag Agrwal:
Digital lowpass filtering using the discrete Hilbert transform. 516-519 - James R. Rowland, Willard M. Holmes:
Nonstationary signal processing and model validation. 520-523
Seismic and Ultrasonic Modeling and Processing
- N. E. Nahl, Jerry M. Mendel, Leonard M. Silverman:
Recursive derivation of reflection coefficients from noisy seismic data. 524-528 - Chi Hau Chen:
On digital signal modelling and classification with the teleseismic data. 529-531 - R. M. Alford, K. R. Kelly, N. D. Whitmore:
Acoustic wave propagation by finite-difference techniques. 532 - Ramesh Shankar, Robert J. McDonough:
Ultrasonic measurements of defects in metals using cepstral processing. 533-537 - A. L. Frisillo, C. F. Hadley:
Digital recording of ultrasonic signals. 538-540
Automatic Phoneme Recognition
- Patrick F. Castelaz, Russell J. Niederjohn:
A comparison of linear prediction, FFT, and zero-crossing analysis techniques for vowel recognition. 541-545 - Vishwa Gupta, J. Kent Bryan, John N. Gowdy:
Speaker-independent vowel indetification in continuous speech. 546-548 - T. J. Edwards:
A probabalistic vector model for identification of intervocalic stop consonants. 549-552 - H. A. Barger, K. R. Rao:
A comparative study of phonemic recognition by discrete orthogonal transforms. 553-556 - Campbell L. Searle, J. Zachary Jacobson, S. G. Rayment:
A phoneme recognition system based on human audition. 557-560 - Matti Jalanko, Teuvo Kohonen:
Application of the subspace method to speech recognition. 561-564 - Seppo Haltsonen, Matti Jalanko, Kalle-J. Bry, Teuvo Kohonen:
Application of novelty filter to segmentation of speech. 565-568
Speech Quality and Enhancement
- Ludwig D. J. Eggermont, Eise C. Dijkmans:
Signal/Quantizing-distortion ratio measurements of fast-adaptive delta modulation systems. 569-572 - Bishnu S. Atal, Manfred R. Schroeder:
Predictive coding of speech signals and subjective error criteria. 573-576 - Christine H. Shadle, Bishnu S. Atal:
Speech synthesis by linear interpolation of spectral parameters between dyad boundaries. 577-580 - Barbara J. McDermott, Carlo Scagliola, David J. Goodman:
Perceptual and objective evaluation of speech processed by adaptive differential PCM. 581-585 - José M. Tribolet, Peter Noll, Barbara J. McDermott, Ronald E. Crochiere:
A study of complexity and quality of speech waveform coders. 586-590 - R. Viswanathan, William Russell, John Makhoul:
Objective speech quality evaluation of narrowband LPC vocoders. 591-594 - Thomas P. Barnwell III, Aubrey M. Bush:
Statistical correlation between objective and subjective measures for speech quality. 595-598 - Jae S. Lim:
Estimation of LPC coefficients from speech waveforms degraded by additive random noise. 599-601 - Robert A. Curtis, Russell J. Niederjohn:
An investigation of several frequency-domain processing methods for enhancing the intelligibility of speech in wideband random noise. 602-605 - Steven F. Boll:
Suppression of noise in speech using the saber method. 606-609 - Marvin R. Sambur:
LMS Adaptive filtering for enhancing the quality of noisy speech. 610-613
Transforms and Algorithms
- R. H. VanderKraats, Anastasios N. Venetsanopoulos:
Two dimensional filtering using fermat number transforms. 614-618 - Pierre R. Chevillat, Felix Closs:
Signal processing with number theoretic transforms and limited word lengths. 619-623 - Eric Dubois, Anastasios N. Venetsanopoulos:
Number theoretic transforms with modulus 22q- 2q+ 1. 624-627 - N. Sridhar Reddy, V. Umapathi Reddy:
Simulation of large length filters using fermat number transforms. 628-631 - James K. Beard
:
An inplace self recordering FFT. 632-633 - K. M. Cho, Gabor C. Temes:
Real-factor FFT algorithms. 634-637 - Henri J. Nussbaumer:
New algorithms for convolution and DFT based on polynomial transforms. 638-641 - Bhagwati Prasad Agrawal, Kishan Shenoi:
M-Adic invariant filters. 642-645 - James A. Cadzow:
Reconstruction of signals from their linear mapping image. 646-650 - William H. Haas, Claude S. Lindquist:
Linear detection filtering for the context of a least-squares estimator for signal processing applications. 651-654 - H. R. Bilger, J. P. Nougier:
Dealiasing of the spectra of sampled noise. 655-658
Underwater Acoustic Signal Processing
- H. J. Young:
Underwater sound arrival angle estimation by multiple cross-correlation measurements. 659-664 - Yiu Tong Chan, R. V. Hattin, J. B. Plant:
The least squares estimation of time delay and its use in signal detection. 665-669 - Everett H. Scannell Jr., G. Clifford Carter:
Confidence bounds for magnitude-squared coherence estimates. 670-673 - James L. Roberts:
An all-digital phase-measurement techniques using clipped-quadrature correlation. 674-677 - John Patrick Kuhn:
Detection performance of the smooth coherence transform (SCOT). 678-683 - David J. Quarmby, Geoffrey M. Duck:
Object classification using sonar data. 684-687 - Jude Franklin:
A study of stochastic processes associated with sonar detection. 688-691 - Albert A. Gerlach:
Acoustic transfer function of the ocean for a motional source. 692-695 - Raymond L. Veenkant:
Display and interpretation of a time-spread underwater acoustic channel's bandlimited impulse response. 696-699
Discrete Word Recognition
- Stephen E. Levinson, Aaron E. Rosenberg:
Some experiments with a syntax directed speech recognition system. 700-703 - Silvano Rivoira, Pietro Torasso:
Performance analysis of syntactic-recognizer of isolated words. 704-707 - Paul Mermelstein:
Recognition of monosyllabic words in continuous sentences using composite word templates. 708-711 - Mark F. Medress, Timothy Diller, Dean R. Kloker, Larry L. Lutton, Henry N. Oredson, Toby E. Skinner:
An automatic word spotting system for conversational speech. 712-717 - Aaron E. Rosenberg, C. E. Schmidt:
Directory assistance by means of automatic recognition of spoken spelled names. 718-721 - Günther Ruske, Thomas Schotola:
An approach to speech recognition using syllabic decision units. 722-725 - Sei-ichi Nakagawa, Toshiyuki Sakai:
A word recognition method from a classified phoneme string in the Lithan speech understanding system. 726-730 - T. K. Raja, B. Yegnanarayana:
Nearest neighbour decision rule for vowel and digit recognition. 731-734 - Ken'iti Kido, Jouji Miwa, Shozo Makino, Yoshihiro Niitsu:
Spoken word recognition system for unlimited speakers. 735-738
Two-Dimensional Filtering
- Russell M. Mersereau:
Two-dimensional signal processing from hexagonal rasters. 739-742 - Gerald M. Flachs, Wiley E. Thompson, U Yee Hsun, Steve Szymanski:
Digital decomposition and representation of video signals using projection theory. 743-746 - Nirode C. Mohanty:
On generation of two-dimensional data. 747-750 - James H. Justice:
Inversion of block - Toeplitz matrices using bivariate szego polynomials. 751-752 - Ronald F. Stork, Elmer A. Hoyer:
Two-dimensional zoom FFT. 753-756 - A. Chottera, Graham A. Jullien:
Recursive digital filters in image processing. 757-760 - Michael P. Ekstrom, Richard E. Twogood, John W. Woods:
Design of stable 2-D half-plane recursive filters using spectral factorization. 761-764 - Amar M. Ali, Anthony G. Constantinides:
Design of inherently stable two-dimensional recursive filters imitating the behaviour of one-dimensional analog filters. 765-768 - Gary A. Shaw:
An algorithm for testing stability of two-dimensional digital recursive filters. 769-772 - P. Karivaratharajan, M. N. Shanmukha Swamy:
Computer aided generation of two dimensional transfer functions from one dimensional transfer functions. 773-776 - Hyokang Chang, Jake K. Aggarwal:
Design of semicausal two-dimensional recursive filters. 777-781 - Tzeng-Tung Hwang:
A well suited two-dimensional linear recursive filter for image processing. 782-787
Signal Processing Hardware and Software
- Ahmad I. Abu-El-Haija, Madihally J. Narasimha, Allen M. Peterson:
A simple hardware implementation of digital notch filters. 788-791 - C. P. Reddy, Erik Fountain:
Digital phase locked loop. 792-795 - K. Wayne Current, Douglas A. Mow:
Parallel counter design using four-valued threshold logic. 796-799 - Ben-Dau Tseng, William C. Miller, Graham A. Jullien, J. J. Soltis, A. Baraniecka:
An error anaylsis of a FFT implementation using the residue number system. 800-803 - W. Kenneth Jenkins:
Techniques for residue-to-analog conversion for high data rate digital filtering. 804-807 - George A. Morris Jr., Helmut C. Wilck:
JPL 220Channel 300 MHz bandwidth digital spectrum analyzer. 808-811 - John D. Mackay, Harvey F. Silverman:
A 16-bit microprocessor-based digital filter architecture. 812-815 - Joseph R. Fisher, Martin E. Kaliski:
The compass block-diagram compiler: A new development in signal-processor programming. 816-819
General Acoustics: Equipment, Measurement and Theory
- Richard C. Cabot:
Impulse response testing of acoustic spaces. 820-823 - Ken'iti Kido, Masaaki Ishigame, Masato Abe:
Super directive spectrum analyzer by use of moving microphones. 824-827 - Tsuneo Nitta, Masatoshi Tanaka:
Free-field measurements for a loudspeaker system in a normal room-Using digital signal processing techniques. 828-831 - W. Marshall Leach Jr.:
Correction of near-field acoustic measurements made with arbitrary measuring transducers. 832-835 - Yoshiro Miida:
Reflection of spherical sound wave from a rigid sphere. 836-839 - Ali Zolfaghari:
Statistical properties of the poisson reverberation envelope. 840-845 - Robert Kalaba, Elena Zagustin:
On an extension of A. N. Krylov's numerical method for determining the frequencies of small vibrations of systems with damping. 846-847
manage site settings
To protect your privacy, all features that rely on external API calls from your browser are turned off by default. You need to opt-in for them to become active. All settings here will be stored as cookies with your web browser. For more information see our F.A.Q.
last updated on 2024-12-27 21:13 CET by the dblp team
all metadata released as open data under CC0 1.0 license
see also: Terms of Use | Privacy Policy | Imprint
dblp was originally created in 1993 at:
since 2018, dblp has been operated and maintained by:
the dblp computer science bibliography is funded and supported by: