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ORCIDICASSP 1984: San Diego, California, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '84, San Diego, California, USA, March 19-21, 1984. IEEE 1984
Narrowband and Low Rate Speech Coding
- Salim E. Roucos, Alexander MacLeod Wilgus:
Speaker normalization algorithms for very-low-rate speech coding. 1-4 - Gérard Benbassat, Xavier Delon:
Low bit rate speech coding by concatenation of sound units and prosody coding. 5-8 - Sharad Singhal, Bishnu S. Atal:
Improving performance of multi-pulse LPC coders at low bit rates. 9-12 - Vijay K. Jain, R. Hangartner:
Efficient algorithm for multi-pulse LPC analysis of speech. 13-16 - A. Parker, S. Thomas Alexander, Henry J. Trussell:
Low bit rate speech enhancement using a new method of multiple impulse excitation. 17-20 - Per Hedelin:
A glottal LPC-vocoder. 21-24 - George S. Kang, Stephanie Everett:
Improvement of the narrowband LPC synthesis. 25-28 - Matthew I. Noah:
Optimal Lloyd-Max quantization of LPC speech parameters. 29-32 - J. Koljonen, Matti Karjalainen:
Use of computational psychoacoustical models in speech processing: Coding and objective performance evaluation. 33-36 - Frank K. Soong, Biing-Hwang Juang:
Line spectrum pair (LSP) and speech data compression. 37-40 - Chia-Chuan Hsiao, Robert W. Brodersen:
A multirate root LPC speech synthesizer. 41-44 - Jean-Pierre Adoul, Claude Lamblin, Alain Le Guyader:
Baseband speech coding at 2400 bps using "Spherical vector quantization". 45-48
Speech Analysis and Synthesis
- Osamu Kakusho, Masuzo Yanagida, Riichiro Mizoguchi:
A sample selective linear prediction analysis of speech. 49-52 - Hynek Hermansky, Hiroya Fujisaki, Yasuo Sato:
Spectral envelope sampling and interpolation in linear predictive analysis of speech. 53-56 - Z. Shpiro, David Malah:
An algebraic approach to discrete short-time Fourier transform analysis and synthesis. 57-60 - Daniel W. Griffin, Douglas S. Deadrick, Jae S. Lim:
Speech synthesis from short-time Fourier transform magnitude and its application to speech processing. 61-64 - Melvyn J. Hunt, C. P. Swail:
Time alignment of natural speech to synthetic speech. 65-68 - P. Jeffrey Bloom:
Use of dynamic programming for automatic synchronization of two similar speech signals. 69-72 - Hong C. Leung, Victor W. Zue:
A procedure for automatic alignment of phonetic transcriptions with continuous speech. 73-76 - Mark Anderson, Janet B. Pierrehumbert, Mark Liberman:
Synthesis by rule of english intonation patterns. 77-80 - Rodolfo Delmonte, Gian Antonio Mian, Graziano Tisato:
A text-to-speech system for italian. 81-84 - Juan Carlos Olabe, Andrés Santos
, R. Martínez, Elias Munoz-Merino, M. Martínez, A. Quilis, Jared Bernstein:
Real time text-to-speech conversion system for spanish. 85-87 - Roger M. Meli, Frank Fallside:
A system for converting teletext into speech. 88-90 - M. Remmel, T. Tago:
Towards increasing the commercial success of speech synthesizers. 91-93
Adaptive Filtering I
- D. Marginedes:
Fast frequency tracking using an adaptive lattice filter for a vortex flowmeter signal. 94-97 - John R. Treichler, Michael G. Larimore:
A real-arithmetic implementation of the constant modulus algorithm. 98-101 - John Y. Cheung:
Convergence conditions for an optimal solution in adaptive recursive filters. 102-105 - C. R. South, A. V. Lewis:
Extension facilities and performance of an LSI adaptive filter. 106-109 - Neil J. Bershad, Lian Zuo Qu:
On the probability density functions of the weight for the complex scalar LMS adaptive algorithm. 110-113 - Neil J. Bershad, Yuh-Huu Chang:
Time correlation statistics of the LMS adaptive algorithm weights. 114-117 - Fuyun Ling, John G. Proakis:
Nonstationary learning characteristics of least squares adaptive estimation algorithms. 118-121 - Yan Chen:
Stable time- and order-recursive algorithm for the adaptive lattice filter. 122-125 - Cumhur Cengiz Evci, Maurice G. Bellanger:
Characteristics of adaptive filters with leakage. 126-129 - Giovanni L. Sicuranza, Andrea Bucconi, Paolo Mitri:
Adaptive echo cancellation with nonlinear digital filters. 130-133 - Dae Hee Youn, S. Prakash:
On realizations and related algorithms for adaptive linear phase filtering. 134-137
2-D Spectral Estimation
- Ahmet H. Kayran, Sydney R. Parker, D. J. Klich:
Two-dimensional spectral estimation with autoregressive lattice parameters. 138-141 - Nan Miao, Zong-Zhi Chen:
Application of SVD to 2-D spectral estimation. 142-145 - Richard M. Leahy, Constantinos E. Goutis, S. Drossos:
Tomographic and spectral analysis using noise statistics. 146-149 - Govind Sharma, Ramalingam Chellappa:
An iterative algorithm for robust 2-D spectrum estimation. 150-153 - Ramalingam Chellappa, Govind Sharma:
A model based approach for 2-D MEPS analysis. 154-157 - David C. Munson Jr., Jorge L. C. Sanz:
The importance of random phase for image reconstruction from frequency offset Fourier data. 158-161 - Li-He Zou, Bede Liu:
Improvement of resolution and reduction of computation in 2D spectral estimation using decimation. 162-165 - Brian L. Hinman, Jared Bernstein, David H. Staelin:
Short-space Fourier transform image processing. 166-169 - Soo-Chang Pei, Eng-Fong Huang:
2D Discrete cosine transform computation by fast polynomial transform algorithms. 170-173 - D. V. Bhaskar Rao, Sun-Yuan Kung:
A state space approach for the 2-D harmonic retrieval problem. 174-176 - Nuthalapati U. Chowdary, Willem J. D. Steenaart:
A high speed two-dimensional FFT processor. 177-180 - S. Hamid Nawab, Farid U. Dowla, R. T. Lacoss:
A new method for wideband sensor array processing. 181-184
Parameter Estimation I
- Vijay K. Jain:
Accurate pole estimation by modified linear prediction. 185-188 - A. R. Gondeck, Vijay K. Jain:
Comparison of three auto-regressive modeling methods. 189-192 - Farid U. Dowla, Jae S. Lim:
Relationship between maximum-likelihood-method and autoregressive modeling in multidimensional power spectrum estimation. 193-196 - Michael T. Manry:
Parameter estimation algorithms utilizing implicit phase. 197-199 - Hossny El-Sherief:
Adaptive least-squares for parametric spectral estimation and its application to pulse estimation and deconvolution of seismic data. 200-203 - Guaning Su:
Signal subspace analysis and improvement of spectral estimation algorithms. 204-207 - Konstantinos Konstantinides, Kung Yao:
Applications of singular value decomposition to system modeling in signal processing. 208-211 - Finley R. Shapiro, Stephen D. Senturia, David Adler:
The analysis of non-oscillating transients using the covariance method with many different orders. 212-215 - Siew Kok Hui, Yong Ching Lim:
A new spectrum analysis approach using autocorrelation technique and MEM. 216-219 - Tryphon T. Georgiou:
Topological aspects of the caratheodory problem. 220-223
Parameter Estimation II
- Richard J. Vaccaro:
On adaptive implementations of Pisarenko's harmonic retrieval method. 224-227 - Yu Hen Hu, Yao-Cheng Ling:
Constrained lattice structures for harmonic retrieval. 228-231 - Mati Wax, Thomas Kailath:
Determining the number of signals by information theoretic criteria. 232-235 - Carlos H. Muravchik, Martin Morf:
Square root normalized feedback ladder algorithm for the identification of moving average systems. 236-239 - J. E. Hudson:
Fitting sinusoids to sampled data using a hybrid S/Z plane moment method. 240-243 - Robert D. Preuss, Rao K. Yarlagadda:
Autoregressive spectral estimation in noise with application to speech analysis. 244-247 - H. Madala:
A new harmonical algorithm for digital signal processing. 248-251 - Neil J. Malloy:
Non-uniform sampling for high resolution spectrum analysis. 252-255 - Fred D. Powell:
Sampling a broad band sparse spectrum without anti-aliassing filters. 256-258 - Koh-ichi Hashimoto, Akira Sano:
On-line trend detection based on ARI modeling. 259-262
Detection and Estimation
- Eugenio J. Tacconi, Michel Bouvet, Bernard C. Picinbono:
A new N.A.R power estimation for adaptive detection. 263-266 - S. L. Earp, Loren W. Nolte:
Multichannel adaptive array processing for optimal detection. 267-270 - Johann F. Böhme:
Estimation of source parameters by maximum likelihood and nonlinear regression. 271-274 - Ratnam V. Raja Kumar, Ranendra N. Pal:
Separation of sinusoids using the constrained adaptive line enhancer. 275-278 - Hong Wang, Mostafa Kaveh:
Estimation of angles-of-arrival for wideband sources. 279-282 - Albert A. Gerlach:
Performance characteristics of biased estimators. 283-286 - D. E. Ohlms, M. L. Hampton:
Track likelihood statistic for the I Q locked loop. 287-289 - Frank Cornett:
Kalman estimation of frequency-domain interference reduction coefficients for angle-modulated signals. 290-293 - Roger F. Dwyer:
Essential limitations to signal detection and estimation: An application to the arctic under ice environmental noise problem. 294-297
Design Methodology and Tools
- Gary E. Kopec:
The integrated signal processing system ISP. 298-301 - S. N. Terepin:
A graphical notation for describing data flow in digital filters. 302-305 - D. A. Schwartz, Thomas P. Barnwell III:
A graph theoretic technique for the generation of systolic implementations for shift-invariant flow graphs. 306-309 - H. V. Jagadish, Thomas Kailath, John A. Newkirk, Robert G. Mathews:
On hardware description from block diagrams. 310-313 - Juan-Manuel Jover, Thomas Kailath:
Design framework for systolic-type arrays. 314-317 - Don H. Johnson:
Signal processing software tools. 318-320 - Brian R. Mears:
A modular method for designing custom signal processing integrated circuits. 321-324 - Clifford T. Mullis, Richard A. Roberts:
Digital processing structures for VLSI implementation. 325-327 - Marc Beacken, Hungwen Li:
Evaluation of signal processing architectures via an integrated simulation environment. 328-331
Isolated-Word Speech Recognition I
- Y. J. Liu:
On creating averaging templates. 332-335 - Vishwa Gupta, Matthew Lennig, Paul Mermelstein:
Decision rules for speaker-independent isolated word recognition. 336-339 - Yasuhiro Matsuda, Shu Tezuka, Mitsuhiko Kanoh, Masafumi Nishimura, Toyohisa Kaneko:
A method for recognizing Japanese monosyllables by using intermediate cumulative distance. 340-343 - David K. Burton, Joseph T. Buck, John E. Shore:
Parameter selection for isolated word recognition using vector quantization. 344-347 - Aaron E. Rosenberg, Kathleen L. Shipley:
Evaluation of an isolated word recognizer in talker-dependent and talker-independent modes using a large telephone-band data base. 348-351 - J. Woodard, Wayne A. Lea:
New measures of performance for speech recognition systems. 352-355 - James K. Baker, Janet M. Baker, Robert Roth, Pard Bamberg:
Cost-effective speech processing. 356-359 - Seppo Haltsonen:
An endpoint relaxation method for dynamic time warping algorithms. 360-363 - B. Yegnanarayana, Sarat Chandran:
Performance of isolated word recognition system for degraded speech. 364-367 - George Vysotsky:
Speaker-independent isolated word recognition using a one-pass analysis. 368-371 - De-Yuan Cheng, Allen Gersho, Bhaskar Ramamurthi, Yair Shoham:
Fast search algorithms for vector quantization and pattern matching. 372-375 - Henry M. Dante:
On the problem of dimensionality and sample size in multi-stage pattern classifiers. 376-379 - Wendy J. Kessler, N. Rex Dixon, Harvey F. Silverman:
An experiment with a non-head-mounted microphone for discrete utterance recognition (DUR). 380-383
Mediumband Speech Coding I
- Michael G. Berouti, H. Garten, Peter Kabal, Paul Mermelstein:
Efficient computation and encoding of the multipulse excitation for LPC. 384-387 - G. A. Senensieb, A. J. Milbourn, A. H. Lloyd, I. M. Warrington:
A non-iterative algorithm for obtaining multi-pulse excitation for linear-predictive speech coders. 388-391 - Isabel Trancoso, Ramón García-Gómez, José M. Tribolet:
A study on short-time phase and multipulse LPC. 392-395 - Peter Kroon, Ed F. Deprettere:
Experimental evaluation of different approaches to the multi-pulse coder. 396-399 - Maurizio Copperi, Daniele Sereno:
9.6 kbit/s Piecewise LPC residual excited coder using multiple-stage vector quantization. 400-403 - Hüseyin Abut, Stephen A. Luse:
Vector quantizers for subband coded waveforms. 404-407 - Allen Gersho, Tor A. Ramstad, I. Versvik:
Fully vector-quantized subband coding with adaptive codebook allocation. 408-411 - Dietrich Wolf, Herbert Reininger, Jürgen Dziumbla:
A comparative study of various speech encoding schemes using vector quantization. 412-415 - Allen Gersho, Yair Shoham:
Hierarchical vector quantization of speech with dynamic codebook allocation. 416-419 - Amine Haoui, David G. Messerschmitt:
Predictive vector quantization. 420-423 - Torbjørn Svendsen:
Tree encoding of the LPC residual. 424-427 - Joël Soumagne, Jean-Pierre Adoul, Sarto Morissette:
A new concept for encoding speech amplitude time quantization. 428-431
Filter Design
- Palghat P. Vaidyanathan, Sanjit K. Mitra:
A new class of very low sensitivity cascade-form digital-filters based on "passive" second order single-input single-output building blocks. 432-435 - James F. Kaiser:
Some properties of a family of generalized time-limited window functions. 436 - Henri J. Nussbaumer, Martin Vetterli:
Computationally efficient QMF filter banks. 437-440 - Marc Beacken:
Efficient implementation of highly variable bandwidth filter banks with highly decimated output channels. 441-443 - Paul C. Millar:
Mirror filters with minimum delay responses for use in subband coders. 444-447 - Hans Wilhelm Schüßler, Peter Steffen:
On the reconstruction of a smooth function out of its samples. 448-451 - Alfred Fettweis, Josef A. Nossek, Klaus Meerkötter:
Reconstruction of signals after filtering and sampling rate reduction. 452-455 - Kenji Nakayama, T. Seki:
A design method for multirate filters. 456-459 - Kazunori Sugahara, Katsuhiko Hayashi, Kotaro Hirano, Sanjit K. Mitra:
N-path digital filters. 460-463 - Sailesh K. Rao, Thomas Kailath:
Pipelined orthogonal digital lattice filters. 464-467 - Adly T. Fam:
A multi-signal bus architecture for FIR filters with single bit coefficients. 468-470 - Yoram Bresler, Albert Macovski:
3-D reconstruction from projections based on dynamic object models. 471-474 - Mehrdad Soumekh, Mostafa Kaveh:
Image reconstruction from frequency domain data on arbitrary contours. 475-478 - Narciso García, Alberto R. Calero:
Faster phase only image reconstruction. 479-482