NDLI: Minimax optimization of two-dimensional focused nonuniformly spaced arrays

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Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181)

1997 IEEE International Conference on Acoustics, Speech, and Signal Processing

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1995 International Conference on Acoustics, Speech, and Signal Processing

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Volume 4 - 1979

2-D FIR filter design from independent "Small" generating kernels using a mean square and Tchebyshev error criterion

A comparison of different window formulations for two-dimensional FIR filter design

Stabilization of two-dimensional recursive filters

Generation of two-dimensional digital functions without non-essential singularities of the second kind

Why filter recursively in two dimensions?

Design of 2-D recursive filters with separable denominator transfer functions

Space-domain design of two-dimensional recursive digital filters

The design of 2-D recursive filters in the 2-D reflection coefficient domain

Two-dimensional half-plane recursive filter design

Design of stable symmetric and non-symmetric half-plane digital recursive filters

On synthesizing natural-sounding speech by linear prediction

Linear prediction of formants for low bit rate digital speech transmission

A two-step speech compression system with vector quantizing

A mixed-phase homomorphic vocoder

Narrowband LPC speech transmission over noisy channels

A robust vocoder with pitch-adaptive spectral envelope estimation and an integrated maximum-likelihood pitch estimator

A linear predictive vocoder with new pitch extraction and exciting source

A novel vocoder concept based on discrete time acoustic tubes

Linear prediction of transformed speech

An analysis/Synthesis framework for transform coding of speech

Computer recognition of stop consonants

Automatic discrimination of fricative consonants based on human audition

On sensitivity of vocal tract area functions

Spectral classification of phonemes by learning subspaces

Speech segmentation and recognition using syntactic methods on the direct signal

An experimental system for acoustic-phonetic decoding of continuous speech

Automatic selection of phonemes from an equally spaced quasi-phoneme string by the entropy principle

A unique real-time speech decoder that operates from new perspectives

Experiments on spectrogram reading

Testing sonar data for multivariate normality

A quantitative study of optimum and sub-optimum filters in the generalized correlator

A parameter estimation approach to time delay estimation

Local estimation of delay parameter following robust detection

Detection performance of an adaptive processor in non-stationary noise

Robust sequential detection of narrowband acoustic signals in noise

A phase-coherence detector/Estimator

Sources of and remedies for spectral line splitting in autoregressive spectrum analysis

Spectral analysis using prediction methods

Spectral line analysis by Pisarenko and Prony methods

Fourier-autoregressive spectral estimation

Maximum entropy (adaptive) filtering applied to radar clutter

Digital signal processing for doppler radar signals

Fast estimation of narrowband spectra

A family of phase complementary filters

Spectral analysis using the Karhunen-Loeve transform

Magnitude-phase relationships for short-time Fourier transforms based on Gaussian analysis windows

Extrapolation and spectral estimation for bandlimited, time-concentrated signals

Optimal estimation of essentially and strictly bandlimited signals and their spectrum by generalized splines

A spectral subtraction algorithm for suppression of acoustic noise in speech

Reduction of nonstationary acoustic noise in speech using LMS adaptive noise cancelling

Enhancement of speech corrupted by acoustic noise

A frequency domain noise cancelling preprocessor for narrowband speech communications systems

Performance of LPC vocoders in a noisy environment

LPC voice digitizer with background noise suppression

Maximum likelihood parameter estimation of noisy data

Estimating the parameters of a noisy all-pole process using pole-zero modeling

CCD adaptive filtering for robust LPC speech processing

An approach to speaker normalization in an automatic speech recognition system

A new system for continuous speech recognition - preliminary results

Automatic recognition of continuous digits sequences by means of segmentation and dynamic programming

Recognition results for several experimental acoustic processors

The structure of a lexicon for a speech understanding system

Phonetic word verification

Performance statistics of the HEAR acoustic processor

Techniques for recognition of spectrogram patterns based on dynamic modeling

Speech recognition in the context of two-way immediate person-machine interaction

A review of adaptive antennas

An array optimization technique

Adaptive array processing: Time vs. frequency domain

Minimax optimization of two-dimensional focused nonuniformly spaced arrays

Effects of random shading, phasing errors and element failures on the beampatterns of line and planar arrays

A computer model for the analysis of source motion, the ocean environment, and interference effects on acoustic signal coherence

The roles of integration time and acoustic multipaths in determining the structure of CW line spectra

Impact of the ocean acoustic transfer function on the coherence of undersea propagations

Influence of the spatial coherence of the background noise on high resolution passive methods

Audio signal processing with transversal filters

Constant-Q analysis using the chirp z-transform

Design of a wideband, constant beamwidth, array microphone for use in the near-field

Time domain measurement of loudspeaker driver parameters

A new computer aided method for the complete acoustical design of broadcasting and recording studios

Auditory backward inhibition can ruin a concert hall

Constrained minimization of roundoff noise in fixed-point digital filters

Normal realizations of IIR digital filters

Reduced multiplier, low roundoff noise digital filters

Linear transformations for the design of digital and active filters

Coefficient quantization effects on pole locations for state model digital filters

A state-space approach for elimination of limit cycles in digital filters with arbitrary structures

Passive cascaded lattice digital filters

Simplified error models for digital filters

Narrowband recursive filters with error spectrum shaping

Some considerations on coefficient sensitivity and noise in direct form IIR interpolators and decimators

Coefficient inaccuracy in FIR filters

Effects of finite coefficient precision on FIR filter spectra

Accumulation of distortion in signal processing systems

Identification of complex autoregressive processes

Geometrical characterization of the canonical coordinate basis underlying a family of error minimizing signal compression techniques

Inversion of signal operations

The solution of discrete convolutions with a bounded error constraint

An experimental study of the effects of noise on a class of iterative deconvolution algorithms

System identification using a maximum-likelihood spectral matching technique

Spectral root homomorphic deconvolution system

Evaluation of the Steiglitz algorithm for estimating the parameters of an ARMA process

Modulo-PCM: A new source coding scheme

A generalized adaptive quantization system with a new reconstruction method, for noisy transmission

High-frequency regeneration in speech coding systems

Bit-rate-halving algorithm for PCM-encoded speech using a new bidimensional data compression scheme

A high-gain DSI-ADPCM system

Digital transmission of commentary-grade (7 kHz) audio at 56 or 64 kb/s

Variable rate coding

A methodology for studying telephone amplitude distortion effects on narrowband speech processors

Optimizing predictive coders for minimum audible noise

Talker variance and phonetic feature variance in diagnostic intelligibility scores for digital voice communications processors

Intelligibility of intervocalic consonants in noise

Optimum linear filter for speech communication

Subjective evaluation of SPAC in improving the quality of noisy speech

The presentation of continuous speech with synchronous printed text

The effects of Teflon injection on laryngeal dynamics

A conceptually unique speech training aid system

SIRENE, a system for speech training of deaf people

New algorithms for fast convolution based on convolution preserving spline signals

Hardware implementation of convolution using number theoretic transforms

Computation of Fourier integral using polynomial interpolation

Round-off error propagation in Durbin's, Levinson's, and Trench's algorithms

Resolution of superposed signals with envelope-constrained filters

On complexity of fast convolution algorithms

New polynomial transform algorithms for fast DFT computation

Parallelism in the computation of the FFT and the WFTA

Complex rectangular transforms

Multidimensional DFT processing in subspaces whose dimensions are relatively prime

A novel approach for implementing pitch prediction in sub-band coding

Sub-band coder design incorporating quadrature filters and pitch prediction

Speech coding based on a composite - Gaussian source model

An efficient coding of the prediction residual

An improved residual encoder for speech compression

Description of a hybrid 7.2 kbps vocoder

A new configuration for speech digitization at 9600 bits per second

Development of a 4.8-9.6 kbps RELP vocoder

Voice-excited LPC coders for 9.6 kbps speech transmission

8 kbps voice transmission by SPAC

Frequency warping for nonuniform talker normalization

Order dependence in templates for monosyllabic word identification

Speaker independent recognition of isolated words using clustering techniques

Considerations in applying clustering techniques to speaker independent word recognition

A voice-controlled mechanical arm for immobilized patients

Syntax and semantics in a word-sequence recognition system

A redundant hash addressing method adapted for the postprocessing and error-correction of computer recognized speech

Speech characterization from a rough spectral analysis

Word spotting in conversational speech

Adaptive linear prediction filtering for airborne underwater acoustic signal processors

A sonar target recognition experiment

A broadband echo ranging system for measuring the frequency characteristics of fish schools

Experiments with a large aperture parametric acoustic receiving array

A passive detection system for a wide class of illuminator signals

Filter design for estimating human blood-flow velocity

A phase plane method for the analysis of seismic signals

Phase in speech and pictures

Two-dimensional signal filters under modeling uncertainties

Multiple model recursive estimation of images

On state-space signal processing with application to image enhancement

Image enhancement by stochastic homomorphic filtering

A two-dimensional filter design for isotropic reconstruction of track type airborne geophysical surveys

Fast least-squares phase estimation in speckle imaging

Frequency and bearing estimation by two-dimensional linear prediction

A two-dimensional maximum entropy spectral estimator

High-resolution two-dimensional spectral analysis

A separable 2-D autoregressive spectral estimation algorithm

Recursive digital filters with low coefficient sensitivity

A high-level block-diagram signal processing language

A design and computing system for signal processing applications

Design of real-time signal processing software for efficient use of high-speed array processors in multitasking environments

Efficient implementation of one and two dimensional digital signal processing algorithms on a multi-processor architecture

A programmable signal processing architecture

A real time analysis/Display system for nonstationary coastal processes

Digital processing of Seasat SAR data

Optimal estimation and speech analysis

Sequential gradient estimation predictor for speech signals

Reflection coefficient estimates based on a Markov chain model

LPC analysis using a variable acoustic tube model

Reduction of computation in pole-zero modeling of speech signals

Statistical properties of an LPC distance measure

A distance measure based on the derivative of linear prediction phase spectrum

Rate/Pitch modification of speech using the constant-Q transform

Features for the identification of mixed excitation in speech analysis

Automatic thresholding for voicing detection algorithms

Rank-order speech classification algorithm (RASCAL)

Multichannel zero-crossing-interval pitch estimation

Pitch extraction using MOS-LSI circuitry

Time-domain pitch period extraction of speech signals using three nonlinear digital filters

A pattern recognition approach to compare natural and synthesized speech

Some factors influencing the performances of a speaker recognition system based on LPC

Accuracy of speaker verification via orthogonal parameters for noisy speech

Structure and performance of an on-line speaker verification system

Toward the development of practical methods of evaluating speaker recognizability

Multistage decision schemes for speaker recognition

New transformed variables for designing recursive digital filters

On the design of complementary filters

A method for the design of phase equalizers

Design of stable all-pass filters

Statistical design of ARMA filters

Linear prediction in the design of Hilbert transformers

Optimal design of digital Hilbert transformers with a concavity constraint

A maximally flat filter design algorithm for quadrature mirror filters (QMF)

Application of transposition to decimation and interpolation in digital signal processing systems

Anti-alias filters for tuner applications

On the design of recursive lowpass digital filters

On the stability of a 1-bit-quantized feedback system

FIR filter hardware reduction with adaptive delta-modulation

Correlation detection of an FTH - signal using a CCD/CZT implementation

Optimization of digital signal processors using array processor and CCD technology

A high data rate, low power all-digital correlation circuit design

New concept of digital multi-frequency receiver

A parallel processor for real-time speech signal processing

Microprocessor based real-time speech processor

A microprocessor-based digital filter laboratory station for an undergraduate signal processing laboratory

Real-time text processing for Italian speech synthesis

Eight-channel digital speech synthesizer based on LPC techniques

A microprocessor based audio response system

Diphone synthesis for phonetic vocoding

Time domain speech synthesis-by-rules using a flexible and fast signal management system

The MISS speech synthesis system

An LPC k-parameter software speech synthesizer via dynamic microprogramming a general purpose computer

A fast FORTRAN implementation of the U. S. naval research laboratory algorithm for automatic translation of english text to VOTRAX parameters

Analysis of formant and pitch information for Spanish phonemes

A microcomputer-based speech synthesizer which speaks Spanish

Speech synthesis from vocal tract area function acoustical measurements

Adaptive structures for multiple-input noise cancelling applications

Adaptive noise cancelling of a sinusoidal interference using a lattice structure

γ-LMS and its use in a noise-compensating adaptive spectral analysis technique

Noise cancellation via linear prediction filtering

On the "Desired behavior" of adaptive signal processing algorithms

Detection of sinusoids by linear prediction filter frequency response

The spectral dynamics of evolving LMS adaptive filters

Filtering of narrowband signals using analytic sianal properties

A fixed-point iteration algorithm for adaptive linear estimation applied to spectral line enhancement

A multidimensional Modelling approach to texture classification and segmentation

Locating a passive source with array measurements a summary of results

A preprocessing filter for enhancing LPC analysis/Synthesis of noisy speech

A 4800 bps voice excited predictive coder (VEPC) based on improved baseband/Sub-bands filters

Optimum quantizer algorithm for real-time block quantizing

Real time signal processing software for multiplierless microprocessors

Evaluation of two narrowband speech algorithms

ICASSP '78. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '77. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '76. IEEE International Conference on Acoustics, Speech, and Signal Processing

Minimax optimization of two-dimensional focused nonuniformly spaced arrays

Content Provider	IEEE Xplore Digital Library
Author	Kraft, R. McDonald, J. Ahlgren, F.
Copyright Year	1979
Description	Author affiliation: Rensselaer Polytechnic Institute, Troy, New york (Kraft, R.)
Abstract	Recently there has been interest in applying the results of digital filter design theory to the optimization of the beam forming pattern of antenna arrays. The techniques employed usually involve minimax criteria for side lobe suppression. Certain applications require focused operation in order to make use of the near field for imaging. Nonuniform array element spacings have been successfully employed to reduce side lobes in phased arrays and to minimize the number of elements. Presented here will be a combination of nonuniform spacing, focusing, and element shading using some optimizing window designs to form a narrow center beam with very good side lobe suppression. The treatment is unique in that it explicitly incorporates the effects of focusing in the optimization procedure.
Starting Page	286
Ending Page	289
File Size	143808
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1979.1170685
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1979-04-02
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Minimax techniques Phased arrays Apertures Focusing Diffraction Interference Lenses Antenna arrays Design optimization Collimators
Content Type	Text
Resource Type	Article

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