NDLI: Performance evaluation of a new robust detector for sonar signals

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2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)

1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258)

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

1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings

1995 International Conference on Acoustics, Speech, and Signal Processing

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1993 IEEE International Conference on Acoustics, Speech, and Signal Processing

[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing

[Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing

International Conference on Acoustics, Speech, and Signal Processing

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ICASSP '82. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '81. IEEE International Conference on Acoustics, Speech, and Signal Processing

Volume 6 - 1981

Front cover and table of contents

Recent developments in vector quantization for speech processing

LPC compressed speech at 850 bits-per-second

A system for very low data rate speech communication

Vector quantization of speech waveforms

Inter-speaker comparison of LPC acoustic space using a minimax distortion measure

Two dimensional prediction and interpolation for data rate compression of LPC parameters

A differential linear predictive voice coder for 1200 bps

A low-rate vocoder based on an adaptive subband formant analysis

Experiments with a pattern-matching channel vocoder

Convergence of iterative signal reconstruction algorithms

Highly parallel architectures for solving linear equations

Some intractable problems in digital signal processing

An improvement of the Van-Cittert's method

Methods for deconvolving sparse positive delta function series

Estimation and detection of variable latency signals via unwrapped phase averaging

A dynamic programming technique for nonlinear smoothing

Analysis of non-stationary signals--Digital filters vs. FFT

Time-frequency signal analysis by means of the Wigner distribution

Spectral modifications using linear shift-variant digital filters

Frequency errors in the spectral estimates of complex sinusoids using the "Tapered" burg technique

Nonlinear estimation of frequency and phase of a sinusoid in noise

On narrow-band spectrum calculation by direct decimation

A new approach to Fourier analysis of randomly sampled data using linear regression technique

Optimum sampling for estimation of Fourier coefficients in noise

On leakage reduction associated with spectral power domain averaging to improve estimate quality

SRS text-to-phoneme rules: A three-level rule strategy

Control of prosodic parameters for a formant synthesizer based on diphone concatenation

An approach to demisyllable speech synthesis of German words

Automatic stress assignment for Italian text-to-speech synthesis

Automatic generation of mixed excitation in a linear predictive speech synthesizer

Pitch invariant frequency lowering with nonuniform spectral compression

Third-octave analysis of multichannel amplitude compressed speech

A procedure for synchronizing continuous speech with its corresponding printed text

Vocal fold vibratory patterns: Comparison of film and inverse filtering

Feature extraction for laryngeal evaluation

Optimal array processing with 3-D random arrays

Array design for MEM and MLM array processing

Matched array processing for wideband passive sonar

Source power estimation method associated with high resolution bearing estimator

Maximum-entropy field-mapping in the presence of correlated multipath

Data turning--Average signal to noise ratio improvement

Improved detection performance of an FM signal by autoregressive spectral analysis

Performance evaluation of a new robust detector for sonar signals

Asymptotic performance measures for sequential partition detectors

Speaker-independent word recognition using sex-dependent clustering

Speaker identification and verification combined with speaker independent word recognition

Telephone-line speaker recognition using clipped autocorrelation analysis

A realtime implementation of a text independent speaker recognition system

Text-independent speaker recognition using orthogonal linear prediction

Rate distortion theory and predictive coding

A tone oriented voice excited vocoder

An allpass transformed lattice filter with improved sensitivity properties

A model for short-time phase prediction of speech

Linear predictor with a new error criterion

HQMF: Halfband quadrature mirror filters

Inverse filtering for systems with unit circle zeroes

Constrained optimization solutions to I.I.R filter design using discrete prolate spheroidal wave functions

On the design and performance of equiripple IIR interpolators

Equal ripple amplitude and maximally flat group delay digital filters

Thinning the impulse response of FIR digital filters

Adaptive design of digital filters

Least-squares algorithms for adaptive linear-phase filtering

On the structure of IIR filters using residue arithmetic

A nested algorithm for improving the accuracy of chirp-Fourier transform implementations

The multibridge charge coupled filter: General synthesis using bias conservative structure

The adaptive lattice array processor

Convergence models for adaptive gradient and least squares algorithms

Response of a delay-constrained adaptive linear predictor filter to a sinusoid in white noise

Enhancement of sinusoids in colored noise and the whitening performance of exact least-squares predictors

Use of linear prediction in detecting narrowband signals in colored noise

A comparison of adaptive gradient and adaptive least-squares algorithms

On the asymptotic behavior of a complex adaptive line enchancer (CALE)

A comparison of two fast linear predictors

Constrained algorithms for multi input adaptive lattices in array processing

An algebraic approach to super-resolution adaptive array processing

Another discrete Fourier transform computation with small multiplications via the Walsh transform

Multifactor algorithms for noncyclic digital convolution

Inverse polynomial transform algorithms for DFTs and convolutions

Corner-turn complexity properties of polynomial transform 2D convolution methods

Fixed-point error bound for convolution by polynomial transforms, with application to FIR filtering

Cyclic convolution algorithms over finite fields: Multidimensional considerations

Long convolutions using transforms over reducible extensions of fermat number rings

A new prime factor FFT algorithm

On the simulation of residue number systems

Speech deconvolution by recursive ARMA lattice filters

A pole-zero model for cepstrally smoothed speech spectra

Estimation of formant parameters by sum-of-poles modeling

An LPC vocoder for efficient implementation

Estimation of the dynamics of vocal tract parameters

A spline approach to speech analysis/Synthesis

QD-an algorithm for non-linear inverse filtering

Speech analysis for Chinese Putonghua (mandarin)

A family of high speed, floating point arithmetic chips

The Am29500 signal processing family

Application of the PDSP chip set to LPC synthesis

Conceptual system design for continuous speech recognition LSI

The implementation of a speech synthesis algorithm

On the use of filter design programs for generating spectral windows

Comparative performance of adaptive and conventional detectors for finite bandwidth signals

Hardware implementation of a 15-channel filter bank

Real-time implementation of sub-band coding on a programmable integrated circuit

Multiport implementation of real time 16kbps sub-band coder

A general purpose real time digital speech processor

LPC speech I.C. using a 12-pole cascade digital filter

A high speed array computer for dynamic time warping

Adaptive ARMA spectral estimation

Additional results on the Cadzow ARMA method for spectrum estimation

A superresolution method of ARMA spectral estimation

A recursive maximum likelihood algorithms for ARMA line enhancement

A comparison of spectral estimators for real data

Improved spectral resolution by energy-weighted prediction method

Pattern recognition of waveforms using modern spectral estimation techniques and its application to earthquake/Explosion data

Identification and spectral estimation of noisy multivariate autoregressive processes

Will the real MEM please stand up?

Modal decomposition of covariance sequences for parametric spectrum analysis

Adaptive MSE estimation

Nonstationary effects in adaptive filtering

Recursive least squares with linear constraints

Bounds on tracking errors for adaptive signal processing algorithms

Sufficient excitation and stable reduced-order adaptive IIR filtering

On adaptive IIR filters and parallel adaptive identifiers with adaptive error filtering

A comparison of adaptive algorithms used for designing phase compensation filters

An adaptive interference canceller using Kalman filtering

Frequency domain non-linear adaptive filter

The application of SHARF to adaptive removal of TV ghosting

Correction algorithms for extended range echo cancellers

Processing techniques for the inspection of offshore structures

An architecture for an unattended radar signal processor

Recursive linear prediction for clock synchronization

Synthesis of timbral families by warped linear prediction

A microprogrammed signal processor

Perceptual issues in digital processing of music

Modulation systems for digital audio discs with optical readout

A modular signal processor for digital filtering and dynamic range control of high quality audio signals

A universal, digital sampling frequency converter for digital audio

Split-band APC System for low bit-rate encoding of speech

Mediumband speech encoding using time domain harmonic scaling and adaptive residual coding

Real-time speech coder implementation at 9.6 and 16 kb/s

Time and frequency domain noise shaping in speech coding

Noisy-channel performance of 16 kb/s APC coders

Comparative study of 16 kbps adaptive predictive coders

A novel adaptive predictor DPCM system for speech signals

Stability analysis of APC systems

The design of a 16 kb/s split-band adaptive predictive coder for noisy channels

Considerations in the implementation of an adaptive transform coder for speech in a 16-bit fixed point arithmetic microprocessor

Efficient coding of the prediction residual

An LSI chip set for DSP hardware implementation

A VLSI speech analysis chip set based on square root normalized ladder forms

Realtime spectrum analysis using a microprocessor peripheral

Architecture for a digital programmable image processing element

Single-board digital signal processor

A feasibility study of microprocessor-based digital filters

Fourier transform algorithm implementations on a general-purpose microprocessor

Control of multiprocessor SPS-1000 configurations using principles of data-flow architecture

Fast convolution using modularized multiprocessor architectures

On the hardware implementation of a multi-processor environment for several DSP applications

Design of 2-D IIR digital filters with circular symmetry by transformation of the variable

2-D separable digital filters

Design of two-dimensional digital filters using analog reference filters without second kind singularities

Design of 2-D rational digital filters

2-D direct form, recursive filter design with magnitude and phase specifications

A method for the design of stable (N-D) analog and digital filters

Extensions of 2-D iterative digital filters

Two-dimensional hexagonal digital recursive filters

An l2-stability theorem for multidimensional IIR digital filters

An optimization algorithm for determining the endpoints of isolated utterances

Isolated word recognition using a two-pass pattern recognition approach

What are the significant variables in dynamic programming for discrete utterance recognition?

Variations on Itakura's spectral match score

Fast nonlinear time alignment for isolated word recognition

A composite source model for speaker and isolated word recognition

Microprocessor implementation of an LPC-based isolated word recognizer

Optical processing techniques for real-time isolated word recognition

A formalization of performance specifications for discrete utterance recognition systems

Evaluating the performance of speech recognisers at the acoustic-phonetic level

The ASSP sound reinforcement systems

Cochlear modeling - 1980

Echoes, reverberation, speech intelligibility and musical performance

Critical band analysis-synthesis

Time-domain measurement of vented-box loudspeaker system parameters

Digital simulation of loudspeaker systems

Second-order gradient noise-cancelling microphone

Phase equalization for magnetic recording

The effects of primitive size on the overall design of real-time digital audio signal processors

Tubes versus transistors in electric guitar amplifiers

Medium band speech coding using a dictionary of waveforms

An experimental study of sub-band coder design incorporating recursive quadrature filters and optimum ADPCM

Using the prediction residual to improve LPC synthesis for 9600 bps applications

Practical considerations for variable length source coding

Mediumband speech processor with baseband residual spectrum encoding

A DFT-based residual-excited linear predictive coder (RELP) for 4.8 and 9.6kb/s

An adaptive delta modulator with noise spectral shaping

Speech coding using modulo-PCM with side information

Bounds on performance and dynamic bit allocation for sub-band coders

Enhanced CVSD--An Embedded speech coder for 64-16 kbps

Wideband quality speech encoders with bit rates of 16-32 kbits/s.

Performance of a 32 kbit/s ADPCM coder for digital long-haul telephone transmission

Fast least squares FIR system identification

Hilbert space array methods for finite rank process estimation and ladder realizations for adaptive signal processing

Schur and Levinson algorithms for nonstationary processes

A modified lattice algorithm for deconvolving filtered impulsive processes

Computer simulation studies of a simplified model for a certain class of autoregressive systems

A time-recursive lattice algorithm for autoregressive estimation

Square-root covariance ladder algorithms

Linear prediction in the singular case and the stability of eigen models

Use of the fast Kalman estimation algorithm for adaptive system identification

A new recursive method for identification of multivariate linear systems from impulse response and output covariance information

Automatic order selection in the design of IIR filters

Rational modeling by pencil-of-functions method

Error smoothing in adaptive LMS algorithms

Connected word recognition using a level building dynamic time warping algorithm

Connected word recognition using a syntax-directed dynamic programming temporal alignment procedure

A segmentation procedure for connected word recognition based on estimation principles

Comparison of several clustering algorithms for data rate compression of LPC parameters

A preliminary study on the use of demisyllables in automatic speech recognition

The efficiency of demisyllable segmentation in the recognition of spoken words

Experiments on an isolated-word recognition system for multiple speakers

An evaluation of hierarchical features in speech recognition

Symmetry in two-dimensional rectangularly-sampled digital filters

A new stabilization technique for 2-dimensional recursive digital filters

A technique for coefficient minimum word length estimation sufficient for stability maintenance in N-D filters

Implementation of two-dimensional semicausal recursive digital filters

A new algorithm for one-dimensional and two-dimensional maximum entropy power spectrum estimation

Evaluation of two-dimensional discrete Fourier transforms via generalized FFT algorithms

An interpolation technique for periodically sampled two-dimensional signals

Multidimensional signal reconstruction from phase or magnitude

Thematic filtering and scene interpretation based on complete invariant systems of features

Movement and deformation prediction for pluvial distributions

Evaluation procedure for environmental acoustic models

Accurate modeling for shallow water wave propagation

Effects of input power fluctuations on passive sonar receiver performance

Conventional and binary time series models of sonar fluctuations

Synthetic multichannel time series for simulating underwater acoustic noise

A new model of the piezoelectric ultrasonic transducer

A novel beamformer for circular sonar arrays

Performance studies of adaptive lattice prediction--Error filters for target detection in a radar environment using real data

Seismic event detection using adaptive predictors

Analytic signal representation of the synthetic seismogram of bright spots

The effects of microphones and facemasks on LPC vocoder performance

Operation of LPC vocoders in noisy environments

Study of echo cancelling algorithms for full duplex telephone networks with vocoders

Low complexity robust linear predictive coding of speech signals

Improving LPC analysis of noisy speech by autocorrelation subtraction method

Acoustic noise suppression in the context of a perceptual model

A technique for perceptually reducing periodically structured noise in speech

Autoregressive models for noisy speech signals

A system for helium speech enhancement using the short-time Fourier transform

Segmental preclassification for improved objective speech quality measures

Improved spectral subtraction for signal restoration

Sequential image enhancement using 2-D adaptive estimation

Application of two dimensional spectral estimation in image restoration

Adaptive filtering for image enhancement

Hybrid DFT/DPCM interframe image quantization

Application of fuzzy set theory in detecting x-ray edges

A new operator for edge detection

Objective quality measures for the design of digital image transmission systems

Three-dimensional motion estimation from image-space shifts

A two-dimensional finite field processor for image filtering

New sensor geometries for image processing: Computer vision in the polar exponential grid

Continuous parameter acoustic processing for recognition of a natural speech corpus

Continuous speech recognition with automatically selected acoustic prototypes obtained by either bootstrapping or clustering

Automatic labelling of continuous speech with a given phonetic transcription using dynamic programming algorithms

Improvement and comparison of three phonemic segmentation methods of speech

Acoustic processing in the conversational speech recognition system

Speech recognition of a natural text read as isolated words

Vowel identification in continuous speech using articulatory parameters

A parallel architecture for speech understanding

Limited wordlength and FDCT processing accuracy

Fixed-point error analysis of Winograd basic DFT algorithms considering correlation between noise sources

Narrowband filtering with frequency sampling filters

Computational and performance aspects of minimax equalizers

Permuted difference coefficient digital filters

Multiplicative linear phase FIR filters

FIR filter structures having low sensitivity and roundoff noise

A unified analysis of roundoff noise reduction in digital filters

Comparison of the effects of quantization on digital filters

Transfer functions available with a new second order digital filter structure

On the overflow problem in a second order digital filter

On limit cycle amplitudes in error-feedback digital filters

Determining exact maximum amplitude limit cycles in digital filters

A high-speed algorithm for mapping 2-D ambiguity surfaces

Generalized wideband crossambiguity function

Distribution of the sample magnitude-squared coherence obtained using overlapped Fourier transforms

Improvement of the imaging of moving acoustic sources by the knowledge of their motion

A comparison of algorithms for adaptive estimation of the time delay between sampled signals

A method for generating a class of time-delayed signals

Some considerations of time delay estimation

Effects of signal and noise spectral slopes on time delay estimates in passive localization

The estimate of time delay between two signals with random relative phase shift

Speaker adaptation for phoneme recognition

Robust arrival time determination of seismic waves in noise

Comparison of detection performance of a time domain and frequency domain adaptive algorithm

A comparison of the effects of component nonidealities on the performance of analog and digital LMS adaptive noise cancellers

Efficient spectral matching of the LPC residual signal

ICASSP '80. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '79. IEEE International Conference on Acoustics, Speech, and Signal Processing

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

Performance evaluation of a new robust detector for sonar signals

Content Provider	IEEE Xplore Digital Library
Author	Vyas, A. Indiresan, P.
Copyright Year	1981
Description	Author affiliation: CARE, Indian Institute of Technology, New Delhi (Vyas, A.)
Abstract	A novel Spectrum Balancing Detector (SBD) for sonar signals is introduced. The detector operates on quadrature signals as well as their Hilbert transforms using cross multiplicative structure in contrast to the square law detector which simply adds up the two components after a squaring operation. It has the special feature of forming an automatic notch around reverberation and indicating the sense of direction of motion of the target. The performance of the SB detector is evaluated on the basis of minimum detectable level of signals and it is shown that the detector is at least as robust as square law detector. Theoretical and experimental results are also presented to establish the properties of the SB detector.
Starting Page	171
Ending Page	175
File Size	134506
Page Count	5
File Format	PDF
DOI	10.1109/ICASSP.1981.1171281
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1981-03-30
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Sonar detection Robustness Detectors Signal detection Signal processing Frequency Doppler shift Equations Motion detection Silicon compounds
Content Type	Text
Resource Type	Article

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