NDLI: Maximum a posteriori array signal processing

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2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2013 IEEE International Conference on Acoustics, Speech and Signal Processing

2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2011 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2010 IEEE International Conference on Acoustics, Speech and Signal Processing

2009 IEEE International Conference on Acoustics, Speech and Signal Processing

2008 IEEE International Conference on Acoustics, Speech and Signal Processing

2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07

2006 IEEE International Conference on Acoustics Speech and Signal Processing Proceedings

Proceedings. (ICASSP '05). IEEE International Conference on Acoustics, Speech, and Signal Processing, 2005.

2004 IEEE International Conference on Acoustics, Speech, and Signal Processing

2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03).

2002 IEEE International Conference on Acoustics, Speech, and Signal Processing

2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)

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

Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing

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

International Conference on Acoustics, Speech, and Signal Processing,

ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '85. IEEE International Conference on Acoustics, Speech, and Signal Processing

Volume 10 - 1985

Proceedings ICASSP '85. IEEE International Conference on Acoustics, Speech, and Signal Processing [Front cover and table of contents]

Isolated word recognition using hidden Markov models

Explicit modelling of state occupancy in hidden Markov models for automatic speech recognition

Recent developments in the application of hidden Markov models to speaker-independent isolated word recognition

Training of HMM recognizers by simulated annealing

Experiments on speaker-independent recognition of hand-segmented French vowels

Discrete utterance speech recognition without time alignment

Comparative study of several distortion measures for speech recognition

Applying matrix quantization to isolated word recognition

Spoken word recognition based on top-down phoneme segmentation

On associative recognition of isolated Chinese word

New design methods for FIR filters with equiripple stopbands and prescribed degrees of passband flatness

Efficient FIR fillter design using differential coding of filter coefficients

Smoothed median filters with FIR substructures

Residue feedback in ladder and lattice filter structures

A new S-Z replacement design technique of IIR digital filters

An approach for designing systems with prescribed behaviour at distinct frequencies regarding additional constraints

Design of linear-phase partly digital anti-aliasing filters

Wave digital filters: Voltage, current, or power waves?

A novel approach to the design of filters for filter banks

Cross spectrum ML estimate

Adptive spectral estimation by the conjugate gradient method

Window functions obtained from B-S

Spectrum estimation of time series with missing data

Spectral estimation statistics for noise corrupted autoregressive series--First-order case

Spectral estimation via minimum energy correlation extension

Superresolution autoregressive spectral estimation technique using multiple step prediction

Adaptive methods for real time Pisarenko spectrum estimate

Adaptive recursive scheme for spectral analysis of sinusoids in signals with unknown colored spectrum

Hierarchical predictive approach to image coding

Ordering predictive coding of digital image

A hybrid image coding technique using a noncausal stochastic model

Use of vector quantizers in image coding

Compression of color digital images using vector quantization in product codes

Adaptive vector quantization by progressive codevector replacement

A fast matrix quantizer for image encoding

A matrix quantizer incorporating the human visual model

Adaptive transform tree coding of images

A hybrid image compression technique

Space-time processing of multisensor time delay data in bearing estimation

Robust adaptive Kalman filtering for systems with unknown step inputs and non-Gaussian measurement errors

Time delay estimation in presence of jitter

Application of robust filtering techniques to time delay estimation in the Arctic environment

Time delay estimation for spread spectrum signals using generalized cross correlation

Time-delay-of-arrival resolution of multipath communication signals

1-D and 2-D syntactic pattern recognition for the detection of bright spots

Predictive deconvolution of seismic array data for inversion

Linear time-invariant space-variant filters and the WKB approximation

Image enhancement using coherence processing with applications to seismograms

A VLSI implementation of an FFT/NTT computational unit

A 32 bit 15M flop floating point programmable signal processor architecture for VLSI implementation

Design and VLSI implementation of a concurrent solver for N coupled least-squares fitting problems

An 18-bit floating-point signal processor VLSI with an on-chip 512W dual-port RAM

Very fast sequencing and data addressing made easy with new CMOS VLSI components

New CMOS chip family facilitates design of high speed DSP hardware

Application of an advanced signal processing engine to adaptive techniques

A custom-designed integrated circuit for the realization of residue number digital filters

High resolution digital filter chip

Architecture and applications of a second-generation digital signal processor

A systolic array computer

The waveform segment vocoder: A new approach for very-low-rate speech coding

A low bit rate segment vocoder based on line spectrum pairs

Application of line-spectrum pairs to low-bit-rate speech encoders

A multiple rate low rate voice codec

Vector quantization and perceptual criteria for low-rate coding of speech

Generalization of the multipulse coding for low bit rate coding purposes: The generalized decimation

Pole-zero multipulse speech representation using harmonic modelling in the frequency domain

Optimal initial conditions and pulse values for multipulse speech coding

Bit rate reduction by Markov-Huffman coding of speech parameters

A multi-processor cellular automaton chip

Towards an FIR filter tissue

A reconfigurable systolic primitive processor for signal processing

VLSI Architecture for solving covariance eigen system

Hierarchical flowgraph integration for VLSI array processors

Design strategies for implementing systolic and wavefront arrays using OCCAM

A practical comparison of the systolic and wavefront array processing architectures

Systematic approaches to the design of algorithmically specified systolic arrays

L1-Norm versus L2-Norm minimization in parametric spectral analysis: A general discussion

An adaptive orthogonal maximum likelihood algorithm for parameter estimation

Solving a class of nonlinear least squares problems

Non-recursive frequency estimation for closely spaced sinusoids

Simple, effective computation of principal eigenvectors and their eigenvalues and application to high-resolution estimation of frequencies

Approximate factorization of unfactorable spectral models

Efficient generation of ARMA cross convariance sequences

Signal detection using autoregressive parameters

Adaptive comb filtering for harmonic signal enhancement

Adptive vector quantization for image sequence coding

An improved transform coder for image sequences using attributes of difference pictures

Motion-compensated interframe coding

Transform/Time domain coding using the method of projection onto convex sets

A minimum risk quantizer for motion compensated image coding

Block-recursive matching algorithm (BRMA) for displacement estimation of viedo images

Motion-compensated adaptive intra-interframe predictive coding algorithm

Motion-compensated hybrid coding at 50 kb/s

A new motion-compensated transform coding scheme

Implicit motion compensated noise reduction of motion video scenes

Investigation of text-independent speaker indentification over telephone channels

Automatic speaker recognition using vocoded speech

A vector quantization approach to speaker recognition

Text-dependent speaker recognition using vector quantization

Periodicity estimation by hypothesis-directed search

A speaker verification system for access-control

A robust realtime pitch extraction from the ACF of LPC residual error signals

Improved pitch detection algorithm for noisy speech

New objective measures for the evaluation of pitch extractors

Design and evaluation of double-transform pitch determination algorithms with nonlinear distortion in the frequency domain-preliminary results

Objective estimation of perceptually specific subjective qualities

Quantification of dysphony with allowance for inter-utterance variation

A computational model of the cochlea used with cochlear prosthesis patients

The effects of controlled speech level input on the intelligibility testing of speech compression algorithms

Iterative deconvolution in noncoherent systems

Deconvolution in the sequency domain

Deconvolution without system model or a new blind deconvolution

Deconvolution by the conjugate gradient method

Time and frequency selective deconvolution uising optimal control

L1deconvolution and its application to seismic signal processing

Signal deconvolution using fuzzy sets

On absolute value minimization approaches to tauberian modelling

On descrete band-limited signal extrapolation

A comparison of three methods for discrete-time signal extrapolation

Robust LPC analysis of speech by extended correlation matching

Perceptual weightings and optimal pulse positioning in multipulse LPC speech coding

All-pole speech modeling with a maximally pulse-like residual

LPC Speech analysis using the L1norm

Speech transformations based on a sinusoidal representation

High quality time-scale modification for speech

Speech signal analysis and synthesis via Fourier-Bessel representation

Speech analysis and restitution using time-depedent autoregressive models

A measure of in-synchrony regions in the auditory nerve firing patterns as a basis for speech vocoding

Perceptually based linear predictive analysis of speech

A new model-based speech analysis/Synthesis system

Statistical design of analysis/Synthesis systems with quantization

A unifying framework for analysis/Synthesis systems based on maximally decimated filter banks

Quadrature mirror filters with perfect reconstruction and reduced computational complexity

Complementary IIR digital filter banks

A filter/Detector interpretation of the short-time Fourier transform magnitude

Design of filters for discrete short-time Fourier transform synthesis

Flexible design of computationaly efficient nearly perfect QMF filter banks

Real-time telephone channel simulation

Properties of the positive time-frequency distribution functions

Properties of eigenanalysis methods for bearing estimation algorithms

Extending the threshold of the eigenstructure methods

High resolution spectral method for spatial discrimination of closely spaced correlated sources

On beamforming in presence of multipath

Normal mode propagation and high resolution methods

On the probability density of signal-to-noise ratio in an improved adaptive detector

High resolution bearing estimation without eigen decomposition

Observation impulse response by two relatively prime pseudorandom sequences

Catholic electroacoustical difficulties

Omnidirectional coded planar loudspeaker arrays

Design and optimization of a digital FM receiver using DPLL techniques

Implementation of a real-time, digital vocoder for tactile hearing prostheses

Signal processing architecture for loudspeaker array directivity control

Reconstruction of tonal sequences

A new auditory model for the evaluation of sound quality of audio systems

The maximum tolerable delay of speech and music

Bounds for ARMA spectral analysis based on sample covariances

The role of spectral decomposition in the pattern recognition of narrowband signals

Perturbation analysis of a SVD based method for the harmonic retrieval problem

Statistical properties of coherence and power contribution ratios via multivariate autoregressive modeling

Event detection using recursively updated lattice filters

Sensitivity and performance analysis of coherent signal-subspace processing for multiple wideband sources

Direction of arrival estimation by eigenstructure methods with unknown sensor gain and phase

Detection algorithms based on prediction error - additive noise - data orthogonality

On nonlinear estimation in presence of non-Gaussian Noise

A nonrecursive filter for edge preserving image restoration

Identification of image and blur parameters for the restoration of noncausal blurs

Image restoration using a parallel indentification and filtering procedure

Optimal techniques for constraint based signal restoration and image reconstruction

The Wilcoxon filter: A robust filtering scheme

Edge preserving signal enhancement using generalizations of order statistic filtering

Constrained optimization for image restoration using nonlinear programming

Post-reconstruction correction in quantitative single photon ECT

Adaptive noise cancelling applied to image restoration

Hierarchical approach to image estimation

Boundary value problem in image restoration

Nonstationary iterative image restoration

A general formulation of constrained iterative restoration algorithms

Adaptive least-squares technique for multi-band image enhancement

A digital filtering system for decoding in-line holograms

Noise-immune speech transduction using multiple sensors

Adaptive noise reduction in aircraft communication systems

Processing of noisy speech using group delay functions

Cross-channel correlation for the enhancement of noisy speech

Adaptive reduction of interfering speaker noise using the least mean squares algorithm

Bandwidth design for speech-seeking microphone arrays

Synthesis by rule: LPC diphones and calculation of formant trajectories

Diphone synthesis using multipulse coding and a phase vecoder

GRAPHON-the Vienna speech systhesis system for arbitrary German text

Voice conversion: Factors responsible for quality

Concatenation rules for demisyllable speech synthesis

Optimum binary windows for discrete Fourier transform

Exact fast digital convolution by using P-adic numbers and polynomial transformations

Complex digital signal processing using quadratic residue number systems

Efficient architectures for implementing the prime-factor Fourier transform modules

Prime factor decomposition of the discrete cosine transform and its hardware realization

DIF Algorithms for DCT and DST

Multiply/Add tradeoffs in $length-2^{n}FFT$ algorithms

Implementation of "Split-radix" FFT algorithms for complex, real, and real symmetric data

The chord property speeds finite field FFTs

Sufficient codition for extendibility and two-dimensional power spectrum estimation

On 2-D prony methods

The optimum approach to multichannel AR spectrum estimation

Simultaneous confidence bands for a class of 2-D spectral estimates

Frequency-wavenumber spectral estimation of non-planar random fields

Phase errors in the cross spectrum estimate due to misalignment

Multidimensional maximum-entropy covariance extension

Resolution property of the improved maximum likelihood method

Under-ice reverberation rejection

Application of a super resolution algorithm to acoustic tomography multipath data

Bayesian estimation of medium properties in wavefield inversion techniques

Inversion of modified beamformed array data

Identification of the modulation type of a signal

Speaker independent telephone speech recognition

Evaluation of a network-based isolated digit recognizer using the TI multi-dialect database

Isolated word recognition using a segmental approach

Phonetically guided clustering for isolated word recognition

A real-time, isolated-word, speech recognition system for dictation transcription

A coarse phonetic knowledge source for template independent large vocabulary word recognition

Large vocabulary speaker-independent Japanese speech recognition system

Recognition error measurements from parameterized distance distributions

An efficient vector-quantization preprocessor for speaker independent isolated word recognition

A new algorithm for isolated word recognition with large within-class variability

Speech recognition in the F-16 cockpit using principal spectral components

Hierarchical DP for word recognition

Knowledge base enhancement of visual tracking

Boundary detection in speckle images

Adaptive hierarchical algorithm for accurate image segmentation

Segmentation of multilevel images using Gibbs distribution

Fast object segmentation in textured backgrouds

Knowledge based object detection

A new approach to parameter estimation for Gibbs random fields

3-D Motion parameters from contours using a canonic differential

Determination of affine transforms from object contours with no point correspondence information

Segmentation of three dimensional range data

Two view motion analysis under a small perturbation

A silhouette-slice theorem for opaque 3-D objects

Code-excited linear prediction(CELP): High-quality speech at very low bit rates

A stochastic model excitation source for linear prediction speech analysis-synthesis

Mid-rate coding based on a sinusoidal representation of speech

A new approach to multipulse LPC coder design

A linear programming approach to multipulse speech coding

Efficient algorithms for obtaining multipulse excitation for LPC coders

A speech coding method using thinned-out residual

Regular excitation reduction for effective and efficient LP-coding of speech

An efficient, pitch-aligned high-frequency regeneration technique for RELP vocoders

The transmission of in-band signaling for medium band voice codec implementations

A systolic array for linearly constrained least squares filtering

Processing speech with the multi-serial signal processor

A VLSI dynamic time warp processor for connected and isolated word speech recognition

A multi-pulse LPC synthesizer for telecommunications use

Image processing address generator chip

VLSI Implementation of two-dimensional generalized mean filter

VLSI Architecture for a one chip video median filter

A two-dimensional digital filter chip set for modular two-dimensional filter implementation

Systolic architectures based on barrel shifters for real-time signal and image processing

Adaptive restoration of unknown samples in certain time-discrete signals

On the optimality of the Wigner distribution for detection

An expansion of Wigner distribution and its applications

Wigner-Ville and evolutionary spectra for covariance equivalent nonstationary random processes

Transformation, inversion and conversion of bilinear signal representations

Generalized ambiguity functions

Signal synthesis from Wigner distribution

A generalized delay operator for shift-variant systems

Time and spatial varying CAM and AI signal analysis using the Wigner distribution

Autoregressive models with time-dependent log area ratios

A method of image reconstruction in fan beam tomography

Reconstruction of two-dimensional signals from threshold crossings

Incorporation of a priori moment constraints into signal recovery and synthesis problems via the method of convex projections

On the stability and sensitivity of multidimensional signal reconstruction from Fourier transform magnitude

Nearest neighbor and generalized inverse distance interpolation for Fourier domain image reconstruction

An inner product framework for image reconstruction

Image reconstruction of the heart using a priori information and spatiotemporal estimation

An image reconstruction from incomplete observation by constrained spectrum extrapolation

Aerodynamic aspects of voicing: Glottal pulse skewing revisited

Variability in closed phase analysis of speech

Measuring source-tract interaction from speech

Analysis of vocal tract lip reflectance by linear prediction

Determining vocal tract shape by applying dynamic constraints

A new articulatory model for speech production

Speech processing by a model of the auditory periphery

Formant tracking using hidden Markov models

A robust formant-based speech spectrum comparison measure

Instantaneous-frequency distribution vs. time: An interpretation of the phase structure of speech

Cubic spline modeling of speech spectra

A compensated-Kalman speech parameter estimator

Spectrum and pitch estimation of speech using a time-varying ARMA estimation algorithm

Bayesian deconvolution of speech containing pulsed excitation

Acoustic transmission-line analysis of formants in hyperbaric Helium speech

Steady-state behavior of RLS adaptive algorithms

Fast recursive least-squares algorithms: Preventing divergence

An efficient step size adaptation technique for LMS adaptive filters

Convergence rates for the constant modulus algorithm with sinusoidal inputs

Global convergence of the constant modulus algorithm

Noise capture properties of the constant modulus algorithm

Identification of sparse impulse response systems using an adaptive delay filter

"Noise cancellation studies using a least-squares lattice filter"

A generalized filter structure for IIR adaptive filters

An efficient algorithm for lattice filter/Predictor

A minimal parameter adaptive notch filter with constrained poles and zeros

The unbiased least-squares lattice

Detection and identification of sinusoids in broadband noise via a parallel recursive ALE

Network-based connected digit recognition using vector quantization

Application of a sequential pattern learning system to connected speech recognition

Context-dependent modeling for acoustic-phonetic recognition of continuous speech

A script-guided algorithm for the automatic segmentation of continuous speech

Speaker dependent connected speech recognition via phonetic Markov models

Integration of segmenting and nonsegmenting approaches in continuous speech recognition

Continuous speech recognition on the bases of vector field model for segmentation and feature extraction, and continuous dynamic programming for pattern matching

Multi-speaker computer recognition of ten connectedly spoken letters

A connected speech recognition system using a diphone-based language model

Keyword recognition using template concatenation

Parallel algorithms for Toeplitz matrix operations

The block-processing FTF adaptive algorithm

Data echo nonlinear cancellation

Equalization for transmission line channels: A discussion of three IIR adaptive filtering algorithms

Some experimental and theoretical results using a new adaptive filter structure for noise cancellation in the presence of crosstalk

Adaptive digital filtering of differentially coded signals

Adaptive systems as optimal processors

Broadband detection of signals with unknown spectra

Adaptive detection of transient signals

Automatic recognition of underwater transient signals - a review

Thresholds in frequency estimation

Detection of a sinusoidal signal in the presence of directional interference

Robust scale invariant detection of coherent narrowband signals in nearly Gaussian noise

Two-dimensional arrays with nonlinear elements

Microscopic correlation signals

Sonar detection in Weibull bottom reverberation

A fast maximum-likelihood estimation and detection algorithm for Bernoulli-Gaussian processes

Parallel block realization of 2-D IIR digital filters

Conjugate transformations preserving the general overflow-stability property of 2-D digital filters

Separating Y, I, Q components of NTSC composite signal

Hadamard transform seperation of NTSC component signals

Lattice-filter modeling of two-dimensional fields

Application of symmetrical decomposition to 2-D FIR filter design

Two-dimensional orthogonal median filters and applications

A unification of linear, median, order-statistics and morphological filters under mathematical morphology

Frequency domain analysis of two-pass rotation algorithm

Constrained prediction

Rational parametric coherence estimation via convolved correlations

Spectrum model-order determination via significant reflection coefficients

Estimation of sequences in a signal class determined from the data

Bispectrum estimation for short length data

Explicit formulas for super-resolution

A new super-resolution spectral estimation technique using staggered PRFs

A comparison of algorithms for polar-to-cartesian interpolation in spotlight mode SAR

A block adaptive approach for clutter suppression

Comparing the radar clutter-suppression performances of lattice prediction error filters using three variations of Burg's algorithm

Programming techniques for $PA^{3}architectures$

A novel architecture design for VLSI implementation of an FIR decimation filter

Cyclo-static multiprocessor scheduling for the optimal realization of shift-invariant flow graphs

The VLSI design of a single chip for the multiplication of integers modulo a fermat number

VLSI Implementation of a linear systolic array

VLSI Implementation of a fast rank order filtering algorithm

An oversampling A-to-D converter structure for VLSI digital codec's

A VLSI design of a pipeline Reed-Solomon decoder

2-D Array processor having a controlled pipelined architecture for elliptical sparse matrices

The application of multi-dimensional access memories to ultra high performance signal processing systems

Signal processing software for a voice messaging system using the TMS32010 processor

32kbs ADPCM/PCM transcoder using TI-320 DSP microprocessor

A CCITT standard 32 kbps ADPCM LSI codec

A multi-pulse LPC speech codec using digital signal processors

Subband coding with silence detection

Real-time speech compression with a VLSI vector quantization processor

On a delta modulation based real time autocorrelator

Single chip implementation of feature measurement for LPC-based speech recognition

Simulation of a highly parallel system for word recognition

Time-power-area tradeoffs for the nMOS VLSI full-adder

An efficient VLSI adder for DSP architectures based on RNS

The systematic exploration of pipelined array multiplier performance

CAD Tools for the optimized design of custom VLSI wave digital filters

A systolic implementation of the Winograd Fourier transform algorithm

VLSI Gate array prime radix Fourier transform processor

FFT Drithmetic element built on VLSI high performance gate array

A proposal for very high performance FFT processor architectures

Bounds on the envelopes of the response of systems to bandlimited inputs

Confidence regions for perturbed singular values in system identification

Orthogonality of oblique projections and lattice-form models

The effects of bandwidth MIS-estimation in bandlimited signal extrapolation

Resolving power of signal subspace methods for finite data lengths

A two stage approach for adaptive prediction of ARMA processes

An extension of the fast algorithm for linear phase system identification

A fuzzy associative memory module and its application to signal processing

On the computation and use of projections of digital images in general purpose image processing pipeline architectures

A fast sort-selection filter chip with effectively linear hardware complexity

A near-neighbor processor for line thinning

Two-dimensional digital filters with minimum cycle time

Vector-radix algorithm for a 2-D discrete Hartley transform

Solution to the indexing problem of multidimensional DFT's on arbitrary sampling lattices

Fast 2-D discrete cosine transform

A Hankel transform algorithm for uniformly sampled data

Computation of the two-dimensional Fourier transform of circularly symmetric functions

Lexical stress determination and its application to large vocabulary speech recognition

Speaker sampling for enhanced diversity

A study on vowel behaviour and its description by a statistical model

Collection of phoneme samples using time alignment and spectral stationarity of speech signals

A frame language for the control of phonetic decoding in continuous speech recognition

Detection of nasalized vowels in American English

A reference speech recognition algorithm for benchmarking and speech data base analysis

Probabilistic grammar for phonetic to French transcription

Some acoustic-phonetic correlates of speech produced in noise

Speaker-independent connected Japanese digit recognition based on phonetic approach

Voiced/Unvoiced decision based on the bispectrum

Identification of unaspirated plosives using integrated temporal and spectral features in dynamic representation as acoustic cues

Matching algorithms between a phonetic lattice and two types of templates -- Lattice and graph

An investigation into the efficiency of a parallel TMS320 architecture: DFT and speech filterbank applications

An implementation of the LMS adaptive filter using an SIMD multiprocessor ring architecture

A multimicroprocessor system with distributed common memory for real-time digital correlation and spectrum analysis

A multi-processor structure for signal processing application to acoustic echo cancellation

Distributed arithmetic implementation of nonlinear echo cancellers

High throughput signal processing system

$GOPS^{TM}Gigital$ signal processor provides 20 MHz analog bandwidth

Effective multifrequency receiver design

Microprocessor based 9600 bps modem

Simulation of signal flow graphs for signal processing systems

Software techniques for programming a general purpose data flow signal processor

A transportable TMS32010 signal processing system

Structural considerations for large FFT programs on the TI TMS 32010 DSP microchip

Concurrent process structured software models for time domain harmonic scaling of speech

An analysis of algorithms for microprocessor implementation of high-speed data modems

Interpretation-guided signal processing via protocol analysis

An SSIMD compiler for the implementation of linear shift-invariant flow graphs

A resonant section filter design method for optimized coefficients in add-and-shift architectures

Subband coding of speech using backward adaptive prediction and bit allocation

A new dynamic bit allocation scheme for sub-band coding

Multirate sub-band coding applied to digital speech interpolation

Dual adaptive delta modulation for mobile voice channel and its DSP implementation

A study of three coders (sub-band, RELP and MPE) for speech with additive white noise

Critical point coding: Design and real-time implementation at 16 kbps

Efficient codebook allocation for an arbitrary set of vector quantizers

Speech and speaker independent codebook design in VQ coding schemes

Embedded coding of speech: A vector quantization approach

Contour vector quantization and waveform coding

Error analysis of linear recursions in floating point

A study of coefficient quantization errors in state space digital filters

On quantization effects in state-variable filter implementations

Roundoff noise in estimation of the signal random parameter

Multiplier-and memory-based digital FIR filters for nyquist channels: An assessment of quantization noise

Novel approximations $to0Σ^{∞}|h(n)|$ for second order digital filters

Quantization noise effects in the complex LMS adaptive algorithm-linearization using dither

New forms of LS lattice algorithms and an analysis of their round-off error characteristics

Adaptive filter improvement using randomly quantized coefficients

Maximum a posteriori array signal processing

Time delay determination: Maximum likelihood and Kalman-Bucy type structures

Lower bounds on the worst case probability of large error for two channel time delay estimation

Source location with arrays subject to travelling wave perturbations

Optimal multiple source location estimation via the EM algorithm

Application of the LMS adaptive line enhancer in time delay estimation

A hybrid parallel-serial approach to nonlinear filtering

A fast local maximum likelihood estimator for time delay estimation

Estimation of frequencies of multiple two-dimensional sinusoids: Improved methods of linear prediction

Multiple beamformer performance analysis of the coherent mode

Bearing accuracy and resolution bounds of high-resolution beamformers

Bearing estimation in the presence of unknown correlated noise

Adaptive beam forming in correlated interference environment

Minimum rejection response array filters in the presence of white noise

A new set of linear constraints for broadband time domain element space processors

Analysis of constrained LMS algorithm with application to adaptive beamforming using perturbation sequences

Error analysis of eigenvector preprocessors used in adaptive beamforming

Finite word length effects on array processing algorithms

A new approach to array geometry for improved spatial spectrum estimation

Sensitivity constrained optimum endfire array gain

Exact maximum likelihood estimation of superimposed exponential signals in noise

Designing filters for interpolation beamforming

Cepstral analysis of electroacoustic transducers

Generalized spline signal interpolation and smoothing with simultaneous optimization in time and frequency domains

Word boundary detection and speech recognition of noisy speech by means of iterative noise cancellation techniques

A new syntactic/Semantic approach to 3D-surface reconstruction

Three dimensional shape estimation from two dimensional images

Custom data-flow machines for speech recognition

A control point theory for boundary representation and matching

Tree structures for implementation of a vector quantized speech coding system

A new look on the parallel implementation of the Shur algorithm for the solution of Toeplitz equations

ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '83. IEEE International Conference on Acoustics, Speech, and Signal Processing

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

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

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

Maximum a posteriori array signal processing

Content Provider	IEEE Xplore Digital Library
Author	Owsley, N.
Copyright Year	1985
Description	Author affiliation: Naval Underwater Systems Center, New London, Connecticut (Owsley, N.)
Abstract	A noncausal estimator is presented for the position versus time sequence of a source relative to an array of passive receivers. The position is estimated in terms of the bearing and range of the source which are modelled as second order Gauss-Markov discrete time sequences. A maximum a posteriori estimation procedure is developed which provides for an exact nonlinear doppler compensated estimator of the position vector for a batch of received sensor data. The result is a delayed, smoothed position vector for the batch as opposed to a casual estimation filter output sequence. The signal spectrum is also estimated for detection purposes.
Starting Page	1747
Ending Page	1749
File Size	39501
Page Count	3
File Format	PDF
DOI	10.1109/ICASSP.1985.1168195
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1985-04-26
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Array signal processing Sensor arrays Gaussian processes Delay estimation Motion estimation Phased arrays Maximum a posteriori estimation Filters Maximum likelihood detection Maximum likelihood estimation
Content Type	Text
Resource Type	Article

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