NDLI: Signal subspace analysis and improvement of spectral estimation algorithms

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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

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

Volume 9 - 1984

Proceedingss ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing [front cover and table of contents]

Speaker normalization algorithms for very-low-rate speech coding

A vector quantizer incorporating both LPC shape and energy

Speaker-independent connected digit recognition

Low bit rate speech coding by concatenation of sound units and prosody coding

Short-cut algorithms for the learning subspace method

Automatic diphone bootstrapping for speaker-adaptive continuous speech recognition

Improving performance of multi-pulse LPC coders at low bit rates

On the use of transient information in speech recognition

Application of allophonic and lexical constraints in continuous digit recognition

Efficient algorithm for multi-pulse LPC analysis of speech

Special feature vector coding and appropriate distance definition developed for a speech recognition system

A directory listing retrieval system based on connected letter recognition

Low bit rate speech enhancement using a new method of multiple impulse excitation

Performance of isolated word recognition system for confusable vocabulary

Syntax driven recognition of connected words by Markov models

A glottal LPC-vocoder

Unsupervised adaptation to new speakers in feature-based letter recognition

Improved hidden Markov modeling of phonemes for continuous speech recognition

Improvement of the narrowband LPC synthesis

Feature-based speaker-independent word recognition without oral learning

Speech recognition under additive noise

Optimal Lloyd-Max quantization of LPC speech parameters

A speaker independent word recognition system based on phoneme recognition for a large size (212 words) vocabulary

One-pass syntax-directed connected-word recognition in a time-sharing environment

Use of computational psychoacoustical models in speech processing: Coding and objective performance evaluation

Auditory models in isolated word recognition

The CDTWP: A programmable processor for connected word recognition

Line spectrum pair (LSP) and speech data compression

Comparison of a model of the peripheral auditory system and L.P.C. analysis in a speech recognition system

Evaluation of time compression for connected word recognition

A multirate root LPC speech synthesizer

Segmentation in isolated word recognition using vector quantization

Computational models of neural auditory processing

Baseband speech coding at 2400 bps using "Spherical vector quantization"

Speech segmentation and recognition using adaptive linear prediction algorithm

Pitch and spectral estimation of speech based on auditory synchrony model

A sample selective linear prediction analysis of speech

A nonlinear spectrum processing technique for speech enhancement

A parametric representation of short-time power spectra based on the acoustic properties of the ear

Modelling of the laryngeal acoustic source by labile nonlinear oscillators

Spectral envelope sampling and interpolation in linear predictive analysis of speech

Optimal estimators for spectral restoration of noisy speech

Automatic glottal inverse filtering

An algebraic approach to discrete short-time Fourier transform analysis and synthesis

Multisensor speech input for enhanced immunity to acoustic background noise

Two channel (speech and egg) analysis for formant and glottal inverse filtering

Speech synthesis from short-time Fourier transform magnitude and its application to speech processing

Adaptive noise cancellation in a fighter cockpit environment

Acoustic tube analysis of formant bandwidths and frequencies in helium speech

Time alignment of natural speech to synthetic speech

The harmonic magnitude suppression (EMS) technique for intelligibility enhancement in the presence of interfering speech

An all-zero model for higher pole correction

Use of dynamic programming for automatic synchronization of two similar speech signals

The GRASP sound separation system

Comparative evaluation of a new procedure for adaptive estimation of noisy images

A procedure for automatic alignment of phonetic transcriptions with continuous speech

Accurate pitch determination of speech signals by means of a laryngograph

An iterative method for restoring noisy blurred images

Synthesis by rule of english intonation patterns

A powerful post-processing algorithm for time-domain pitch trackers

One-dimensional processing for adaptive image restoration

A text-to-speech system for italian

A pitch detection algorithm with hypothesis and test strategy by means of fast surface AMDF

Nonstationary 2-D recursive restoration of images with signal-dependent noise

Real time text-to-speech conversion system for spanish

A performance comparison of pitch extraction algorithms for noisy speech

A system for converting teletext into speech

Image restoration by the method of generalized projections with application to restoration from magnitude

Investigation of text-independent speaker identification techniques under conditions of variable data

Towards increasing the commercial success of speech synthesizers

Singular value decomposition, singular vectors, and the discrete prolate spheroidal sequences

A speaker recognizability test

Fast frequency tracking using an adaptive lattice filter for a vortex flowmeter signal

Experiments on the use of local statistics for adaptive image processing

An endpoint detector for LPC speech using residual error look-ahead for vector quantization applications

A real-arithmetic implementation of the constant modulus algorithm

Homomorphic restoration of images degraded by light cloud cover

Analog interface chips for audio band digital signal processing

Convergence conditions for an optimal solution in adaptive recursive filters

Iterative generalized inverse image restoration

Experimental hardware for real-time wideband speech coding

Extension facilities and performance of an LSI adaptive filter

Testing of wideband digital coders

A two-dimensional adaptive image deblurring filter

On the probability density functions of the weight for the complex scalar LMS adaptive algorithm

A flexible sampling-rate conversion method

Recursive Wiener filtering for image restoration

Time correlation statistics of the LMS adaptive algorithm weights

A new digital voice summing technique for teleconferencing

An improved signal restoration method using frequency domain information

Nonstationary learning characteristics of least squares adaptive estimation algorithms

Automatic gain control

A parametric approach to bispectrum estimation

Stable time- and order-recursive algorithm for the adaptive lattice filter

Modelling musical instruments in the digital domain

ARMA model order/Data length tradeoff for specified frequency resolution

Characteristics of adaptive filters with leakage

A speech direction finder

Computationally efficient estimation of the mean frequency for real-valued signals

Adaptive echo cancellation with nonlinear digital filters

Sound quality measurements of audio systems based on models of auditory perception

Invariant detection of transient ARMA signals with unknown initial conditions

On realizations and related algorithms for adaptive linear phase filtering

The minimum detectable delay of speech and music

Explicit formulas for estimating the frequencies of sine waves in noise with few samples

Two-dimensional spectral estimation with autoregressive lattice parameters

A quantitative comparison of generalized correlator window functions in presence of strong spectral peak for spatially separated sources

Application of a simple hearing model to the design of audio filters

Application of SVD to 2-D spectral estimation

The complete characterization of a multiple sinusoid signal

Requirements for loudspeaker crossover networks

Tomographic and spectral analysis using noise statistics

A technique for spectral component location within a FFT resolution cell

Initial condition transient suppression for two dimensional recursive digital filters

An iterative algorithm for robust 2-D spectrum estimation

Scale-space filtering: A new approach to multi-scale description

On 2-D FIR and IIR filter design

A model based approach for 2-D MEPS analysis

The signal data base system SDB

Circularly symmetric filter design using 2D prolate spheroidal sequences

The importance of random phase for image reconstruction from frequency offset Fourier data

Signal-to-symbol transformation: Reasoning in the HASP/SIAP program

Realization of two-dimensional digital filters by LU decomposition of their transfer function

Improvement of resolution and reduction of computation in 2D spectral estimation using decimation

Knowledge based speech analysis and enhancement

Uncertainty, eigenvalue problems and filter design

Short-space Fourier transform image processing

"Ignorance-based" systems

Computation of the threshold of stability for N-dimensional digital filters

2D Discrete cosine transform computation by fast polynomial transform algorithms

Spectral estimation using level-crossing data

Two-dimensional passive state-space digital filter design

A state space approach for the 2-D harmonic retrieval problem

Expert systems as automated decision aids

Structural pseudolosslessness and structural pseudopassivity of 1-D and 2-D digital filters

A high speed two-dimensional FFT processor

Automated detection in multiple-target environments using the censored mean-level detector

Two stability tests for two-dimensional digital filters

A new method for wideband sensor array processing

High-resolution techniques for two-dimensional estimation of angle-of-arrival for planar arrays

Transfer function realization of a class of doubly-terminated two-variable lossless networks and their application in linear-phase 2-dimensional digital filter design

Accurate pole estimation by modified linear prediction

The effect of an auxiliary source on the performance of a randomly perturbed array

Median filters: Analysis for 2 dimensional recursively filtered signals

Comparison of three auto-regressive modeling methods

Joint source and sensor location estimation

Adaptive signal processing for adaptive control

Relationship between maximum-likelihood-method and autoregressive modeling in multidimensional power spectrum estimation

A point mechanical model for the dynamics of towed arrays

Detection of multiple sinusoids using a parallel ALE

Parameter estimation algorithms utilizing implicit phase

Optimal passive localization from a single sensor using multiple linear hypotheses

Detection of multiple sinusoids using an adaptive cascaded structure

Adaptive least-squares for parametric spectral estimation and its application to pulse estimation and deconvolution of seismic data

An all-digital realization of a baseband DLL implemented as dynamical state estimator

Adaptive algorithms that restore signal properties

Signal subspace analysis and improvement of spectral estimation algorithms

A convergence analysis of an adaptive underwater passive tracking system

A parametric method for computing magnitude squared coherence

Applications of singular value decomposition to system modeling in signal processing

A new approach to decentralized array processing

Suppression of FM adjacent channel interference

The analysis of non-oscillating transients using the covariance method with many different orders

Optimal linear arrays for narrow-band beamforming

The implementation of digital filters using a modified Widrow-Hoff algorithm for the adaptive cancellation of acoustic noise

A new spectrum analysis approach using autocorrelation technique and MEM

Composite complex sinusoidal modeling for the estimation of directions and spectra of incident plane waves

A fast least squares linear phase adaptive filter

Topological aspects of the caratheodory problem

Modal decomposition for detection of plane waves with a line array receiver

A new realization scheme of periodically time-varying digital filters

On adaptive implementations of Pisarenko's harmonic retrieval method

A new class of broadband time domain element space antenna array processors

Time-varying autoregressive modeling of a class of nonstationary signals

Constrained lattice structures for harmonic retrieval

A bit serial linear array DFT

Signal estimation using modified Wigner distributions

Determining the number of signals by information theoretic criteria

Hardware realization of Mersenne number transforms for fast digital convolution

The design of well-conditioned discrete imaging systems

Square root normalized feedback ladder algorithm for the identification of moving average systems

A bit serial LDI recursive digital filter

Recognition of time-varying signals in the time-frequency domain by means of the Wigner distribution

Fitting sinusoids to sampled data using a hybrid S/Z plane moment method

A residue to mixed radix converter and error checker for a five-moduli residue number system

Fast sequential least-squares processing

Autoregressive spectral estimation in noise with application to speech analysis

Parallel processing of digital convolution using finite polynomial rings

A framework for the evaluation of spectral analysis techniques

A new harmonical algorithm for digital signal processing

Pipelined cordic architectures for fast VLSI filtering and array processing

Applications of the median filter to digital radiographic images

Non-uniform sampling for high resolution spectrum analysis

Distributions in signal theory

Statistical analysis of two dimensional median filtered images

Sampling a broad band sparse spectrum without anti-aliassing filters

Gabor representation and Wigner distribution of signals

On-line trend detection based on ARI modeling

A comparison of image filtering algorithms

Measuring the degree of non-stationarity by using the Wigner-Ville spectrum

A new N.A.R power estimation for adaptive detection

Algorithms for the simulation of geometric distortion in a satellite imaging system

Some features of time-frequency representations of multicomponent signals

Multichannel adaptive array processing for optimal detection

Solving for 3D motion parameters of a rigid body by a vector geometrical approach : Uniqueness and numerical results

Time-frequency analysis using time-dependent ARMA models

Estimation of source parameters by maximum likelihood and nonlinear regression

Calculation of displacement fields by means of the motion detection transform

Joint representations (JR) in signal theory (ST) and hilbertian analysis: A powerful tool for signal analysis

Separation of sinusoids using the constrained adaptive line enhancer

Filtering and correlation of time-sequentially sampled spatiotemporal signals

On the time-frequency discrimination of energy distributions: Can they look sharper than Heisenberg ?

Estimation of angles-of-arrival for wideband sources

Object identification and orientation determination in 3-space with no point correspondence information

Spectrograms and generalized spectrograms for classification of random processes

Performance characteristics of biased estimators

A method for finding straight line and plane correspondences in stereopair images

A range and azimuth estimator based on forming the spatial Wigner distribution

Track likelihood statistic for the I Q locked loop

Adaptive image smoothing algorithms for edge and texture preservation

Voiceless stop consonant identification using LPC spectra

Kalman estimation of frequency-domain interference reduction coefficients for angle-modulated signals

New results in seismic deconvolution using lattice processors

Recognition of consonants using an ARMA model of the speech signal

Essential limitations to signal detection and estimation: An application to the arctic under ice environmental noise problem

Beam-steered vertical seismic arrays for wave classification

Properties of consonant sequences within words and across word boundaries

The integrated signal processing system ISP

Slowness aliasing in the Radon transform

How to avoid vowel normalization in identification of vowels in continuous speech

A graphical notation for describing data flow in digital filters

A model for sonar bottom sounding processing

An information theoretic approach to the automatic determination of phonemic baseforms

A graph theoretic technique for the generation of systolic implementations for shift-invariant flow graphs

Coherent recombination of sediment borne and water path acoustic signals

Modeling of english speech for the design of a distributed speech understanding system

On hardware description from block diagrams

The influence of Doppler effect on information transmission in sound channel

Algorithms for acoustic prosodic analysis

Design framework for systolic-type arrays

Spatial properties of pulsed-Doppler current profiling systems

An expert system for the automatic reading of French spectrograms

Signal processing software tools

A frequency domain method for time-shift estimation and alignment of discrete time signals

Integration of acoustic, phonetic, prosodic and lexical knowledge in an expert system for speech understanding

A modular method for designing custom signal processing integrated circuits

Characterization and recognition of underwater transient signals

The French language database: Defining, planning, and recording a large database

Digital processing structures for VLSI implementation

Observer adaptive PRF for synthetic aperture radar imaging

Evaluation of signal processing architectures via an integrated simulation environment

A database for speaker-independent digit recognition

Highly parallel architectures and algorithms for speech analysis

On creating averaging templates

Linear complexity fast algorithms for a class of linear equations

An algorithm basis for systolic/Wavefront array software

Decision rules for speaker-independent isolated word recognition

A pipeline architecture for implementing Durbin's recursive procedure

Systolic architectures for deconvolution algorithms

Application of fast algorithms to spectral analysis of speech signal

A method for recognizing Japanese monosyllables by using intermediate cumulative distance

A fast tally structure and applications to signal processing

An efficient algorithm for characteristic polynomial of 2-D state-space structures

Parameter selection for isolated word recognition using vector quantization

Fast transform processor implementation

Efficient algorithms and structures for lagged least squares (LS) FIR filters in the case of prewindowed signals

Evaluation of an isolated word recognizer in talker-dependent and talker-independent modes using a large telephone-band data base

Multiprocessor system for speech processing and telecommunications

A divide-and-conquer approach to the evaluation of the characteristic polynomial of a symmetric tridiagonal matrix

New measures of performance for speech recognition systems

A family of custom VLSI circuits for speech recognition

On the computational organization of fast sequential algorithms

Cost-effective speech processing

Architecture for a real-time LPC-based feature measurement integrated circuit

A decomposition of the arithmetic for NTT's with 2 as a root of unity

An endpoint relaxation method for dynamic time warping algorithms

A VLSI parallel machine for speech recognition

A step toward real-time interactive FIR filter design

Performance of isolated word recognition system for degraded speech

A wafer scale integration systolic processor for connected word recognition

New CMOS chip facilitates multibit correlation

Speaker-independent isolated word recognition using a one-pass analysis

The DTWP : An LPC based dynamic time warping processor for isolated word recognition

A low cost FFT chip set

Fast search algorithms for vector quantization and pattern matching

A CMOS-VLSI rate conversion digital filter for digital audio signal processing

A dynamic time warp IC for a one thousand word recognition system

On the problem of dimensionality and sample size in multi-stage pattern classifiers

A digital signal processor module architecture and its implementation using VLSIs

Recognition complexity with large vocabulary

An experiment with a non-head-mounted microphone for discrete utterance recognition (DUR)

An IEEE696 compatible signal processor

Preliminary results for an operational definition and methodology for predicting large vocabulary DUR confusability from phonetic transcriptions

Efficient computation and encoding of the multipulse excitation for LPC

MOS-VLSI pipelined digital filters for video applications

Suprasegmentals in very large vocabulary isolated word recognition

A non-iterative algorithm for obtaining multi-pulse excitation for linear-predictive speech coders

Architecture and control of a distributed signal processor

A model of lexical access from partial phonetic information

A study on short-time phase and multipulse LPC

Circuits for digital signal processing

Some experiments with large-vocabulary isolated-word sentence recognition

Experimental evaluation of different approaches to the multi-pulse coder

Some experiments in automatic recognition of a thousand word vocabulary

A logarithmic arithmetic unit for signal processing

9.6 kbit/s Piecewise LPC residual excited coder using multiple-stage vector quantization

Telephone communication between deaf and hearing persons

One approach to microprocessor implementation of 4800 b/s data modem for telephone channels

Vector quantizers for subband coded waveforms

Speaker adaptation in large-vocabulary voice recognition

Implementation of a full duplex 2.4 kbps LPC vocoder on a single TMS-320 microprocessor chip

Fully vector-quantized subband coding with adaptive codebook allocation

Phrase speech recognition of large vocabulary using feature in articulatory domain

A single-processor LPC vocoder

A comparative study of various speech encoding schemes using vector quantization

Connected digit recognition using vector quantization

A DFT-based front-end for word recognition systems

Hierarchical vector quantization of speech with dynamic codebook allocation

Effect of vector quantization on a continuous speech recognition system

A CCD chip for parallel pulse-Doppler radar processing

Predictive vector quantization

A procedure for designing exact reconstruction filter banks for tree-structured subband coders

A programmable signal processing element designed for an efficient data-driven signal processing architecture

Tree encoding of the LPC residual

16 kbps APC with hybrid quantization

System identification techniques for noise reduction in evoked potential processing

A new concept for encoding speech amplitude time quantization

Backward adaptive lattice and transversl predictors for ADPCM

On all-zero modelling of a recursive digital filter

A new class of very low sensitivity cascade-form digital-filters based on "passive" second order single-input single-output building blocks

Efficient nonlinear system identification

Performance analysis of DPCM codecs operating on adaptively frequency mapped 7.6 khz speech

Some properties of a family of generalized time-limited window functions

Sources of numerical errors in both the square-root normalized and unnormalized least squares lattice algorithms

Computationally efficient QMF filter banks

Variable-frequency synthesis: An improved harmonic coding scheme

Asymptotic performance of eigenstructure spectral analysis methods

Efficient implementation of highly variable bandwidth filter banks with highly decimated output channels

Magnitude-only reconstruction using a sinusoidal speech modelMagnitude-only reconstruction using a sinusoidal speech model

Mirror filters with minimum delay responses for use in subband coders

Non-linear system modelling : Concept and application

Peak-and-valley impulse excited LPC voice

On the reconstruction of a smooth function out of its samples

ARMA system identification: An algebraic approach

Implementation of the RELP vocoder using the TMS320

Reconstruction of signals after filtering and sampling rate reduction

An iterative algorithm for locating the minimal eigenvector of a symmetric matrix

Real-time 9600 bits/sec speech coding on the TI Professional Computer

A design method for multirate filters

A qualitative instability theory for lattice filters

Voice excited predictive coder (VEPC) implementation on 10 MIPS signal processor

The generalized phase spectrum method for time delay estimation

N-path digital filters

A 16kbps APC codec using digital signal processors

Correlation of interpolated time delayed communication signals

Pipelined orthogonal digital lattice filters

A 16 kbit/s speech coder using a single DSP device

On the convergence behaviour of frequency-domain LMS adaptive digital filters

A multi-signal bus architecture for FIR filters with single bit coefficients

A modified discrete fourier transform algorithm: A delta rotation algorithm

3-D reconstruction from projections based on dynamic object models

On the histrograms of the correlation functions of product codes

An in-order, in-place radix-2 FFT

Image reconstruction from frequency domain data on arbitrary contours

A blind equalization algorithm without decision

FCT -- A fact cosine transform

Faster phase only image reconstruction

Comments on parametric and non-parametric detection of epileptiform spike activity

Complex number theoretic transform in p-adic field

An image reconstruction algorithm for time varying X-ray projection image sequences

Phase unwrapping via median filtering

Application of Quadratic-Like complex residue number system arithmetic to ultrasonics

Adaptive equalizer using finite-bit power-of-two quantizer

Signal reconstruction from one bit of Fourier transform phase

An efficent implementation of a hexagonal FFT

Adaptive array cancellation of multipath interference

Soft constraint iterative reconstruction from noisy projections

Prime factor FFT algorithms for real-valued series

Tracking a meneuvering target in clutter-a new approach

Radon transform theory for random fields and optimum image reconstruction from noisy projections

Signal deconvolution by projection onto convex sets

Modified distribution free CFAR processor for clutter edges and multi-target situations

A new approach to the solution of the maximum entropy image reconstruction problem

Performance of minimum-variance deconvolution filter

Clutters cancellation using autoregressive techniques

Two special digital processing systems for recognition of moving objects

Fast minimum variance deconvolution

Target-clutter identification by lattice processors

Image restoration by parallel processing

modelling and deconvolution of nonstationary acoustic signals from moving sources using a convariance equivalent formulation

Distribution free Doppler processor

Enhancement of audio signals based on digital techniques

Abel inversion by Kalman filtering

Resolution of range and Doppler ambiguities in medium PRF radars in multiple-target environment

Interpolation of band-limited discrete-time signals by minimizing out-of-band energy

Optimal constrained restoration and dynamic programming

Side-lobe canceller response for pulse interference

Algorithm design for real-time audio signal processing

Compression of underwater images for their transmission on a low bit rate acoustic channel

On applications of linear prediction filtering to small wavelength Doppler weather radar signal processing

Editing digital audio

Morphological skeleton representation and coding of binary images

An efficient procedure for broadband Doppler compensation

Applications of a digital audio-signal processor in T.V. sets

Quad-tree predictive coding: A new class of image data compression algorithms

Modern, active sonar AGC design considerations

A high-resolution modified Burg algorithm for spectral estimation

Recovery of gray scale images from approximations of their contours

A method for computation of wideband ambiguity function and the numerical analysis of the bat's sonar signal

Efficient maximum entropy spectral estimation using non-multiplication methods

Multichannel linear predictive coding of color images

Maximum likelihood estimation of object size and orientation from projection data

ARMA spectrum estimation by extended lattice

Color image compression by adaptive vector quantization

A comparison of several optimal random quantization algorithms for correlation estimation

A matrix quantizer for image processing

CMT and hybrid coding of the component color TV signal

Two step color shrinking algorithm for encoding of graphics

Artificial intelligence applied to spectrum estimation

Solid state colour camera simulator: Preliminary results

Bandwidth reduction and image distortion characteristics of the LMS image compression algorithm

Multichannel relative-entropy spectrum analysis

A new image coding technique using transforms vector quantization

Universal compression lossless code statistically built

ARMA spectral estimation of time series with missing observations

Block classification image coding with combined transforms

Real time discrete cosine transform an original architecture

Power spectrum estimation by means of relative-entropy minimization with uncertain constraints

An optimal transform tree coding method applied to images

A coefficient energy concentration model for motion compensated image coding

Maximum entropy spectrum estimation from noisy correlation measurements

Design of optimal finite wordlength linear phase FIR filters: New applications

A transformation for clustering noisy shape observations

Maximum likelihood and bartlett spectrum estimates

An echo canceller having reduced word size taps and using the sign algorithm with extra controlled noise

Lag-invariant adaptive spectrum estimation

Segmentation of 2-D seismic data

Numerical accuracy and stability: Two problems of adaptive estimation algorithms caused by round-off error

On the computation of the Cramer Rao bound for ARMA parameter estimation

Edge-oriented spatial filtering of images with application to post-processing of vector quantized images

Reduction of quantization effects in adaptive filters

Confidence regions for spectral bounds

Edge detection using the linear model

Accumulator quantization noise in the single-weight complex LMS adaptive algorithm

Performance comparison of autoregressive estimation methods

Seismic signal character recognition using analysis techniques

Statistical design of cascade finite wordlength FIR digital filters

The magnitude squared coherence estimate: A geometric view

Robust image estimation in signal-dependent noise

Application of the random reference quantization principle to radix-2 FFT computation

Signal processing models for point processes

Optimal data estimation system over multipath channel

Optimum entropy-coded quantizer design for a class of discrete-time sources

Adaptive algorithms for estimating eigenvectors of correlation type matrices

Accuracy of frequency estimation and its relation to prediction filter order

On redefining the optimal least squares filter under floating point operations

Complex white noises and autoregressive signals

Synthesis by rule of voice fundamental frequency contours of spoken Japanese from linguistic information

Error analysis of LMS adaptive digital filter implemented with logarithmic number system

Signal rank and model order determination

Estimation of frequencies of sinusoids using the extended Kalman filter

Efficient CAD tools for the coefficient optimisation of arbitrary integrated digital filters

On the asymptotic normality of autoregressive spectral density estimates for the noise corrupted case

Automatic recognition of spoken words from a large vocabulary using syllable templates

Effects of finite Kernel word length in signal processing

Measuring true spectral density from ML filters (NMLM and q-NMLM spectral estimates)

Increasing the parallelism of filters through transformation to block state variable form

Implications of threshold effects for coherent and incoherent processing techniques of time delay estimation

Quantization reduction for evaluating laser gyro performance using a moving average filter

Time delay estimation with low signal to noise ratios

Procedures for signal reconstruction from noisy phase

Comparison of the Ziv-Zakai lower bound on multipath time delay estimation with autocorrelator performance

Iterative restoration by generalized replacement

Alternatives to the generalized cross correlater for time delay estimation

Extrapolation of discrete-time band-limited signals using singular value decomposition with decimation

Performance of the deskewed short-time correlator

Spectral estimation and matched filter performance

An efficient Doppler compensated correlation and low doppler interference removal

Envelope modelling of impulse response functions of systems having a high modal density

Bias free estimation of the variance of time of arrival differences

Band-limited signal extrapolation in the presence of noise

A new method for the passive locating of unknown source by using a vertical dipole receiver

Spectra using data distribution and covariance modelling

Adaptive multipath delay estimation

Nonlinear filters for reducing spiky noise: 2-dimension

Adaptive realization of the phase transform for time delay estimation

Experiments with extrapolation of band-limited signal

A comparative study of the LMS adaptive filter versus generalized correlation method for time delay estimation

Relationship between Paley-Wiener theorem and the stationary phase method?

Motion parameter estimation of a fast moving sound source using the retardation effect

Difference operation for Pre/Post-emphasis in linear prediction analysis

Digital signal processing capabilities of CUSP, a high performance bit-serial VLSI processor

A one-source photometric method for N-order specular surfaces

The SP16 signal processor

Two dimensional stochastic fractional AR models with strong periodicities

SASP-A digital signal processor system for speech processing applications

Image segmentation using spatial linear prediction

A real time general purpose signal processor

Image segmentation using maximum entropy techniques

A VLSI digital filter bank

Application of the Gibbs distribution to image segmentation

A high-speech 32 bit IEEE floating-point chip set for digital signal processing

Bayes smoothing algorithms for segmentation of images modeled by Markov random fields

A 16-bit three port arithmetic logic and shift unit

A maximum-likelihood approach to image segmentation by texture

VLSI Architecture for signal processing with alternate low-level primitive structures (ALPS)

Multi-level thresholding and its application to feature extraction in machine parts

Adder-based digital signal processor architecture for 80 NS cycle time

Classification of textures using Markov random field models

Image vector quantization with a perceptually-based cell classifier

Machine recognition of partial planar shapes using feature vectors

Segmentation of computerized tomography images for three dimensional analysis

Bearing estimation in a shallow water environment using the eigenvector decomposition method

Decreasing high resolution method sensitivity by conventional beamformer preprocessing

An improved eigenvector beamformer

Matrix decomposition and multiple source location

New adaptive processor for coherent signals and interference

Resolution of incoherent and coherent sources by autoregressive beamforming

Sea surface reverberation rejection

Structure of covariance matrix and eigenvalues of broadband tapped delay line adaptive array

The effect of interference extent on LMS spatial cancellation

Application of the maximum entropy beamformer to a shallow water line array

Application of singular value decomposition to adaptive beamforming

Synthetic aperture sonar: An analysis of beamforming and system design

Degradation of angular resolution for eigenvector-eigenvalue (EVEV) high-resolution processors with inadequate estimation of noise coherence

Systolic array technique applied to symmetric FIR filters

Concurrent array processor for fast eigenvalue computations

The application of a systolic least squares processing array to adaptive beamforming

An FFT systolic processor and its applications

VLSI architectures for dynamic time warping using systolic arrays

A three dimensional systolic array architecture for fast matrix multiplication

A manpack portable LPC 10 vocoder

The VLSI design of a sub-band coder

Real time implimentation of 16kbs APC with Hybrid Quantization

A speech input and processing board for a personal computer

Architecture for a VLSI implementation of an LPC-based, isolated-word recognition system

ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing [Back cover]

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

Signal subspace analysis and improvement of spectral estimation algorithms

Content Provider	IEEE Xplore Digital Library
Author	Guaning Su
Copyright Year	1984
Description	Author affiliation: Defense Science Organization, Ministry of Defence, Singapore (Guaning Su)
Abstract	The signal subspace approach was introduced by Schmidt [1] for the narrowband source location problem and generalized by Su and Morf ([2] - [5]) for wideband source location and multidimensional spectral estimation. Many authors have made contributions to related methods of spectral estimation and source location, generally using eigenstructure or singular value decomposition. In this paper three methods of spectral estimation are related to the signal subspace approach. Besides demonstrating the common thread in their development, the signal subspace approach suggests immediate modifications and generalizations of the methods. The methods are those of Prony [6], Tufts-Kumaresan [7] and Cadzow [8].
Starting Page	204
Ending Page	207
File Size	76352
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1984.1172414
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1984-03-19
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Signal analysis Algorithm design and analysis Sampling methods Additive noise Equations Position measurement Narrowband Wideband Multidimensional systems Singular value decomposition
Content Type	Text
Resource Type	Article

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