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Volume 8 - 1983

A new adaptive parameter estimation structure applicable to ADPCM

Wideband frequency domain adaptive filter module

Time-varying spectral estimation using symmetric smoothing

Convergence behavior of the constant modulus algorithm

IIR algorithms for adaptive line enhancement

"Design and realisation of adaptive lattice filters"

An adaptive short-term correlator algorithm

A round-off error analysis of the LMS adaptive algorithm

Fixed-point implementation of adaptive digital filters

An adaptive nonlinear digital filter with lattice orthogonalization

The filtered error LMS Algorithm

On computational complexity in adaptive digital filters

A new structure for adaptive echo cancellation

Application of modified least-squares algorithms to adaptive echo cancellation

Application of adaptive noise cancelling in a noisy reverberant environment

A segment vocoder at 150 b/s

Very low data rate speech compression with LPC vector and matrix quantization

A comparison of methods for 300-400 b/s vocoders

An 800 bps adaptive vector quantization vocoder using a perceptual distance measure

Data rate reduction of gain and pitch parameters in an LPC vocoder

Efficient coding of LPC parameters by temporal decomposition

Delta coding of LPC parameters

Improvement of the LPC analysis

Cepstral analysis synthesis on the mel frequency scale

Design and test of a real-time floating point LPC vocoder

Vocoded speech through fading channels

Two-dimensional linear predictive analysis of arbitrarily-shaped regions

Object detection by two-dimensional linear prediction

The converging squares algorithm: An efficient multidimensional peak picking method

Recovery of gray scaled images from contour processed representations

Determining 3-D motion and structure of a rigid body using straight line correspondences

Estimating 3-D motion parameters and object surface structures from the image motion of conic arcs, I: Theoretical basis

Fast parallel image processing for robot vision for the purpose of extracting information about objects

Reconstruction from projections based on detection and estimation of objects

The use of a priori information in image reconstruction from limited data

Algorithms and experimental results in acoutistic tomography using Rytov's approximation

Image restoration in CT by the method of projection onto convex sets

Reconstruction algorithms for limited view projection data

Iterative reconstruction of space-limited scenes from noisy frequency-domain measurements

Numerical evaluation of Fourier-Bessel series expansion

Number theoretic transform based on ternary arithmetic

Longer NTT's with 2 as a root of unity

On the structure of efficient DFT algorithms

A fast algorithm for solving Toeplitz system of equations

Efficient inversion of doubly block Toeplitz matrix

A fast non-recursive algorithm and a parallel processor architecture for smoothing spline fitting

Fast algorithms for linear prediction and system identification filters with linear phase

Efficient computation of the covariance sequence of an autoregressive process

Fast Kalman type algorithms for sequential signal processing

An in-place and in-order WFTA

Constrained Chebyshev approximation for FIR filters

Design of FIR filters with flatness constraints

Simultaneous design in both magnitude and group-delay of IIR and FIR filters: Problems and results

A new class of narrow transition band digital filters

Multiplier-less design of FIR filters

On the synthesis of lattice parameter digital filters

Synthesis and fixed-point implementation of pipelined true orthogonal filters

Complex quadrature mirror filters

Design and evaluation of parallel quadrature mirror filters (PQMF)

A novel approach to the design of analysis/Synthesis filter banks

A least squares approximation design of complex FIR filters

Iterative algorithms for optimal signal reconstruction and parameter identification given noisy and incomplete data

Maximum likelihood parameter estimation with a min/Max criterion

Stochastic state-space models from empirical data

Identification and estimation algorithms for a Markov chain plus AR process

An improvement of non-stationarity detection by level-crossing analysis of linear prediction error

The composite pseudo Wigner distribution (CPWD): A computable and versatile approximation to the Wigner distribution (WD)

Systems modeling by digital signal processing

Evolutionary spectral density models for random processes having a frequency modulated structure

Non-stationary AR model identification by batch estimation

Estimation of non-stationary moving-average models

Time series modeling via general linear estimation theory

One approach to simulation of modulated signals

Application of a cepstral distance measure in evoked potential processing

An efficient on-line load-modeling algorithm for short-term forecasting of interconnected power systems

Experiments in connected word recognition

Some dimensionality reduction studies in continuous speech recognition

A connected spoken word recognition method by O(n) dynamic programming pattern matching algorithm

Secondary testing techniques for word recognition in continuous speech

The Erlangen system for understanding continuous German speech

A Chinese speech recognition system

Connected-word recognition system using speaker-independent phonetic features

Phonetic feature hypothesization in continuous speech

Segmentation-free syllable recognition in continuously spoken Japanese

Considerations on articulatory dynamics for continuous speech recognition

A generalized approach to high-resolution array processing

Eigensystem properties of the sampled space correlation matrix

Improving the resolution performance of eigenstructure-based direction-finding algorithms

Modal decomposition signal subspace algorithms

Capability of array processing algorithms to estimate source bearings

A triangular adaptive lattice filter for spatial signal processing

Analysis and synthesis of general planar interferometer arrays

An escalator structure for adaptive beamforming

Evaluation of an orthogonal beamforming procedure using real data

Signal separation theory by using adaptive array

Digital extrapolation beamforming

Localization with arrays subject to sensor motion

Noise normalization of broadband sonar data in bearing space

Design of FIR two-dimensional digital filters by successive projection

Optimal design of two-dimensioal FIR-filters

Equiripple two-dimensional interpolation filters

Methods for image interpolation through FIR filter design techniques

Design of low sensitive 2-D analog and recursive digital filters with prescribed magnitude and group delay specifications

A new method for the design of 2-dimensional stable recursive digital filters satisfying prescribed magnitude and group delay response

2-D non-causal systems: State space modeling for half-plane support

Image convergence under two dimensional separable median filtering

Some generalizations of median filters

2-D state-space realizations with fewer multipliers and invariant norms

Sectioning implementation of two-dimensional symmetric half-plane filters

A class of data-flow architectures for speech recognition

Architecture and programming of a multirate digital filter

MUSEC, a powerful network of signal microprocessors

Architecture and instruction set of a programmable LSI digital filter

A distributed signal processor incorporating VLSI and high order language programming

Efficient FIR filter implementation using microprocessor

Enhancing microcomputer for high-speed vector processing

Algorithm and a new processor architecture for computing the DFT

Modular architectures for adaptive multichannel lattice algorithms

A reconfigurable signal processor for high throughput applications

A multichannel input subsystem

An adaptive echo canceller using digital signal processor LSI chips

Parallel processing for computationally intensive speech analysis operations

A programmable voice digitizer using the T.I. TMS-320 microcomputer

Real time implementation of a speech coding algorithm using time-domain harmonic scaling and adaptive residual coding

The APS-II processor for speech recognition

Implementation of various LPC algorithms using commercial digital signal processors

On the Wigner distribution

Architecture and application of a commercially available speech recognition board

A combination speech synthesis and recognition integrated circuit

A microprocessor for speech recognition

Systolic arrays for dynamic programming in speech recognition systems

A custom IC for automatic gain control in LPC vocoders

Design of LSI speech spectrum analyzer using switched capacitor filter techniques

Comparison of high speed hardware and software implementation approaches for a multichannel PCM-Delta converter

Speech recognition and noise

Speech recognition performance assessments and available databases

The effects of five voice characteristics on LPC quality

Perceptual evaluation of synthetic speech: Some considerations of the user/System interface

An algorithm for designing optimum quantizers subject to a multiclass distortion criterion

Objective speech quality evaluation of mediumband and narrowband real-time speech coders

The estimation and evaluation of pointwise nonlinearities for improving the performance of objective speech quality measures

Further investigation of probabilistic methods for text-independent speaker identification

An approach to text-independent speaker recognition with short utterances

A comparison of distance measures for text-independent speaker identification

Further experiments in text-independent speaker recognition over communications channels

An ARMA modeling method for estimation of coherence and time delay

Autoregressive detection by generalized Burg algorithm

Improved detection and tracking of dynamic signals by Bayes-Markov techniques

ORing loss model for implementation in signal processing systems for data display

Broadband correlation processing

Detection performance of the parametric correlator

Maximin sonar system design for detection

Performance bounds for SNR enhancement of narrowband signals in surface reverberation

Application of adaptive linear predictor structures to the prewhitening of acoustic reverberation data

MAP detection and ML estimation of signals overlapping in time and frequency for both Gaussian and Laplacian noise

Detection of non-Gaussian signals by frequency domain Kurtosis estimation

Sequential robust m-interval polynomial approximation (MIPA) partition detectors

A new approach for synthesis of low sensitivity digital filter structures based on lossless building blocks

Roundoff noise in error-feedback state-space filters

A modular and orthogonal digital filter structure for parallel processing

Finite word effects in 2-D block implemented fixed point digital filters

FFT-accuracy - new insights and a new point-of-view

Self-sustained stable oscillations of second order recursive algorithms

Noniterative techniques for minimum phase signal reconstruction from phase or magnitude

The initial estimate in constrained iterative restoration

In-time non-iterative fast-algorithm for support constrained deconvolution

A positive optimal deconvolution procedure

Application of pseudoinversion to scattering function estimation with discrete measurements

Iterative deconvolution using several different distorted versions of an unknown signal

Stochastic convergence properties of the adaptive gradient lattice

Convergence analysis of LMS and NLMS adaptive algorithms

Tracking characteristics of the Kalman filter in a nonstationary environment for adaptive filter applications

A new derivation for fast recursive least squares and Levinson algorithms by the conjugate direction method

Fast, fixed-order, least-squares algorithms for adaptive filtering

A new stable feedback ladder algorithm for the identification of moving average processes

Time-domain estimation of unambiguous Doppler frequency in low and medium PRF radars

A fast and accurate method for estimating target Doppler

Evaluation of angle of arrival estimators using real multipath data

Random phase tracking of non-uniformly sampled signals

Signal detection performance of lattice processors

Some new approaches for the implementation of optimum MTI filters

A new approach to scan to scan correlation and its implementation

A speaker independent word recognition based on transient matching

Syntactic pattern recognition of discrete utterances

Isolated word recognition using phoneme-like templates

Speaker independent isolated word automatic speech recognition using computer generated phonemes

Feature-based speaker-independent recognition of isolated english letters

Dynamic speaker adaptation for isolated letter recognition using MAP estimation

Recognition of consonant based on the perceptron model

Selecting acoustic features for stop consonant identification

A speech spectrogram expert

An adaptive feature extraction expert

Microcomputer implementable low cost speaker-independent word recognition

Vocabulary selection for high performance speech recognition

Bayesian adaptation in speech recognition

A comparison of three feature vector clustering procedures in a speech recognition paradigm

Automatic prediction of linear frequency warp for speech recognition

An investigation of spectral match statistics using a phonemically marked data base

Analysis and synthesis of speech based on spectral transform linear predictive method

Optimizing LPC filter parameters for multi-pulse excitation

Effect of a time-weighted error criterion on recursive estimation of speech parameters

On the role of the partial trigonometric moment problem in AR speech modelling

Stable modelling of a continuous covariance function with application to continuous speech

The phase-only version of the LPC residual in speech coding

Algorithms for signal reconstruction from short-time Fourier transform magnitude

Signal estimation from modified short-time Fourier transform

Finite impulse response (FIR) filters for speech analysis and synthesis

Some comments on the design and implementation of FIR filterbanks for speech recognition

ZIP: A dynamic programming algorithm for time-aligning two indefinitely long utterances

Image restoration based on an anisotropic noncausal stochastic model

Noncausal modeling and restoration of noisy images

Image restoration using order-constrained least-squares methods

A multiple model algorithm for the adaptive restoration of images

Image enhancement using the median and the interquartile distance

A hybrid technique to restore the Fourier phase and magnitude spectra of noisy linearly degraded images

New interpretations for the MLM and DASE spectral estimators

A linear programming approach to bounding spectral power

Improved maximum likelihood method for two-dimensional spectral estimation

Two-dimensional spectral estimation using spatial autoregressive models

A new class of high-resolution and robust multi-dimensional spectral estimation algorithms

Eigenfilter methods for 2D spectral estimation

Deterministic estimation of two-dimensional signals

On time delay estimation involving received signals

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

Accuracy of linearized performance predictions for wavefront curvature ranging systems

Comparison of two adaptive methods for time delay estimation

A parametric technique for estimation of delay and Doppler

Covariance eigenstructure approach to 2-D harmonic retrievel

Optimal delay estimation in a multiple sensor array having spatially correlated noise

Improved time delay estimation in the presence of interference

Comparison of error detectors for time delay tracking

Modified polar coordinates for ranging from Doppler and bearing measurements

The use of linear constraints to reduce the variance of time of arrival difference estimates for source location

An effective implementation of a time delay estimator for source location

A comparison of two SAR processing architectures for VLSI implementation

A VLSI architecture for digital filters using complex number-theoretic transforms

VLSI implementation of rotations in pseudo-Euclidean spaces

VLSI bit-sequential architectures for digital signal processing

Optimal choice of intermediate latching to maximize throughput in VLSI circuits

Case studies in VLSI signal processing using a silicon compiler

A 22-bit floating point registered arithmetic logic unit

A single board floating point signal processor

Digital signal processing with VLSI technology

CMOS LSI DSP and its application to voice band signals

High resolution spectral analysis and the acoustic remote sensing of ocean current velocity

The application of linear prediction techniques to background noise generation for sonar trainers

Applications of adaptive signal processing to the reduction of flow-induced noise in small underwater vehicles

Modal dispersion effects on coherent signal processing parameters

A detector for acoustic directionality

The impact of refraction on time-delay estimation

Lower bounds on pulsed-Doppler current profiler accuracy

Optimum array filters for array data signal processing

Primitive extraction for a syntactic pattern recognizer of features in seismic signals

The wavelet reconstructed with phase-only and a study of its uniqueness

Seismic velocity estimators

Systems of equations with near-to-Toeplitz or near-to-Hankel parameters and applications to signal processing

Deconvolution of band-limited signals

A model reduction algorithm by spline approximation in the deconvolution of seismic signals

Detecting and estimating parameter jumps using ladder algorithms and likelihood ratio tests

A generalization of isolated word recognition using vector quantization

Experimental comparison among data compression techniques in isolated word recognition

On the use of time compression for word-based recognition

Speech recognition using variable frame rate coding

Some techniques for incorporating local timescale variability information into a dynamic time-warping algorithm for automatic speech recognition

The discriminative network: A mechanism for focusing recognition in whole-word pattern matching

Use of a priori knowledge of vocabulary for real time discrete utterance recognition

Speaker independent isolated digit recognition using hidden Markov models

A learning procedure for speaker-dependent word recognition systems based on sequential processing of input tokens

Probabilistic model for the performance of speech recognition systems

On the use of filter bank features for isolated word recognition

Recognition of isolated-word sentences from a 5000-word vocabulary office correspondence task

Maximum entropy power spectrum estimation with uncertainty in correlation measurements

High-resolution frequency estimation via a weighted forward and backward autoregressive modelling

On the double Levinson recursion formulation of ARMA spectral estimation

An ARMA model for power spectrum estimation

A spectral matching technique for ARMA parameter estimation

Reduction of all-pole parameter estimator bias by successive autocorrelation

Effect of record length on the correlation of complex exponentials

On the resolution of autoregressive spectral estimation

On harmonic retrieval

A finite step adaptive implementation of the Pisarenko's harmonic retrieval method in colored noise

A fast algorithm for unmodified ARMA spectrum estimation

Fast recursive AR estimation from an overdetermined system of extended Yule Walker equations

Spectrum analysis of surface electromyogram (EMG)

Speech enhancement using optimal non-linear spectral amplitude estimation

Speech enhancement with harmonic synthesis

A simple adaptive filtering technique for speech enhancement

Speech noise reduction by means of multi-signal minimum-cross-entropy spectral analysis

Noise suppression methods for speech applications

Subjective evaluation of a PCM speech coding system with quantization noise reduction

Parameter isolating transforms and admissible distortions

Modelling and enhancement of reverberant speech using an envelope convolution method

A computational model of binaural localization and separation

Event driven speech enhancement

Experiments in isolated word recognition using noisy speech

Helium speech enhancement by frequency-domain processing

General - N Winograd D.F.T. programs with inverse option

The signal representation language SRL

A common operational software (ACOS) approach to a signal processing development system

Development of signal processing algorithms via functional programming

A new level of signal processing software: Automatic buffer address generation

Software structure for a distributed signal processing system

Realization of digital filter algorithms by use of a high speed parallel processing architecture

A preprocessor architecture to equalize focal-plane detector-array elements

Non-recursive digital FIR filter implementation using stored square ROM multipliers

A tale of two architectures: TI TMS 320 SPC vs. DEC Micro/J-11

Fiber-optic delay line devices for GigaHertz signal processing

An adaptive transversal filter

Reduction of blocking effect in image coding

Adaptive compression of teleconference sequences using the LMS Algorithm

Some experimental results in adaptive prediction DPCM coding of images

A comparison of interframe coding techniques

An objective quality measure for still moncromatic DPCM-coded images

A two-component image coding scheme based on two-dimensional interpolation and the discrete cosine transform

A high speed maximum entropy encoder for images

Linear predictive coding of imagery for data compression applications

Maximum likelihood estimation of composite source models for image coding

Implementation for two-dimensional FIR filters using the number theoretic transform

Transform algorithm for computing two-dimensional convolutions

The discrete cosine transform--A new version

New algorithms for Abel inversions and Hankel transforms

Row-column algorithms for the evaluation of multidimensional DFT'S on arbitrary periodic smapling lattices

High speed 2D hexagonal convolution by polynomial transform

Multiframe image point matching and 3-D surface reconstruction

A 32-band sub-band/Transform coder incorporating vector quantization for dynamic bit allocation

Polyphase quadrature filters--A new subband coding technique

Multirate sub-band coder with embedded bit stream: Application to digital TASI

Design of a novel 32.8 kbs speech coder using modulo-PCM principles

A maximum entropy encoder for speech

Enhancement of differentially coded speech signals by means of quantization error correction

On the tandem connection of differential encoding systems: The case of cascaded quantizers

Frequency compression of 7.6 kHz speed into 3.3 kHz bandwidth

Fixed block rate quantizer using entropy coding

Design of a 4.8/9.6 kbps baseband LPC coder using split-band and vector quantization

Implementation of a robust RELP speech coder

RELP-vocoding with uniform and non-uniform down-sampling

Pitch-congruent baseband speech coding

On the generation and use of a segment dictionary for speech coding, synthesis and recognition

Glottal sensing for speech analysis and synthesis

A look at the SRS synthesis rules for Japanese

A Swedish text-to-speech system based on an area function model

A text-to-speech development system

Knowledge-based pitch detection

An integrated pitch tracking algorithm for speech systems

Joint estimation of excitation and vocal tract response

SpireX: Statistical analysis in the spire acoustic-phonetic workstation

Speech analysis/Synthesis based on perception

A speech signal processing system based on a peripheral auditory model

Characteristics pf pitch period and amplitude perturbations in pathologic voice

Speech processing for artificial tactile speech displays

Pitch translation of trumpet tones

An algorithm and architecture for constant-Q spectrum analysis

Restoration of nonlinearly distorted magnetic recordings

Observation of reverberation curves by two relatively prime pseudorandom sequences

Perceptual identification and discrimination of phase distortions

Development of a technique for high frequency audiometry

A preliminary model of subjective voice quality

A harmonic distortion standard

Omnidirectional coded loudspeaker arrays

Complex matrix inversion algorithm for calculating digital system frequency response

A fast computational algorithm for and performance of the Kumaresan-Prony method of spectrum analysis

Highly concurrent Toeplitz eigen-system solver for high resolution spectral estimation

Simple, effective MA and ARMA techniques

An optimization theoretic framework for spectral estimation

On the sampling rate issue in spectral analysis

Reducing aliasing in the Wigner distribution using implicit spline interpolation

Cepstrum constraints in ME spectral estimation

The representation of signals in terms of spectral amplitude

On the bandlimited extrapolation of discrete signals

Confidence interval determination for spectral estimates using "Tilted densities"

Signal restoration from phase by projections onto convex sets

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

Estimation of non-stationary moving-average models

Content Provider	IEEE Xplore Digital Library
Author	Grenier, Y.
Copyright Year	1983
Description	Author affiliation: Ecole Nationale Superieure des Telecommunications, Paris, France (Grenier, Y.)
Abstract	This paper describes two methods for time-dependent moving-average modelling of non-stationary signals. In both of them, the finite-order MA model is approximated by an infinite-order auto-re gressive model. In the second one, the use of the AR model is explicit : by means of the inversion of this model, the innovation is estimated and the MA model is then obtained by a least-square procedure. In the first method, the AR model is implicit : a vector signal is associated to the scalar one, and the MA algorithm contains two steps : Schur parametrization of an estimate of the covariance of the vector process, followed by a reduction of the vector model to a scalar one.
Starting Page	268
Ending Page	271
File Size	68698
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1983.1172159
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1983-04-14
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Signal processing algorithms Speech synthesis Technological innovation Signal processing Adaptive signal processing Signal synthesis Speech analysis White noise
Content Type	Text
Resource Type	Article

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