NDLI: A 4.8 kbps multi-band excitation speech coder

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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

Neutral nets for computing

Hidden Markov models: a guided tour

Application of digital signal processing in professional audio

Decoder selection based on cross-entropies

On the relations between modeling approaches for information sources (speech recognition)

Application of the Gibbs distribution to hidden Markov modeling in isolated word recognition

A speaker-independent isolated-word recognizer based on polynomial classifiers

A family of distortion measures base upon projection operation for robust speech recognition

Speech recognition with continuous-parameter hidden Markov models

A neutral network for isolated-word recognition

Discriminant clustering using an HMM isolated-word recognizer

Dynamic adaptation of Hidden Markov models for robust isolated-word speech recognition

Isolated word recognition over the DDD telephone network. Results of two extensive field studies

A connectionist model for consonant-vowel syllable recognition

Speech signal detection at the output of a cochlear model

A multi-step excited model for speech parameter trajectories

Psychoacoustically based scalar quantization of the LPC poles

Improved speech recognition using a reduced auditory representation

A linear filter for time scaling of speech

Multi-frame approach for parameter estimation of a physiological model of speech production

Information theoretic factorization of speaker and language in hidden Markov models, with application to speaker recognition

On a general time-varying model for speech signals

A neural net approach to speech recognition

Use of neural networks for the recognition of place of articulation

Phoneme recognition: neural networks vs. hidden Markov models vs. hidden Markov models

Phonetic recognition using hidden Markov models and maximum mutual information training

1000-word speaker-independent continuous-speech recognition using hidden Markov models

High performance connected digit recognition, using hidden Markov models

Large-vocabulary speaker-independent continuous speech recognition using HMM

Stochastic segment modelling using the estimate-maximize algorithm (speech recognition)

Improved speaker adaption using text dependent spectral mappings

Lexical stress recognition using hidden Markov models

On the application of spectrum target prediction model to speech recognition

Rule-based speech analysis and application of CELP coding

Synthesis filter optimization and coding: Applications to CELP (speech analysis)

Strategies for improving the performance of CELP coders at low bit rates (speech analysis)

Improved speech quality and efficient vector quantization in SELP

CELP base-band coder for high quality speech coding at 9.6 to 2.4 kbps

Multiple-stage vector excitation coding of speech waveforms

High-quality vector adaptive transform coding at 4.8 kb/s

Modelling LPC-residue by components for good quality speech coding

2.4 kbps pitch prediction multi-pulse speech coding

Robust syntax free speech recognition

A large vocabulary word recognition system based on syllable recognition and nonlinear word matching

New speech recognition approaches based upon finite state vector quantization with structural constraints

Speech recognition with high recognition rate by smoothed space pseudo Wigner-Ville distribution (SSPWD) and overlap slide window spectrum methods

A new technique for speaker-independent isolated word recognition

Speaker-independent word recognition using dynamic programming matching with statistic time warping cost

Very large vocabulary isolated utterance recognition: a comparison between one pass and two pass strategies

Speaker adaptation method for HMM-based speech recognition

A large vocabulary word recognition system using rule-based network representation of acoustic characteristic variations

Speaker dependent and independent speech recognition experiments with an auditory model

Optimization of perceptually-based ASR front-end (automatic speech recognition)

Spectral movement function and its application to speech recognition

Speech codec for the European mobile radio system

Medium band speech coding for mobile radio communications

A sub-band coder designed for combined source and channel coding (speech coding)

Impact of noise and encoder/decoder mistracking on ADPCM system performance

A low delay 16 kbits/sec speech coder

Predictive trellis coded quantization of speech

Backward adaptive tree coding of speech at 16 kbps

Decision trees for vector quantizer codebook searching

An 8 kbps TC-MQ (time domain compression ADPCM-MQ) speech codec

Experiments with a high quality, low complexity 4800 bps residual excited LPC (RELP) vocoder

Statistical language modeling using a small corpus from an application domain

Syntactic analysis for large vocabulary speech recognition using a context-free covering grammar

Parsing spoken phrases despite missing words

Interaction between fast lexical access and word verification in large vocabulary continuous speech recognition

Speaker stress-resistant continuous speech recognition

Selection of optimal features for phoneme recognition in different phonetic environments

Continuous speech recognition results of the BYBLOS system on the DARPA 1000-word resource management database

Parallel level-building on a tree machine (speech recognition)

An event-synchronous signal processing system for connected-speech recognition

A versatile speaker-dependent continuous speech understanding system

On short-time cepstra of voiced speech

Joint solutions for formant and pitch predictors in speech processing

Reconstruction of Arabic long vowels using time varying linear prediction technique

An analysis-synthesis hidden Markov model of speech

Event-based multiple-resolution analysis of speech signals

Acoustic-phonetic analysis of loud and Lombard speech in simulated cockpit conditions

Adaptive L/sub p/ approach to speech coding

Phase compensation in all-pole speech analysis

Adaptive speech modification by spectral warping

Continuously adaptive linear predictive coding of speech

Decomposition of the speech signal into short-time waveforms using spectral segmentation

On estimating rate of change of pitch

The spectral autocorrelation applied to the linear prediction residual of speech for robust pitch detection

A geometric approach to real time pitch detection

A robust pitch boundary detector

Computationally efficient sine-wave synthesis and its application to sinusoidal transform coding

A 4.8 kbps multi-band excitation speech coder

Parametric models of the magnitude/phase spectrum for harmonic speech coding

Quantization issues in harmonic coders (speech coding)

Hardware implementation of a 16 kbps subband coder using vector quantization

Analysis of the self-excited subband coder: a new approach to medium band speech coding

Optimal quantization of LSP parameters (speech coding)

Quantizer design in LSP speech analysis and synthesis

Encoding of LPC spectral parameters using switched-adaptive interframe vector prediction (speech coding)

Multiply occupied cells (speech codecs)

A network-based frame-synchronous level building algorithm for connected word recognition

Experimental results on large-vocabulary continuous speech recognition and understanding

Learning spectral-temporal dependencies using connectionist networks

Some phonetic recognition experiments using artificial neural nets

Unstable connectionist networks in speech recognition

Multi-level acoustic segmentation of continuous speech

A procedure to generate training sequences for a connected word recognizer using the segmental k-means training algorithm

Phoneme modelling using continuous mixture densities

Speech recognition in VODIS II

Temporal decomposition and acoustic-phonetic decoding of speech

Broad class network generation using a combination of rules and statistics for speaker independent continuous speech

Segmentation and broad classification of continuous speech

Evaluating the performance of connected-word speech recognition systems

Transient part recognition for continuous speech using transition spotting

Automatic speech recognition using acoustic sub-words and no time alignment

An unanchored matching algorithm for lexical access

Connected sentence recognition using diphone-like templates

Frame compression in hidden Markov models

Noise compensation algorithms for use with hidden Markov model based speech recognition

A study of line spectrum pair frequencies for speech recognition

Obtaining candidate words by polling in a large vocabulary speech recognition system

A new algorithm for the estimation of hidden Markov model parameters

Acoustic Markov models used in the Tangora speech recognition system

A segment model based approach to speech recognition

Large vocabulary speech recognition using a hidden Markov model for acoustic/phonetic classification

Modeling acoustic-phonetic detail in an HMM-based large vocabulary speech recognizer

Three probabilistic language models for a large-vocabulary speech recognizer

Speech recognition using noise-adaptive prototypes

Adaptive labeling: normalization of speech by adaptive transformations based on vector quantization

The short-time modified coherence representation and its application for noisy speech recognition

Enhancing the performance of speech recognition with echo cancellation

On the application of hidden Markov models for enhancing noisy speech

Automated two speaker separation system

Estimation and vector quantization of noisy speech

A multipulse excited pole-zero filtering approach for speech enhancement

Speech enhancement using vector quantization and a formant distance measure

Noise reduction using connectionist models

A new class of sequential and adaptive algorithms with application to noise cancellation

Constrained iterative speech enhancement with application to automatic speech recognition

An approach to co-channel talker interference suppression using a sinusoidal model for speech

Optimal quantization of noisy sources

Frequency warping by dynamic programming

Frequency-warped spectral distance measures for speaker verification in noise

Text-dependent speaker identification using circular hidden Markov models

Variants of cepstrum based speaker identity verification

Computer assisted voice identification system (CAVIS)

Consistent speaker identification via Wigner smoothing techniques

Normalizations and selection of speech segments for speaker recognition scoring

A TMs32020-based real time, text-independent, automatic speaker verification system

Automatic recognition of gender by voice

Phonetic typewriter for Finnish and Japanese

Automatic alignment of speech with phonetic transcriptions in real time

Stochastic model of diphone-like segments based on trajectory concepts

Speech recognition with the discrete rectangular wave transform

Training of lexical models based on DTW-based parameter reestimation algorithm

A specialist society for continuous speech understanding

A 16 kb/s APC system with adaptive postfilter and evaluation of its performance

TMS 320C25 based enhanced 32 kbps ADPCM transcoder

Single chip multi-pulse coder with pitch prediction

Definition of relations in an acoustic phonetic database

A very large telephone-speech database collected using an automated voice-interactive dialogue

The DARPA 1000-word resource management database for continuous speech recognition

Voice conversion through vector quantization

Automatic generation of synthesis units based on context oriented clustering

Realization of linguistic information in the voice fundamental frequency contour of the spoken Japanese

Text-to-speech algorithms based on FFT synthesis

Speech production modelling with variable glottal reflection coefficient

Word synthesis based on line spectrum pairs

Speech synthesis by rule using an optimal selection of non-uniform synthesis units

Extracting speech-rate values from a real-speech database

A generalized vocal tract model for pole-zero type linear prediction (speech processing)

A comparative performance evaluation of adaptive ARMA spectral estimation methods for noisy speech

A comparison of five 16 kbps voice coding algorithms

A connectionist approach to speech recognition using peripheral auditory modelling

Speech signal reconstruction based on higher order spectra

Sampling schemes for reconstruction of multidimensional signals from multiple level threshold crossing

Phase unwrapping for multidimensional rational and finite-length sequences

ARMA modeling and phase reconstruction of multidimensional non-Gaussian processes using cumulants

High resolution model based 2-D spectrum estimation

Multi-dimensional power spectrum estimation using noncausal rational models

An efficient maximum entropy technique for 2-D isotropic random fields

A nonlinear simplex algorithm for minimum order solutions

Multidimensional algorithms for VLSI processors

Subband coding of color images using finite state vector quantization

An optimal bit allocation algorithm for sub-band coding

Sub-band coding of digital images using symmetric short kernel filters and arithmetic coding techniques

Image coding based on selective quantization of the reconstruction noise in the dominant sub-band

Tree coding of image sub-bands

Variable block rate and blockwise spectral adaptation in cosine transform image coding

Transform image coding with a new family of models

Reduction of blocking effects in image coding with a lapped orthogonal transform

Interpolative, predictive and pyramid transform coding of color images

Transform coding of images using directionally adaptive vector quantization

Primitives chain code (image analysis)

Application of multichannel 2-D linear prediction to color image coding

Low bit-rate image coding using 2-D linear prediction and 2-D stochastic excitation

Sketch based and context sensitive coding of Laplacian pyramids

An adaptive pyramid image coding system

Pyramid transform coding using vector quantization

Vector quantization of color digital images within a human visual model

Optimal order statistic filters with coefficient censoring

Multilevel median filters: properties and efficacy

Nonlinear recursive filters and applications to image processing

A novel method for nonuniform frequency-sampling design of 2-D FIR filters

Surface fitting and compression of two-dimensional scattered data

Boundary value transient suppression for N-D digital systems

Design of 2-D digital filters with finite wordlength coefficients

Approximation and implementation of multidimensional digital filters

A simple circuit-theoretical approach for the stability testing of 2-D digital filters

Conditions for the 2-D characteristic polynomial of a matrix to be very strict Hurwitz

Remote image processing in a workstation environment

Reconstruction techniques for the inspection of composite materials using thermal images

A comparison of wavelet deconvolution techniques for ultrasonic NDT

Geometry-free X-ray reconstruction using the theory of convex projections

A robust iterative spatial algorithm for correcting the transmission attenuation and detector response in single photon emission tomography

A knowledge-based multi-grid ML reconstruction algorithm for positron emission tomography

Automatic segmentation of biomedical images

3-D electron microscopy and incomplete angular coverage: a restoration scheme based on projections onto convex sets

The missing cone problem and low-pass distortion in optical serial sectioning microscopy

Seismic migration algorithms using the FFT approach on the NCUBE multiprocessor

Convolution decomposition of 1-D and 2-D linear stationary signals

Transient effects at a velocity discontinuity in up/down filtering

The cross Wigner-Ville distribution-a two dimensional analysis method for the processing of Vibroseis seismic signals

Coherent noise suppression in seismic data using eigenvalue/eigenvector decomposition

Using vision technique for the bridge deformation detection

Matched filter design for fingerprint image enhancement

A new approach to image registration utilising multidimensional LMS adaptive filters

Halftone patterns for arbitrary screen periodicities

Recognition of handwritten script: a hidden Markov model based approach

Bayesian imaging using Good's roughness measure-implementation on a massively parallel processor

Neural networks for planar shape classification

Stereo matching using a neural network

Symbolic signal representation using nonlinear filtering

Morphological skeleton representation and shape recognition

Mathematical morphology for 3-D image analysis

A representation for the orientation-dependent appearance of 3-D objects

Shape matching using curvature processes

Shape analysis for object recognition using number theoretic transforms

On the application of autoregressive models to the representation of planar curves

Maximum likelihood decision for recognition of noisy shapes

Hierarchical shape recognition using polygon approximation and dynamic alignment

Decision-directed segmentation for the restoration of images degraded by a class of space-variant blurs

Nonlinear image restoration using a segmentation-oriented expert system

Comparative study of some recent statistical and set-theoretic methods for image restoration

Iterative identification and restoration of images

Recovering images from partial information using Delta /sup 2/G filtering

Image restoration using a reduced order model Kalman filter

Restoration of degraded images using Markov random fields

Restoration of images with nonstationary mean and autocorrelation

Ringing reduction in images restoration using mirror images and adaptive Kalman filtering

Image estimation by stochastic relaxation in the compound Gaussian case

Mutually compatible Gibbs images: properties, simulation and identification

2-D Kalman filtering for the restoration of stochastically blurred images

Iterative color image restoration algorithms

State space approach to constrained recursive deconvolution of a noisy image sequence

Long correlation random field image models

Maximum likelihood estimation of Gaussian Markov random field parameters

Renormalization group approach to hierarchical image analysis

Regularized edge detection

Detection and smoothing of edge contours in images by one dimensional Kalman techniques

An ASIC architecture for contour line filtering

Generalized Levinson and fast Cholesky algorithms for three-dimensional random field estimation problems

Quality measures for SAR images

Detection of small moving objects in image sequences using multistage hypothesis testing

Estimation of image motion fields: Bayesian formulation and stochastic solution

Segmentation and 2-D motion estimation of noisy image sequences

Automatic detection of image impairments

Change detection for target detection and classification in video sequences

Real-time automatic target tracking based on spatio-temporal gradient method with generalized least square estimation

Video data compression using three-dimensional segmentation based on HVS properties

Space-variant regular decomposition quadtrees in adaptive interframe coding

Three dimensional sub-band coding of video

Post- and pre-processing in coding of image sequences using filters with motion compensated history

Segmented video coding

Texture classification based on higher-order fractals

Fractal dimension estimation by adaptive mask selection

Stochastic and deterministic algorithms for MAP texture segmentation

Segmentation of textured images using a multiple resolution approach

Textured image recognition using hidden Markov model

A fast technique for automatic segmentation and classification of textured images

Multi-slice image texture edge detection by local vector mapping

Quadratic linear filters for signal detection and image segmentation

Application of Markov random fields to smoothing and segmentation of noisy pictures

A model-fitting approach to cluster validation with application to stochastic model-based image segmentation

Image edge detection and segmentation based on the Hilbert transform

Variable-rate multi-stage vector quantization for image coding

Variable bit rate vector quantization of video images for packet-switched networks

High quality image coding with a model-testing vector quantizer and a human visual system model

A fast quantization algorithm for lattice quantizer design

An adaptive image coding algorithm based on noncausal LPC and vector quantization

Simulated annealing and codebook design

New finite state vector quantizers for images

Adaptive transform coding with multi-band vector quantization for video signals

A finite state/frame difference interpolative vector quantizer for low rate image sequence coding

Tomographic processing of spotlight-mode synthetic aperture radar signals with compensation for wavefront curvature

Radar imaging based on rotating antenna

Efficient order recursive LS algorithms for 2D SAR model fitting

Segmentation of multi-look speckled images

Application of migration to ground probing radars

Application of the Hough transform to Doppler-time image processing

Corner extraction from radar images

Convex boundary arrays for coherent and incoherent imaging

Signal abstractions in the machine analysis of radar signals for ice profiling

A new approach to near optimum MTI filter design using adaptive techniques

A new signal subspace approach for angle of arrival estimation in radar low-angle environment

Application of an autoregressive reflection model for the signal analysis of radar echoes from rotating objects

Adaptive signal processing and mapping in Doppler radar

Multi-channel sampling and reconstruction of bandlimited two-dimensional signals

Broad-band generalized sidelobe canceller using multichannel least-squares lattice structure

Cumulant based parameter estimation of multichannel moving-average processes

Robust detection of object boundaries in Weibull radar imagery

On the decorrelation of scatter using forward-backward averaging

A minimax approach to development of robust discrimination algorithms for multivariate mixture distributions

A projection space map method for limited angle reconstruction

High quality CT image reconstruction from a small number of projections

Tomographic image reconstruction from hollow projections

Binary image reconstruction from four projections

Performance of maximum entropy algorithms for image reconstruction from projections

Reconstruction from samples on linear and expanding spiral scans

Image reconstructions from limited-data with histogram constraint

Automated 3D reconstruction of tree-like structures from two orthogonal views

Iterative image restoration with ringing suppression using the method of POCS

Reconstruction from incomplete data in echo diffraction imaging

Linear block prediction with source classification for image encoding applications

Error resilience of a video codec for low bitrates

Optimal deconvolution filter for high-resolution NMR spectral parameter estimation

Edge-preserving image coding based on local modeling of images

A motion compensated vector quantizer with filtered prediction

Stability analysis of the fast RLS adaptation algorithm

Numerically stable fast recursive least-squares transversal filters

Stabilization of the RLS algorithm in the absence of persistent excitation

A new method to stabilize fast RLS algorithms based on a first-order of the propagation of numerical errors

A fast quasi-Newton adaptive filtering algorithm

The double/direct RLS lattice

A fast recursive least-squares second order Volterra filter

Effects of ill-conditioned data on least squares adaptive filters

Fast least squares adaptive notch filtering

Non-quadratic recursive algorithms (RLK) for transversal plant identification

FFT implementation on the TMS320C30

A connection between bit-reverse and matrix transpose, hardware and software consequences

An improved bit-reversal algorithm for the fast Fourier transform

Fast DFT algorithms for diagonally symmetric 2-D data

Split-radix algorithms for length-p/sup m/ DFTs

Error analysis and resulting structural improvements for fixed point FFTs

Minimization of rounding errors in WFTA programs

RNS Fourier transforms

FFT multi-frequency synthesizer

Wavelet transformations in signal detection

Minimum mean absolute error nonlinear filtering

Design of IIR filters in the complex domain

Design of ARMA digital filters with arbitrary log magnitude function by WLS techniques

Recursive digital filter design with linear phase

Linear phase realization of wave digital lattice filters

Designing FIR filters in a new vector space

Variable linear phase FIR filters

Eigenfilters for the design of special transfer functions with applications in multirate signal processing

Improved approach for design of perfect reconstruction FIR QMF banks, with lossless lattice structures

The design of tree-structured M-channel filter banks using perfect reconstruction filter blocks

Adaptive filtering via cumulants and LMS algorithm

A class of nonlinear adaptive filters

Nonlinear compensation for finite word length effects of an LMS echo canceller algorithm suitable for VLSI implementation

Nonlinear quantization effects in the LMS and block LMS adaptive algorithms-a comparison

Rejection of narrow-band interference in PN spread spectrum system using adaptive QC-LMSE filter

Convergence characteristics of LMS and LS adaptive algorithms for signals with rank-deficient correlation matrices

Stability of adaptive recursive filters

Stability of adaptive FIR filters

Adaptive rank-2 update algorithm for eigenvalue decomposition

Fractionally-spaced equalizers based on singular value decomposition

An adaptive edge enhancing order statistic filter

Frequency selective signal restoration using nonlinear combination filters

Relating analog and digital order statistic filters

Algorithms for real time trend detection

A first-order AR model for non-Gaussian time series

Adaptive estimation of a time-varying array shape using the tracking properties of the LMS algorithm

Time-varying coefficient tracking and noise suppression properties of a class of adaptive algorithms

Robust communication in a time-varying noisy environment

Two signal synthesis algorithms for pseudo Wigner distribution

On lattice-form algorithms for adaptive IIR filtering

FDM/TDM transmultiplexer design using adaptive filters

The composite regressor algorithm

On the performance surfaces of a frequency domain adaptive IIR filter

Counter examples to Stearn's conjecture on error surfaces of adaptive IIR filters

Adaptive filtering in sub-bands

Multiplication-free adaptive filters

Adaptive lattice filter implementations on pipelined multiprocessor architectures

High-speed systolic implementation of fast QR adaptive filters

Hybrid adaptive feedback nulling in the presence of channel mismatch

Saturation effects in the LMS adaptive line enhancer-transient response

Synchronous phase estimation of a signal corrupted by a close sinusoid

A new adaptive echo canceller used in the ISDN U-interface

Adaptive notch filtering in the presence of colored noise

The impact of an adaptive equalizer's update behavior on symbol error rate in a nonstationary environment

Analysis of bursting in telephony loops with adaptive hybrids

Efficient block implementation of exact sequential least-squares problems

A stable and distortion-free echo and howling canceller (telephone networks)

ARMA estimation using the general order two channel fast transversal filter (FTF) algorithm

An echo cancellation algorithm for operation with a digital speech coder in a single signal processor

Reflections on Levinson's recursion (lattice filters)

Inverse eigenvalue problem for real symmetric Toeplitz matrices

New split Levinson, Schur, and lattice algorithms for digital signal processing

Levinson and Schur algorithms for Toeplitz matrices with singular minors

An eigenvalue recursion for Hermitian Toeplitz matrices

Efficient procedures for stepup and stepdown computations

The lattice algorithm of linear prediction applied to the inverse scattering problem given transmission data

A rotation based method for solving covariance and related linear systems

Iterated Toeplitz approximation of covariance matrices

New connections between asymmetric Toeplitz matrix algorithms and inverse scattering

A graphics tool for model based signal processing

A practical C language compiler/optimizer for real-time implementations on a family of floating point DSPs

The application of high-level languages to single-chip digital signal processors

QuickSig-an object-oriented signal processing environment

An algorithm for order statistic determination and L-filtering

Experiments in partitioning and scheduling signal processing algorithms for parallel processing

Optimal multiprocessor implementations from a serial algorithm specification

A hypercube multiprocessor for digital signal processing algorithm research

DSP system architecture using signed-digit number representation

An ultra high-performance DSP vector processor

Reconfigurable pipeline for fast, versatile computation

Improved accuracy for logarithmic addition in DSP applications

A DFT using number theoretic logarithms

An all digital 8-DPSK TCM modem for land mobile satellite communications

Implementation of the CCITT wideband coder using a high performance, fixed point DSP

Real-time VQ codebook generation hardware for speech processing

An all digital automatic gain control

Parallel error correction algorithm in RNS VLSI digital circuits

Developments in techniques for enhancing the dynamic range of analog to digital converters

A lattice algorithm for Lp deconvolution

An efficient, noise tolerant, linear extrapolator

A band-limited signal extrapolation algorithm using pseudo inverse filtering and heuristic optimization

Bounds on restoration quality using a priori information

Interpolation of averaged data

Nonnegativity constrained spectral factorization for image reconstruction from autocorrelation data

A maximum a posteriori approach to the restoration of randomly distorted signals

A unified approach to iterative signal restoration

Television signal deghosting by noncausal recursive filtering

An exact numerical algorithm for computing the unwrapped phase of a finite-length sequence

Design of nearly perfect non uniform QMF filter banks

A generalized polyphase principle for uniform digital filter banks

Non convexity of the stability domain of discrete time linear filters

Signed binary delta modulation digital filter

New GIC digital filter structures and their characteristics

An efficient gradient-based iteration for direct form II sensitivity reduction

Proper scalings in recursive lattice digital filters for minimum round-off noises

The design of cascade form FIR filters for discrete space implementation

Effects of FFT coefficient quantization on sinusoidal signal detection

The quantization effects of the CORDIC algorithm (coordinate rotation digital computer)

On the sensitivity of transversal RLS algorithms to random perturbations in the filter coefficients

Low-sensitivity and low-roundoff noise multiple-loop feedback digital filters

Fast algorithms for wide sense stability of ARMA filters

Highly parallel algorithms for LS FIR smoothing and MEM spectral analysis

A single-channel ROM-based complex digital filter implementation in the quadratic residue number systems

A new structure for narrow transition band FIR filter design based on hardware

A new digital recursive filter structure based on delta modulation encoding

Novel decimators and interpolators using computationally efficient halfband subfilters

Multi-channel signal processing for data communications in the presence of crosstalk

Bounds on maximum throughput for digital communications with finite-precision and amplitude constraints

Analysis of delta-sigma modulators with bandlimited Gaussian inputs

Multi-loop sigma-delta quantization: spectral analysis

Optimal pulse shape design using projections onto convex sets

Constellation identification using the Radon transform

New DSP type phase synchronizer with the method of least squares

A new carrier frequency estimator for modem signals

Adaptive equalization for time division multiplex channels

Efficient computation of the short-time fast Fourier transform

A nesting algorithm for very fast discrete Fourier transforms

Fast algorithms for the real discrete Fourier transform

The complex form of a fast algorithm for frequency estimation

Fast convolution using generalized Fermat/Mersenne number transforms

A unified approach to the fast FIR filtering algorithms

On the fast computation of cross-covariance and auto-covariance sequences

Multi-dimensional systolic networks for DSP algorithms

Multichannel adaptive least squares-relating the 'Kalman' recursive least squares (RLS) and least squares lattice (LSL) adaptive algorithms

New data movement algorithms for processor arrays

Equivalences and optimal designs of PTV systems

A VLSI architecture for real-time image convolution with large symmetric kernels

A single chip VLSI architecture for a real time stereo vision processor

An expandable VLSI processor array approach to contour tracing

VLSI implementation of a 16*16 DCT

A chip set for real-time 20-MHz DSP

A load balancing technique for video signal processing on a multicomputer type DSP

Customized architectural methodologies for high-speed image and video processing

An application specific DSP chip set for 100 MHz data rates

A unified DCT/IDCT architecture for VLSI implementation

Implementation of 2-D DFT algorithms on a loosely-coupled parallel system

On the use of modulo arithmetic comb filters in sigma delta modulators

Monolithic decimation filtering for custom delta-sigma A/D converters

A floating point DSP with optimizing C compiler

Asynchronous processor design for digital signal processing

Code optimization technique for general-purpose-DSP compiler

Automatic generation of a custom digital signal processor for an adaptive robot arm controller

Converting sequential iterative algorithms to recurrent equations for automatic design of systolic arrays

On the optimization of regular wavefront arrays

Mapping algorithms to VLSI array processors

Mapping signal processing algorithms to fixed architectures

QR algorithm and array architectures for multichannel adaptive least squares lattice filters

SAC: a systolic array controller chip

A high throughput systolic implementation of the second order recursive filter

A reconfigurable systolic array for real-time image processing

Sparse matrix-vector multiplication on a systolic array

Implementation of factorized filter in systolic arrays

Wafer-scale systolic array for adaptive antenna processing

Systolic IIR filters with bit level pipelining

Reliable modulo systolic arrays for DSP algorithms

A VLSI design of processing element for reconfigurable systolic architectures based on LNS

Mapping linear algebra algorithms into array processors: a case study

Real time connected word recognition with finite state grammar on the graph search array processor

Parallel implementation of the tracking Kalman filter

A high-performance multiprocessor system for speech processing applications

A fault-tolerant bit-serial array structure for digital filters

Architecture and performance of a multicomputer type digital signal processing system 'NOVI'

A multiprocessor structure based on commercial DSP

VLSI architecture of a real-time Wigner distribution processor for acoustic signals

A new, highly parallel, 32 bit floating point DSP vector signal processor

Pipelined VLSI recursive filter architectures using scattered look-ahead and decomposition

Seismic migration algorithms on multiprocessors

Hardware approaches to vector plane rotation

VLSI structures for computing the Wigner distribution

On uniform one-chip VLSI design considerations for some discrete orthogonal transforms

An N-dimensional SIMD ring architecture for implementing very large order adaptive digital filters

DSP analog front end

A decision-theoretic performance benchmark for neural networks trained to discriminate two autoregressive processes

Computationally efficient tracking of multiple targets by probabilistic data association using neural networks

Bearing estimation using neural networks

Efficient modeling for multilayer feed-forward neural nets

Neural networks for narrowband and wideband direction finding

Neural net based pattern recognition on the graph search machine

A CMOS VLSI cochlea

Maximum signal energy concentration in a time-frequency domain

Tracking of unknown non-stationary chirp signals using unsupervised clustering in the Wigner distribution space

Simulation and analysis of Suzuki fading processes

Instantaneous mean qualities in time-frequency analysis

On a class of nonstationary signals

Time-frequency representations of broad-band signals

Time-varying modelling of arbitrary non-stationary signals

Decision- and classification-directed methods in nonstationary signal analysis

Unbiased parameter estimation of non-stationary signals on the block processing

Time frequency filtering

The Wigner distribution, multi-equilibria nonlinear systems and chaotic behaviour

Application of polynomial splines to a time-varying autoregressive model of speech

Optimal parameter tracking and control of nonlinear systems with time-varying parameters

New method for time-varying harmonic frequency tracking

A new system identification algorithm based on iterative algebraic filtering

Near optimal blind deconvolution

Robust parametric approach for impulse response estimation

A geometrical approach for the identification of state space models with singular value decomposition

Adaptive system identification using cumulants

Finite precision analysis of the lattice and the Schur algorithms for the autocorrelation method of linear prediction

Pole focusing: a new approach to LPC-based speech analysis offering superior formant resolution

A formant extraction method using autocorrelation domain inverse filtering and focusing method

Cumulant-based approach to the harmonic retrieval problem

A new class of nonlinearly constrained linear estimator

A method for constructing NARMAX models of a class of nonlinear systems

A test of nonlinear autoregressive models

Moving average separation

A new class of focussing matrices for coherent signal-subspace method

Stability of the linear prediction filter: a set theoretic approach

Statistically/computationally efficient frequency estimation

MUSIC, maximum likelihood and Cramer-Rao bound

Signal modeling with stochastic and deterministic data: a lossless inverse scattering approach

An aggregate group approach for AR spectral estimation of noisy signals

ARMA processes: order estimation

True lattice algorithms for square root solution of least squares linear prediction problems

Maximum entropy extrapolation of cumulant statistics: linear processes

Estimation of the highly damped sinusoidal signals in additive noise

Some structured matrix approximation problems

Robust linear prediction of band-limited signals

Determining the closest stable polynomial to an unstable one

Estimating the number of signals by chi-square statistics

On non-negative definiteness of estimated moving average autocovariance sequences

An approximate maximum likelihood ARMA estimator based on the power cepstrum

Parameter estimation of quasi-periodic sequences

An improved Toeplitz approximation method

Harmonic decomposition methods in cumulant domains (array processing)

The partial realization problem for moving average models

Recursive autoregressive spectral computations directly from reflection coefficients

High performance SVD-like procedure for spectral estimation using Rayleigh function estimates

On selection of maximum cumulant lags for noncausal autoregressive model fitting

On the asymptotic relative efficiency of the MUSIC algorithm

On the rate of convergence of the ML spectral estimate for identification of sinusoids in noise

An efficient algorithm for implementing the relative entropy method

A consequence of signal normalization in spectrum analysis

Optimized parametric bispectrum estimation

Maximum entropy spectrum estimation with imprecise autocorrelation function lags by fuzzy set theory

Volume of the space of autocorrelation sequences

Bispectrum estimation of transient signals

Statistical analysis of two non-linear least-squares estimators of sine waves parameters in the colored noise

Performance analysis of MA parameter estimation algorithms based on high-order moments

The threshold analysis of SVD-based algorithms

Successive correlation based Prony modeling very low SNR behavior analysis

Performance evaluation of an ARMA estimator

A performance comparison of smoothing approaches for high-resolution active direction-finding of completely-correlated targets

Regression surfaces for estimated reflection coefficients

Statistical properties of a novel recursive estimation algorithm with information-dependent updating

A new analysis method for acoustic signals composed of cosine waves

Toeplitz-Hankel decomposition for direction of arrival estimation using transient signals

Efficient conjugate directions method for adaptive eigen-spectrum estimation

On fundamental performance limitation of direction finding of multiple narrow-band sources under sensor position and pattern perturbations

ESPRIT and TAM (Toeplitz approximation method) are theoretically equivalent

A criterion for identifying dominant singular values in the SVD based method of harmonic retrieval

On harmonic analysis via Gram-Schmidt orthogonal procedure

Spectrum analysis using a new autocorrelation measure

Lowering the threshold SNR of singular value decomposition based methods

Matrix pencil method and its performance

Notes on eigenvalue distribution of Toeplitz matrices

An RML algorithm for retrieval of sinusoids with cascaded notch filters

A real-time digital speech processing system for an auditory prosthesis

Integrating uncertain speech evidence into a knowledge-based message generation system for the nonverbal, severely motor disabled

A new strategy for tactile hearing aids: tactile identification of preclassified signals (TIPS)

Computerized intonator: design of a microprocessor pitch-aid

Programmable master hearing aid with adaptive noise reduction using a TMS320-20

VLSI based design of a battery-operated digital hearing aid

Design and evolution of a powerful pocket-sized DSP speech processing system for a cochlear implant and other hearing prosthesis applications

The perceptual significance of inter-subject differences in the directional transfer function (DTF)

Estimation of perceptual entropy using noise masking criteria

Fast signal processor encodes 48 kHz/16-bit audio into 3-bit in real time

An independent evaluation of the performance of the CCITT G.722 wideband coding recommendation using music signals

High-quality sound coding within 2*64 kbit/s using instantaneous dynamic bit-allocation

Sub-band ADPCM coding for high quality audio signals

An independent evaluation of the performance of the CCITT G.722 wideband coding recommendation

A new application of adaptive filters for restoration of archived gramophone recordings

The architecture of DSP: a DSP multiprocessor

A real-time digital signal processing system for bioelectric control of music

Sinewave-based phase dispersion for audio preprocessing

The use of line spectrum pairs for the production of pipe organ-like sounds

Experiments to identify and track non stationarities in audio conference rooms

Analysis and design of multirate systems for cancellation of acoustical echoes

A new structure for sub-band acoustic echo canceler

A microphone array with adaptive post-filtering for noise reduction in reverberant rooms

Fast algorithms in active noise or vibration control

Synchronous adaptive filters with delayed coefficient adaption

Multichannel active sound control using adaptive filtering

Active noise control using adaptive digital signal processing

Hardware and software considerations for active noise control

Active systems for sound attenuation in ducts

Static and dynamic noise transmission over an oxygen mask (military aircrew applications)

A constrained frequency-domain Prony's algorithm with application to time delay estimation

Time delay estimation for Poisson derived processes

Bearing resolution limits for broadband signals

Optimal estimation of time-varying delay

Path resolution by coherent averaging: trading spatial and temporal degrees of freedom

An asymptotic maximum likelihood estimator for source parameters in a multipath environment

An alternative method for locating a passive source by using the bearing estimations

Time delay estimation in unknown Gaussian spatially correlated noise

Cumulant cepstrum of FM signals and high-resolution time delay estimation

Adaptive calibration of randomly perturbed sensor arrays

Digital realizations of the real-time coherent signal-subspace transformation for array signal processing

Broadband source direction estimation using a circular array

High-resolution broadband beamforming using a doubly-steered coherent signal-subspace approach

Resolution of spatially close broadband coherent sources incident on a general array

Wideband direction-of-arrival estimation using the ESPRIT algorithm

Array shape calibration using sources in unknown locations-a maximum likelihood approach

An algorithm for optimal placement of sensor elements

Resolution of coherent signals using random array

Eigenstructure methods for direction finding with sensor gain and phase uncertainties

Segmentation of non-stationary signals with applications

Classification of transient signals (acoustic signals)

Experiments on automatic classification of shallow water acoustic signal sources using two pattern recognition methods

An approach towards improving the efficiency of sonar projector arrays

Element-level reverberation time series synthesis (sonar)

Seismic profiling as a contaminant in the measurement of ambient noise spectral and spatial characteristics

Incoherent coding techniques and performance characterization for multibeam sonar systems

The modified Wigner distribution with application to acoustic well logging

Multipaths resolution by a combination of homomorphic and Wiener filtering

Performance of a slow frequency hopping BFSK system using convolutional coding in underwater acoustic media

Time-frequency receivers for locally optimum detection

Entropy-based receiver for detection of random signals

Noise background normalization for simultaneous broadband and narrowband detection

Estimating steady-state response of a resonant transducer in a reverberant underwater environment

Maximum likelihood parameter estimation by simulated annealing

Generalized coherence (signal detection)

Estimating time delay and transfer function parameters using wideband transient signals

Broadband beamforming on the 2-D transform plane

Adaptive beamforming for correlated signal and interferences: A frequency-domain smoothing approach

Multipath propagation and the optimal SNR for adaptive arrays

A multimode adaptive beamformer for quadrature-amplitude-modulated signals

A new criterion for the determination of the number of signals in high-resolution array processing

Detection: a new approach (signals)

Detection of fully correlated signals by the MDL principle

Linear-quadratic array processing for coherent sources

Systolic implementation of a general adaptive array processing algorithm

A pole-zero complex adaptive joint process least-squares lattice

Improved power minimization based partially adaptive beamformer design

Optimal projection for multidimensional signal detection

Robustness of a CFAR detector to errors on the signal

Performance of redundant and distributed detection systems with processor faults

A simple algorithm for broadband detection and sorting

Representing sonar/radar returns as a Markov process

Time-frequency analysis applied to signaturing of underwater acoustic signals

Detection and estimation of transient signals in coloured Gaussian noise

Threshold extension of SVD-based algorithms

An efficient signal subspace algorithm for source localization in noise fields with unknown covariance

On least squares methods for direction of arrival estimation in the presence of unknown noise fields

Active high-resolution multiple-target direction finding

Minimum variance spectral estimation for broadband source location using steered covariance matrices

Robust maximum-likelihood estimation of structured covariance matrices

Direction finding in the presence of fully correlated specular multipath

On a modified Duvall beamformer

A structured gradient algorithm for adaptive beamforming

Novel techniques for estimation of array signal parameters based on matrix pencils, subspace rotations, and total least squares

The effects of spatial averaging on coherence and resolution

A solution space approach to achieving partially adaptive arrays

Perturbation analysis of the MUSIC algorithm

Iterative pre-filtering over non-uniform sampling grids

More on optimality of EVD methods for bearing estimation

Sector interpolation of non-uniform arrays for efficient high resolution bearing estimation

New maximum likelihood approach to multiple signal estimation

Maximum likelihood direction finding of stochastic sources: a separable solution

An adaptive recursive array processing algorithm for resolving closely spaced signals with low SNR and widely disparate signal power

Improved maximum likelihood spectrum for direction of arrival (DOA) estimation

Separately estimating the frequency-wavenumber of coherent sources by signal eigenvector smoothing technique

Estimating two-dimensional angles of arrival in coherent source environment

Signal subspace algorithms for direction of arrival estimation in the presence of a class of unknown noise fields

Calibration-free bearing estimation for arrays with randomly perturbed sensor locations

Active control of vibrations in spacecraft

Transient signal enhancement by linear predictive eigenanalysis

Formant tracking with quasilinearization

An efficient triarray systolic design for real-time Kalman filtering

ICASSP 88: 1988 International Conference on Acoustics, Speech, and Signal Processing (Cat. No.88CH2561-9)

Auditory neural feedback as a basis for speech 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

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

A 4.8 kbps multi-band excitation speech coder

Content Provider	IEEE Xplore Digital Library
Author	Hardwick, J.C. Lim, J.S.
Copyright Year	1988
Description	Author affiliation: Res. Lab. of Electron., MIT, Cambridge, MA, USA (Hardwick, J.C.; Lim, J.S.)
Abstract	A speech model, referred to as the multiband excitation (MBE) speech model, has been shown to be capable of synthesizing speech without the artifacts common to model-based speech systems and has been used to develop a 4.8 kb/s speech coder. This system was developed using several new approaches to quantize the MBE model parameters. These techniques were designed to utilize additional redundancy amongst these parameters, thereby permitting more efficient quantization. The result of information listening tests indicate that this system can achieve high quality for both clean and noisy speech as the MBE speech is extremely robust to the presence of background noise in speech.<>
Starting Page	374
Ending Page	377
File Size	406469
Page Count	4
File Format	PDF
ISSN	15206149
DOI	10.1109/ICASSP.1988.196595
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1988-04-11
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Speech synthesis Vocoders Speech coding Speech analysis Speech enhancement Background noise Signal synthesis Noise robustness Parameter estimation Laboratories
Content Type	Text
Resource Type	Article

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