NDLI: Composite permutation coding of speech waveforms

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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

Volume 11 - 1986

Digital audio magnetic recording: Progress in packing density and key technology

Perpendicular magnetic recording and its application to digital recording

Error correction method for R-DAT and its evaluation

Efficient data reduction for digital audio using a digital filter array

Digital data equalizing in multitrack digital audio recording

A comparative study of the proposed high quality coding schemes for digital music

Permissible value of group delay distortion on tone quality due to low-pass filters

A study on the digitization of audio signals for video tape recorder

An audio sampling frequency conversion using digital signal processors

An approach to high resolution D/A converter utilizing a linear predictive coding

Mixture autoregressive hidden Markov models for speaker independent isolated word recognition

Markov modeling of continuous parameters in speech recognition

Maximum mutual information estimation of hidden Markov model parameters for speech recognition

An IBM PC based large-vocabulary isolated-utterance speech recognizer

A syllable-based isolated word recognition experiment

Quantitative knowledge on word structure, from a phonetic corpus, with application to large vocabularies recognition systems

Word preselection for large vocabulary speech recognition

Compact isolated word recognition system for large vocabulary

Coarse classification using a hierarchical decision tree and top down parsing

Incremental network generation in word recognition

A computational model for separating two simultaneous talkers

Evaluation of multisensor speech input for speech recognition in high ambient noise

Enhancement of noisy speech by forward/Backward adaptive digital filtering

Performance characteristics of a hardware implementation of the cross-talk resistant adaptive noise canceller

Acoustic noise reduction by two dimensional spectral smoothing and spectral amplitude transformation

Speech enhancement using multi-pulse excited linear prediction system

Waveform substitution techniques for recovering missing speech segments in packet voice communications

Fast and accurate pitch detection using pattern recognition and adaptive time-domain analysis

Pitch detection using the short-term phase spectrum

Frequency identification by complex spectrum

An adaptive non-uniform sign clipping preprocessor (ANUSC) for real-time autocorrelative pitch detection

A high quality 9.6 kbps speech coding system

Adaptive discrete cosine transform image coding using gain/Shape vector quantizers

Vector quantization of color images

Low-rate image coding with finite-state vector quantization

A gain-shape vector quantizer for image coding

Vector quantization in transformed image coding

A technique using a one-dimensional mapping for vector quantisation of images

A hybrid coding involving ADM and vector quantization for digital video image compression

The performance of a hybrid videoconferencing coder using displacement estimation in the transform domain

Contour based representation of the displacement field for motion compensated image coding

Hybrid interframe coding of video signals with backward-acting motion detection

High-resolution spectral estimation: Rethinking the Fourier transform

Single vector approaches to eigenstructure analysis for harmonic retrieval

Estimation of frequencies of sinusoids in colored noise

High resolution spectral estimation of sinusoids in colored noise using a modified Pisarenko decomposition

Spectral and parameter estimation problems arising in the metrology of high performance mirror surfaces

On the use of linear programming for spectral estimation

A Toeplitz approximation approach to coherent source direction finding

Performance of multichannel autoregressive spectral estimators

Linear predictive spectral analysis via the Lpnorm

Refined ARMA digital lattice filter

ARMA parameter estimation based on sample covariances, for missing data

A new FFT approach to the interpolation of discrete-time signals

Fast algorithms for the discrete Fourier transform and for other transforms

Performance of fixed-point FFT's: Rounding and scaling considerations

A fast Fourier transform algorithm using Hadamard transform

Cyclic convolution of real sequences: Hartley versus Fourier and new schemes

Computation of complex number theoretic transforms using quadratic residue number systems

A hardware efficient realisation of number theoretic convolvers

Half-Fourier transform and application to radar signals

Optimal design of multiplierless DFTS and FFTS

An efficient vector implementation of the FFT algorithm on IBM 3090VF

Immitance-domain Levinson algorithms

Efficient multi-processor implementation of recursive digital filters

A unified derivation of the fast RLS algorithms

A fast codebook search algorithm for nearest-neighbor pattern matching

The continuous-time limit of the discrete-time stability theory

Maximum entropy deconvolution

L1-norm noisy Tauberian deconvolution

A new homomorphic deconvolution system

Generalized Lanczos method for signal smoothing

FFT and convolution algorithms on DSP microprocessors

A unified transform architecture

The array matrix generalization for signal processing

A fast B-spline transform and its applications

A generalization of Levinson algorithm for solving Toeplitz systems

A demiphoneme network representation of speech and automatic labeling techniques for speech data base construction

A speech data base at the united states air force academy

A PCM/VCR speech database exchange format

A Japanese language speech database

B.D.L.E.X. : A data and cognition base of spoken French

The development of the MIT Lisp-machine based speech research workstation

A microprocessor-based word recognition system for large vocabularies

Signal processor application to voice dialing equipment

The analog voice privacy system

Development of a 16-kb/s speech codec for telephone applications

24-channel 32 kb/s ADPCM transcoder using the CCITT recommendation G.721

A special architecture for dynamic programming

A modular architecture for dynamic programming and maximum likelihood sequence estimation

Speech recognition on the DADO/DSP multiprocessor

A new systolic decomposition for the dynamic time warping algorithm

VDP : A versatile high performance vector distance processor

Design of an efficient dynamic time warping LSI

A single-chip speaker independent voice recognition system

VLSI Architecture for a real-time LPC-based feature extractor

On the IC architecture and design of a 2 µm CMOS 8 MIPS digital signal processor with parallel processing capability: The PCB5010/5011

VLSI Digital signal processor (PSI)

Architecture and applications of a 100-ns CMOS VLSI digital signal processor

A high performance microprocessor for DSP applications

A 50ns floating-point signal processor VLSI

Architecture of high-speed 22-bit floating-point digital signal processor

Advanced single-chip signal processor

A next-generation 32-bit VLSI signal processor

Architectural considerations for a sub 10 nanosecond DSP building block family

The architecture, instruction set and development support for the $WE^{®}DSP32$ digital signal processor

Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3)

A high performance digital voice echo canceller on a single TMS32020

Hidden Markov models applied to very low bit rate speech coding

Design and performance of trellis vector quantizers for speech signals

Joint time-spectral vector quantization and inverse filter set

Fully vector-quantized multipulse LPC at 4800 bps

Low-rate speech encoding using vector quantization and subband coding

The self excited vocoder - an alternate approach to toll quality at 4800 bps

Low bit rate multi-pulse speech coder with natural speech quality

A novel LPC synthesis model using a binary pulse source excitation

High quality glottal LPC-vocoding

On the behaviour of reduced complexity code-excited linear prediction (CELP)

Voiced/Unvoiced classification of speech with applications to the U.S. government LPC-10E algorithm

Optical circuitry and architectures for digital optical computing

Optical parallel array logic system

Use of optical signal processing techniques to spectrum analysis of speech

Optical signal processors consisting of collinear acousto-optic channel waveguides

Reproduction of the sounds from old wax phonographic cylinders using the laser-beam reflection method

New cascaded lattice structures for FIR filters having extremely low coefficent sensitivity

Synthesis of minimum sensitivity structures in linear systems using controllability and observability measures

Embedded max quantization

Nonlinear quantization and data communication

Floating point error analysis of recursive least squares and least means squares adaptive filters

Design of limit-cycle-free digital biquad filters

Predictive coding of multi-viewpoint image sets

An architecture for the universal video codec

A fractal based approach to image compression

Bit-sliced progressive transmission and reconstruction of transformed images

A binary representation of mixed documents (text/Graphic/Image) that compresses

A representation method of color pictures by arranging fixed size dots of the primary colors

Design of 3-D IIR filters via transformations of 2-D circularly symmetric rotated filters

A computer program for designing optimum 2-D FIR linear phase digital filters

An algorithm for the design of optimal finite wordlength 2-D FIR digital filters

Reduction of the number of parameters in the design of 2-D digital filters

On the stability test for 2-D digital recursive filters

A fast image filtering processor-FIFP

Lower bounds in parameter estimation - summary of results

New forms of least squares lattice algorithms and a comparison of their round-off error characteristics

Effective adaptive Pisarenko spectrum estimate

Determining MA models as salvos of pulses

On the limiting behavior of estimates based on sample covariances

Performance of narrowband signal-subspace processing

Phase-strip sequence estimation: A phase unwrapping/Tracking algorithm

A new functional method of parameter estimation for the energy acoustic system based on an Lxevaluation criterion

Estimation of anti-resonance frequencies by using an over-determined high-order Yule-Walker equation

A new methodological trial on state estimation of linear structure vibration model with noisy power observation mechanism of non-Gaussian type

Asymptotic properties of high-order Yule-Walker estimates of frequencies of multiple sinusoids

Assessment of speech discrimination ability of hearing impaired subjects using FIR digital filter

A speech processor with lateral inhibition for an eight channel cochlea implant and its evaluation by subject testing

Sound processing for cochlear implant

A wearable, pocket-sized processor for digital hearing aid and other hearing prostheses applications

Assessment of electronic aids for the hearing impaired which transmit visible or tactile speech

Speech training devices for profoundly deaf children

Speech training systems for handicapped children using vocal tract lateral shapes

Speech-analysis-based devices for diagnosis and education of speech and hearing impaired people

Refutation based recognition to help vowel articulation

Memory-intensive recognition for word articulation training

Speech control of assistive devices for the physically disabled

Integrated communication aids for the blind

Japanese speech synthesis system in a book reader for the blind

A lexical prediction system

An adaptive comb filtering method as applied to acoustic analyses of pathological voice

Multidimensional analysis of phonological degeneration in pathological voices

Acoustic measurement of pathological voice qualities for medical purposes

A microcomputer based system for acoustic analysis of voice characteristics

Wave-flow index: A measure of vocal efficiency

Electroglottography: Assessing articulatory function to aid the handicapped

Analysis and classification of snoring signals

Speaker independent isolated Arabic word recognition system

Isolated-word recognition of the complete vocabulary of spoken Chinese

On hidden Markov models in isolated word recognition

Large-vocabulary isolated word recognition with fast coarse time alignment

A proposal of a knowledge based isolated word recognition

Isolated word recognition based on finite-state vector quantization

Speech processing with a Boltzmann machine

Simulating an acoustic recognizer

Experiments in speech recognition over the telephone network

Recognition of speech under stress and in noise

Word recognition using multisensor speech input in high ambient noise

Noise compensation for speech recognition using probabilistic models

Optimal and suboptimal training strategies for automatic speech recognition in noise, and the effects of adaptation on performance

Improved speech recognition in noise

Clustering acoustic prototypes with self organizing distortion measures

Spectral slope based distortion measures for all-pole models of speech

A weighted cepstral distance measure for speech recognition

On the use of bandpass liftering in speech recognition

Comparative study of various spectrum matching measures on noise robustness

An adaptive filtering approach for speaker independent vowel recognition

Systolic image processor for automatic detection and recognition of traffic violations

Image processing on linear transputer arrays

Data flow chip ImPP and its system for image processing

Construction methods and performance evaluation for image processing systems with ISP LSIs

Architecture of a real-time video signal processor

Video signal processor configuration by multiprocessor approach

A custom chip set for real-time image processing

A single chip video rate 16×16 discrete cosine transform

Video signal processing LSI and its application to TV CODEC

A study of VLSI design for DPCM coding

CCITT Standardizing activities on speech coding

An advanced 32 kbit/s ADPCM coding to transmit speech and high-speed voiceband data

A real-time ADPCM encoder using variable order prediction

Adaptive postfiltering of 16 kb/s-ADPCM speech

Amplitude normalization and its application to speech coding

Range extended speech decoder for Modulo-PCM

Considerations on quantization and dynamic bit-allocation in subband coders

16 kbit/s Split-band APC coder using vector quantization and dynamic bit allocation

Adaptive predictive coding with dynamic quantization adjustment (APC-DQA) at 16 kbits/s

Performance of 16 kbit/s APC codec with maximum likelihood quantization for low-speed voiceband data

A robust 16 kbits/s vector adaptive predictive coder for mobile communications

Language identification using noisy speech

Methods and experiments for text-independent speaker recognition over telephone channels

A new method of text-independent speaker recognition

Evaluation of a vector quantization talker recognition system in text independent and text dependent modes

On the use of instantaneous and transitional spectral information in speaker recognition

High performance speaker verification using principal spectral components

The effects of voice disguise upon formant transition

Contributions of pitch, formant frequency and bandwidth to the perception of voice-personality

The effect of formant trajectory and spectral shape on the tense/Lax distinction in American vowels

A pitch perception model

Comparative evaluation of the speech quality of speech coders and text-to-speech synthesizers

Acoustical transformation technique as a practical tool

Grasp and development of spatial informations in a room by closely located four-point microphone method

Acoustic noise cancellation

Inverse control of room acoustics using multiple loudspeakers and/Or microphones

Modelling of reverberations and audioconference rooms

Extraction of fundamental physical parameters in the sound field by multidimensional analysis

Impulse response measurement using Golay codes

A new method to locate sound sources by searching the minimum value of error function

Headphone responses on real ears and a head and torso simulator

Pre-howling howlback detection method

Octave and fractional octave band digital filtering based on the proposed ANSI standard

Data equalization based on the constant modulus adaptive filter

The least-squares CMA: A new technique for rapid correction of constant modulus signals

An adaptive filter convergence method for echo cancellation and decision feedback equalization

Convergence of an adaptive echo cancellation system with an augmented predictor

An adaptive echo canceller using parallel Kalman filters

Adaptive rate conversion filter

Interference cancellation for enhanced detection of frequency hopped signals

Noise cancellation for hearing aids

Adaptive optimization of microphone arrays under a nonlinear constraint

Adaptive discrete cosine transform coding with vector quantization for color images

Motion compensated vector quantization

A novel method for constructing a set of Hadamard matrices

A nearly optimal transformation

Two-dimensional transform domain decimation techniques

Sub-band coding of images

Contour coding of images

An adaptive state-prediction digital facsimile coding technique

Proposal of line element chain coding

Two alternatives for motion compensated coding of composite NTSC video signal

2-D Multilevel FIR-median hybrid filters

High speed delayed multipath 2-D digital filter structures

Optimized shift-varying filters for image interpolation

New interpolatory approximation method with application to the design of multi-dimensional FIR filters

2D Spectral factorization and stability test for 2D matrix polynomials based on the radon projection

A unified study on the roundoff noise in 2-D state space digital filters

A generalized state-space model for noncausal 2-D systems

On compression of multi-level document images

3-D Lattice predictive modeling of random fields

Theory and realization of M-D nonlinear digital filters

A continuous training procedure for connected digit recognition

An efficient algorithm for combining vector quantization and stochastic modeling for speaker-independent speech recognition

Connected digit speech recognition by using demi-word pair reference pattern (DWPR)

A speaker independent recognition algorithm for connected word using word boundary hypothesizer

Global connected digit recognition using Baum-Welch algorithm

Fast feature-based preclassification of segments in continuous digit recognition

Lexical access and verification in a broad phonetic approach to continuous digit recognition

Speaker-independent recognition of connected Swedish digits

Network-based connected digit recognition using explicit acoustic-phonetic modeling

Connected word recognition by overlap and split of reference patterns and its performance evaluation tests

Two novel algorithms for variable frame analysis and word matching for connected word recognition

A new network-based speaker-independent connected-word recognition system

New DP matching algorithms for connected word recognition

Syllable-based connected spoken word recognition by two pass O(n) DP matching and hidden Markov models

Voice-activated word processor with automatic learning for dynamic optimization of syllable-templates

Japanese text input system based on continuous speech recognition

A "Bunsetsu" speech recognition system using a top-down processor controlled by bottom-up information

Speech understanding system using knowledge engineering techniques

A linguistic processor for Japanese continuous speech recognition

A grammatical approach to reducing the statistical sparsity of language models in natural domains

N-gram driven search for sentences in a syntactic network

On the analysis of systolic arrays

Implementations of arbitrarily fast adaptive lattice filters with multiple slow processing elements

A systolic discrete Fourier transform using residue number systems over the ring of Gaussian integers

A high speed CMOS/SOS implementation of a bit level systolic correlator

Systolic implementation for deconvolution iterative algorithm

Doubly pipelined Cordic array for digital signal processing algorithms

High speed signal processing, pipelining, and VLSI

A dataflow algorithm for digital filtering

A round-robin architecture for relaxation operation

A parallel architecture for recursive least square identification

Fault-tolerant digital filtering structures for wafer scale VLSI

An expert system for speech spectrogram reading

An expert spectrogram reader: A knowledge-based approach to speech recognition

APHODEX, design and implementation of an acoustic-phonetic decoding expert system

A representational approach to knowledge-based acoustic-phonetic processing in speech recognition

Producing and organizing phonetic knowledge from acoustics facts in multi-level data-information

A hybrid recogniser for speech patterns

Plan refinement in a knowledge-based system for automatic speech recognition

A continuous speech recognition system based on knowledge engineering techniques

A family of formant trackers based on hidden Markov models

A new algorithm for estimation of formant trajectories directly from the speech signal based on an extended Kalman-filter

A background for sinusoid based representation of voiced speech

A speech waveform analysis and reconstruction process based on non-euclidean error minimization and matrix array procesing techniques

Continuously variable duration hidden Markov models for speech analysis

Analysis of speech signals of short pitch period by the sample-selective linear prediction

Efficient implementation of the multipulse LPC method for speech

A speech analysis-synthesis system based on the ARMA model and its evaluation

Analysis of time fluctuating characteristics of linear predictive coefficients

Duality theory of composite sinusoidal modeling and linear prediction

Optimal speech data reduction using a stochastic realization algorithm

Musical sound models for digital synthesis

A signal source in a digital musical synthesizer

Transcription of sung song

A study on annoyance of musical signal using LAeq measurement and digital signal processing

Model analysis of a hammer-string interaction

Source separation and note identification in polyphonic music

Piano tone synthesis using digital filters by computer simulation

Digital and analog bows: Hybrid mechanical-electrical systems

Hardware implementation of FM multipath distortion canceller

Multipath echo cancellation in FM transmission using adaptive FIR filters

Echo return loss required for acoustic echo canceller based on subjective assessment

Dynamically-reduced complexity implementation of echo cancelers

An acoustic echo canceller for teleconference systems

Stereophonic speech teleconferencing

A new adaptive summing technique for audio teleconferencing

Stability and insertion loss required for audio conference bridges

An integrated voice codec and echo canceller implemented in a single DSP processor

Decision-directed echo cancellation for full-duplex data transmission at 4800bps

A system design for real time signal processing

Spectral analysis using the QD algorithm

A new series of digital filter windows B-spline windows

Spectral analysis of order statistic filters

Spectral analysis for speech signals with bounded disturbance

Analysis of a moving sound source - compensation of the Doppler effect

Parallel AR computation with a reconfigurable signal processor

A noise-compensated long correlation matching method for AR spectral estimation of noisy signals

Subsets of reflection coefficients

Maximum-entropy spectrum from a non-extendable autocorrelation function

Accurate estimation of transfer function of all-pole model system from the signal buried in noise

New relationships between ARMA and AR processes

Maximum entropy pole-zero estimation

Prony's method based on eigenanalysis and overdetermined system approach

Improved frequency estimation using total least square approach

Adaptive algorithms for estimating the complete covariance eigenstructure

Some results on constrained maximum likelihood estimation

"Fast maximum likelihood joint estimation of frequency and frequency rate"

Estimation of partitioned set of parameters

Multidimensional spectral analysis and the decomposition of ionospheric mode structures

Complexity of waveform formulated from Fourier spectrum -- Theory and experiment

On estimation with bilinear time series

Adaptive spectrum estimation by including prior information into the window method

Describing greylevel textures through curvature primal sketching

Partitioning of texture image using two-dimensional linear prediction model

Automatic classification and recognition of defects in an eddy current non destructive testing

Feature extraction by uniform structure threshold logic networks

Non-stationary 2D estimation: An application to turbulence field analysis

An automatic visual inspection method of pattern using a circular feature extraction filter

Silhouette understanding system

Image recognition with inexact processing

Edge detection using zero crossings of directional derivatives of a random field model

Fourier analysis of object boundaries from two dimensional digitized images

The use of context in image restoration

Automatic classification of zooplancton by image analysis

An iterative method for restoring noisy images

Regularized iterative image restoration in a weighted Hilbert space

A parallel identification procedure for images with noncausal symmetric blurs

Karhunen-Loeve transform-based fast algorithms for image restoration

A fast local PPF restoration filter

Two-dimensional target recognition using normalization and recorrelation to eliminate noise

Restoration theory for digital images under additive noises by a constrained generalized inverse matrix

A projection operator for reconstruction images from sparse, ungridded data

Least square method for enhancement of laser radar images based on piecewise linear transformations of gray scales

A real-time digital mixer IC

A single-chip SC line equalizer system for full duplex multi-bit rate digital transmission

VLSI Architecture for an adaptive equalizer in ISDN line termination

An adaptive transversal filter VLSI

FIR Lowpass filter for signal decimation with 15 MHz clock frequency

Application specific integrated filters for HIFI digital audio signal processing

Experiences with automatic generation of audio band digital signal processing circuits

VLSI- A to D and D to A converters with multi-stage noise shaping modulators

A dynamic time warp VLSI processors for continuous speech recognition

System design for large-vocabulary speech recognition LSI

A wafer scale DTW multiprocessor

Syllable-based acoustic-phonetic decoding and wordhypotheses generation in fluently spoken speech

Generating word hypotheses in continuous speech

An efficient word lattice parsing algorithm for continuous speech recognition

A distributed system architecture for speech recognition

A speech recognition system of continuously spoken Japanese sentences and an application to a speech input device

Representation of a continuous speech understanding and dialog system in a homogeneous semantic net achitecture

A new architecture of a speech understanding system - A hybrid of a hierarchical and a network models

A network model dealing with focus of conversation for speech understanding system

The role of word-dependent coarticulatory effects in a phoneme-based speech recognition system

A continuous parameter and frequency domain based Markov model

Analysis, synthesis and perception of the French nasal vowels

Proposal and evaluation of models for the glottal source waveform

On the source-filter model of the vocal tract

Area estimation from ARMA analysis based on vocal-tract model

A model for nonstationary analysis of speech

Measurements on the effects of glottal opening and flow on the glottal impedance

Measurement and analysis of speech sound radiated from vocal tract wall

Effects of tempo and context on jaw openings for vowel in vowel sequence words

Simultaneous high-speed digital recording of vocal fold vibration and speech signal

A hierarchical classification of signals and corresponding approximation method based on minimum norm criterion

Effects of representative distortion and noise on wavelet recovery via the complex cepstrum

Performance and limitation of discrete band-limited signal extrapolation

Signal recovery from nonuniform samples and spectral analysis on random nonuniform samples

Spline fitting in presense of uniform noise

A unified approach to generalized sampling theorems

Integer programming approach to optimal smoothing of two-state Markov sequences

A time-domain signal resolution problem

Discrete signal reconstruction from its spectral magnitude and some samples

Ignorance-based signal estimation given multiple noisy realizations

An algorithm for bandlimited signal interpolation

High-quality speech at low bit rates: Multi-pulse and stochastically excited linear predictive coders

CELP Coding for high-quality speech at 8 kbit/s

High quality multi-pulse speech coder with pitch prediction

Vector adaptive predictive coding of speech at 9.6 kb/s

8 kbits/s Speech coder with pitch adaptive vector quantizer

Speech coder using phase equalization and vector quantization

Time-scale modification in medium to low rate speech coding

A study on the realtionships between stochastic and harmonic coding

Phase modelling and its application to sinusoidal transform coding

Single DSP 8kbps speech CODEC

A high-efficiency speech coding algorithm based on ADPCM with multi-quantizer

Rapid magnetic resonance imaging using the Hankel tranform approach

Consistency in tomographic reconstruction by iterative methods

Image reconstruction from truncated projections: A linear prediction approach

High speed image reconstruction using direct two-dimensional Fourier transform method

3-Dimensional reconstructing algorithm for digital tomo-synthesis

Detector width artifact correction in computer assisted tomography

Maximum entropy diffraction tomography

Constrained resolution enhancement in optical microscopic tomography

Blood flow phase detection on digital subtraction angiography

Computation of synthetical radiographs application to the analysis of high velocity flows

An automatic analyzing system of left ventricular cineangiograms

Signal restoration of distorted kinesiographic measurement

Ultrasonic beamwidth effect on evaluation of attenuation by sampling medical B-mode imaging data

Measurement methods of ultrasonic velocity by dip point analysis from echo spectrum and ultrasonic attenuation considering a dispersion

One approach to color image processing

Compensation of motion related blur in CCD color endoscope images

A semi-automatic system for schistosome eggs detection

Patterns in Hilbert transforms and Wigner-Ville distributions of electrocardiogram data

Novel method for real-time detection of QRS complex in the presence of strong noise

New signal processing methods applied in ECG microcomputer automatic diagnosis system recently designed

Estimation of intramuscular EMG signals from surface EMG signal analysis

A nonlinear digital filter for separating nonstationary waves from stationary waves of the EEG

Design and construction of a multiplexed electrode for monitoring evoked responses in cerebral cortex

A precise bearing estimation method of digital split-beam array system

A simple fast adaptive array based on a null steering beamformer

Noise influence on the accuracy of source localising, comparison of various receivers with an imposed achitecture

Multichannel lattice filter for an adaptive array processor with linear constraints

Eigenstructure approach to direction-of-arrival estimation in IR detector arrays

Processing technique for a linear FM pulse array

Bearing estimation with a linear array

An adaptive recursive algorithm for array processing of coherent signal

A novel compact beamformer for a high resolution sonar

A partitionable distributed beamformer and signal conditioner architecture

A digital signal processing approach to data handling for ultrasound beam-forming

The integrator as a time delay estimator

Frequency domain compensation for failed elements in linear sonar arrays

Eigenstructure based broadband source location estimation

Toeplitzization of correlation matrix in multipath environment

High frequency acoustic propagation in a range dependent guiding-to-antiguiding ocean channel transition

The application of linear constraints to an adaptive beamformer

Source localization with two dimensional array subject to uncertainty in sensor location

The effect of the character of the noise field upon the error in the optimum gain of an antenna array

Time delay estimation via generalized correlation with adaptive spatial prefiltering

Analysis of a moving sound source-Orbit estimation using linear sensor array

Characterization and identification of trajectories using a leg by leg approach of bearings only target motion analysis

On smoothed rank profile tests in eigenstructure approach to directions-of-arrival estimation

Performance analysis of the MUSIC algorithm

A fast and accurate algorithm for SAR Doppler center frequency estimation

Eigenstructure approach to Doppler estimation for wideband signals

Complex deconvolution in non-coherent radar system

Constraint optimization filter design for radar target enhancement

Iterative enhancement of noncoherent radar data

Group delay processing of radar signals

Resolution of coherent radar targets in the near field of an array antenna

A fast algorithm for evaluating the false alarm probability of a CFAR detector

Fundamental limits on multiple PRF radar performance

A microprocessor controlled CCD programmable transversal filter for radar applications

An improvement of phase correction for spotlight mapping radar

A model of peripheral auditory transduction using a phase vocoder with modified channel signals

A new approach to continuous speech recognition based on considerations on human processes of speech perception

How to make more efficient use of the fact that the speech signal is dynamic and redundant

Toward a massively parallel system for word recognition

Perceptually based processing in automatic speech recognition

Experiments with a computational model of the cochlea

Speech recognition using a cochlear model

A computational model for the peripheral auditory system: Application of speech recognition research

Representation of vowels in connected speech based on speech perception experiments

Speaker-independent isolated word recognition based on emphasized spectral dynamics

Speech analysis/Synthesis based on matching the synthesized and the original representations in the auditory nerve level

A nonlinear articulatory speech synthesizer using both time- and frequency-domain elements

Dynamically controlled excitation source for a time-varying format synthesizer

A modified-superposition speech synthesizer and its applications

Improved vocal tract models for speech synthesis

Diphone synthesis using an overlap-add technique for speech waveforms concatenation

The speech synthesis system for an unlimited Japanese vocabulary

Acoustic parameters and phonological rules of a text-to-speech system for Chinese

The effects of speaking rate on formant transitions in French synthesis-by-rule

The Japanese CV-Syllable positioning rule for speech synthesis

Microprosodic timing rules for Italian consonant clusters

Acoustic characteristics and the underlying rules of intonation of the common Japanese used by radio and television announcers

Pitch contour control in Japanese conversational speech

Multi-scale edge detection with Gaussian filters

A unified method for segmentation and edge detection using graph theory

Detection of object boundaries in synthetic aperture radar imagery using a human visual model

Contour recognition: A syntax-directed approach

Fast Hough transform for multidimensional signal processing

Applications of morphological filtering to image analysis and processing

Script recognition using hidden Markov models

Dynamic programming for region based pattern recognition

Constraint and consistency in stereo matching

A multi-resolution representation of 3D imagery for efficient image enhancement and processing

Iterative local parallel operations for structural analysis of line patterns using distance transformation

A new IIR type adaptive algorithm of parallel type structure

Derivation of the fast pole-zero (ARMA) recursive least squares algorithm using geometric projections

An adaptive IIR channel equaliser: A Kalman filter approach

Adaptive filters realized with second order sections

Bandpass adaptive pole-zero filtering

An adaptive stochastic filter with no positive real condition

A covariance RLS lattice for adaptive filtering

Rounding error analysis of the triangular lattice and escalator algorithms

An adaptive delay filter with varibale gains and variable taps

A flexible, numerically robust array processing algorithm and its relationship to the givens transformation

Numerical design of nonlinear adaptive filters

A high-speed programmable architecture using barrel shifters for FFT and digital filtering

Kalman filtering and systolic processors

A systolic vector quantization processor for real-time speech coding

A VLSI array for computing the DFT based on RNS

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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

Composite permutation coding of speech waveforms

Content Provider	IEEE Xplore Digital Library
Author	Luzheng Lu Cohen, G. Godlewski, P.
Copyright Year	1986
Description	Author affiliation: C.N.R.S., Paris, France (Luzheng Lu)
Abstract	A new vector coding (quantization) system and its application to speech waveforms are presented. This system consists of several permutation codes of Variant I. An algorithm for the system design is developed. An encoding procedure which minimizes the distortion and finds out the reproduction index to be transmitted using a algebraic method is proposed. The system was designed and tested for the encoding of speech waveforms. The results of experimental simulations show that this is a prospective vector coding scheme with simple encoding procedure, smaller memory required, and not large computational burden of the design algorithm.
Starting Page	2359
Ending Page	2362
File Size	120770
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1986.1168668
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1986-04-07
Publisher Place	Japan
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Speech coding Algorithm design and analysis Bit rate Yttrium Distortion measurement Samarium Partitioning algorithms
Content Type	Text
Resource Type	Article

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