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Volume 3 - 1978

Studies on pattern recognition approach to voiced-unvoiced-silence classification

Improvement of voicing decisions by use of context

Epoch extraction from linear prediction residual

Maximum likelihood pitch estimation using state-variable techniques

Interactive digital inverse filtering and its relation to linear prediction methods

A new method of Cepstrum analysis by using comb lifter

Arma modeling applied to linear prediction of speech

Study of an adaptive lattice structure for linear prediction analysis of speech

Real-zeros in pitch detection

Identification of nonstationary components by complex spectral zero analysis

Two-dimensional processing of spectrogram data

A high-quality all-digital sound spectrograph developed for speech signal analysis

Realization of digital transfer functions using cascaded normalized two-pairs

Calculation of all equivalent and canonic 2nd order digital filter structures

State-structures and minimal state-structures for arbitrary digital filters

Realization of a class of two-dimensional analog ladders with applications to wave digital filters

A figure of merit for digital filters

Formats in audio bandwidth processing applications

Digital filter realizations without overflow oscillations

An approach to eliminate roundoff errors in digital filters

Selection of data windows for digital signal processing

Adaptive lattice methods for linear prediction

An adaptive lattice structure for noise-cancelling applications

Non-optimal convergence of adaptive LMS with uncorrelated data

Selecting the length of an adaptive transversal filter

An adaptive equalizer with significantly reduced number of operations

Real-time signal processing for unbiased system identification

Adaptive data orthogonalization

Some mathematical results on the effects on digital adaptive filters of implementation errors and noise

Simple adaptive IIR filtering

A short-term sequential regression algorithm

On the maximum achievable conversion efficiency on a parametric acoustic array

Vibration sensitivity of the parametric acoustic receiving array

Errors in array response calculations due to incorrectly folding the vertical acoustic arrival structure into beam pattern

Control of the directional response of acoustic transducer array elements on curved surfaces by modification of the diffracted field

Array processing using the frequency-wavenumber approach

Effect of sampling errors on array gain

Synthetic array processing for underwater mapping applications

A flexible towed sonar for ocean acoustic measurements

An audio response method for CAI services

Speech synthesis using real-time software

A mixed-source model for speech compression and synthesis

Implementation of a channel vocoder synthesizer using a fast, time-multiplexed digital filter

An excitation function for LPC synthesis which retains the human Glottal phase characteristics

Votrax real time hardware for phoneme synthesis of speech

Speech generation through waveform synthesis

Comparisons of system identification methods in the presence of high noise levels and bandlimited inputs

Linear prediction and maximum entropy spectral analysis of finite bandwidth signals in noise

Comparative study of iterative deconvolution algorithms

Image restoration of space variant blurs by sectioned methods

Nonlinear filter for inversion of channel distortion

Observations on linear estimation

New stochastic realization algorithms for identification of ARMA models

Analysis and representation of composite signals by cepstral inverse filtering

Multilevel crossing rates for automated signal classification

Target identification from radar signatures

Multiband FIR digital filter design algorithm for radar clutter suppression

Typical performance characteristics of a two-dimensional CFAR

A technique for pole-zero modeling of complex-valued autocorrelations

A programmable sonar signal processor

Reconnaissance sonar for deep ocean seamount detection

An at-sea system for the prediction of underwater sound propagation

Inherent errors in sonar range prediction

Design considerations for feedback amplifiers

Audibility of transient intermodulation distortion

Omitted factors in audio circuit design

Computer analysis of transient distortion and low transient distortion amplifier design

Electroacoustic distortions: Multidimensional analysis of hearing aid transduced speech and music

A comparison between two approaches to automatic speaker recognition

Extraction of speaker-specific features from spoken code sentences

New techniques for text-independent speaker identification

Text-independent speaker identification from a large linguistically unconstrained time-spaced data base

Text-independent speaker identification based on piecewise canonical discriminant analysis

Automatic speaker identification for a large number of speakers

2400/16,000 Bps Multirate voice processor

High quality adaptive predictive coding of speech

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An optimal adaptation logic for delta modulation

Forward-adaptive delta modulator without explicit transmission of step size

32 Kbps CCITT Compatible split band coding scheme

Evaluation of LPC/CVSD tandem connections

A spectral enhancement procedure for the wideband/Narrowband tandem

Towards a variable frame rate speech transmission system with frame selection by time-domain segmentation - a status report

Two dimensional data compression of speech

Two-dimensional speech compression

High resolution autoregressive spectrum analysis using noise power cancellation

High resolution spectral analysis of sinusoids in correlated noise

Frequency estimation by linear prediction

Improvement of autoregressive spectral estimates in the presence of noise

The accuracy of center frequency estimators using linear predictive methods

Complex covariance/Maximum entropy doppler estimates for pulsed CO2lidar

An error formula for iterative prefiltering frequency estimates

Extrapolation and spectral estimation for bandlimited signals

Constant-Q signal analysis and synthesis

Spectrum analysis using frequency-domain adaptive windowing

Spectrum estimation of non-uniform sampled data

Input signals for sampling and identification of ecological systems

Input, signals and control in ecosystems

Extension of Levins loop analysis to transient and periodic disturbances

Estimating Oxygen demand in aquatic ecosystems

Gaps in the technology of speech understanding

Computing relative redundancy to measure grammatical constraint in speech recognition tasks

Dynamic programming, the viterbi algorithm, and low cost speech recognition

Automatic recognition of continuously spoken sentences from a finite state grammer

Recognition of continuously read natural corpus

A voice-input programming system using basic-like language

The Dawid speech recognition system

Fuzzy semantic network for a speech understanding system - an experimental study

A simple vocoder

Low bit rate cepstral vocoder using the log magnitude approximation filter

CCD CZT Spectral analysis applied to real time homomorphics speech analysis-synthesis

Implementation of a pole-zero analysis - synthesis system for speech

A variable frame length linear predictive coder

A method for reducing the transmission rate of a channel vocoder by using frame interpolation

Providing channel error protection for a 2400 bps linear predictive coded voice system

Channel coding considerations for digital speech encoded by linear prediction

Linear predictive coding of speech signals in a high ambient noise environment

Linear prediction techniques in the Walsh spectral domain for speech analysis and synthesis

Application of canonical coordinate methods to the characterization of a family of error minimizing signal compression techniques

An estimate of the order of an optimal FIR band-pass digital filter

Digital filters with prescribed zeros

Synthesis of digital filters with very low sensitivity of the frequency response to the change of the cofficients

Design of IIR filters using Pseudo-Boolean methods

Design and applications of uniform digital bandpass filter banks

A multiplexing scheme for multirate digital filtering with half-band filters

New class of recursive digital filters for decimation

An optimal filter design for variable sampling rates

Digital lowpass filtering using the discrete Hilbert transform

Nonstationary signal processing and model validation

Recursive derivation of reflection coefficients from noisy seismic data

On digital signal modelling and classification with the teleseismic data

Acoustic wave propagation by finite-difference techniques

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Digital recording of ultrasonic signals

A comparison of linear prediction, FFT, and zero-crossing analysis techniques for vowel recognition

Speaker-independent vowel indetification in continuous speech

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A comparative study of phonemic recognition by discrete orthogonal transforms

A phoneme recognition system based on human audition

Application of the subspace method to speech recognition

Application of novelty filter to segmentation of speech

Signal/Quantizing-distortion ratio measurements of fast-adaptive delta modulation systems

Predictive coding of speech signals and subjective error criteria

Speech synthesis by linear interpolation of spectral parameters between dyad boundaries

Perceptual and objective evaluation of speech processed by adaptive differential PCM

A study of complexity and quality of speech waveform coders

Objective speech quality evaluation of narrowband LPC vocoders

Statistical correlation between objective and subjective measures for speech quality

Estimation of LPC coefficients from speech waveforms degraded by additive random noise

An investigation of several frequency-domain processing methods for enhancing the intelligibility of speech in wideband random noise

Suppression of noise in speech using the saber method

LMS Adaptive filtering for enhancing the quality of noisy speech

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Signal processing with number theoretic transforms and limited word lengths

Number theoretic transforms with modulus $2^{2q}-$ $2^{q}+$ 1

Simulation of large length filters using fermat number transforms

An inplace self recordering FFT

Real-factor FFT algorithms

New algorithms for convolution and DFT based on polynomial transforms

M-Adic invariant filters

Reconstruction of signals from their linear mapping image

Linear detection filtering for the context of a least-squares estimator for signal processing applications

Dealiasing of the spectra of sampled noise

Underwater sound arrival angle estimation by multiple cross-correlation measurements

The least squares estimation of time delay and its use in signal detection

Confidence bounds for magnitude-squared coherence estimates

An all-digital phase-measurement techniques using clipped-quadrature correlation

Detection performance of the smooth coherence transform (SCOT)

Object classification using sonar data

A study of stochastic processes associated with sonar detection

Acoustic transfer function of the ocean for a motional source

Display and interpretation of a time-spread underwater acoustic channel's bandlimited impulse response

Some experiments with a syntax directed speech recognition system

Performance analysis of syntactic-recognizer of isolated words

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Directory assistance by means of automatic recognition of spoken spelled names

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Two-dimensional zoom FFT

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Design of stable 2-D half-plane recursive filters using spectral factorization

Design of inherently stable two-dimensional recursive filters imitating the behaviour of one-dimensional analog filters

An algorithm for testing stability of two-dimensional digital recursive filters

Computer aided generation of two dimensional transfer functions from one dimensional transfer functions

Design of semicausal two-dimensional recursive filters

A well suited two-dimensional linear recursive filter for image processing

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Digital phase locked loop

Parallel counter design using four-valued threshold logic

An error anaylsis of a FFT implementation using the residue number system

Techniques for residue-to-analog conversion for high data rate digital filtering

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Impulse response testing of acoustic spaces

Super directive spectrum analyzer by use of moving microphones

Free-field measurements for a loudspeaker system in a normal room--Using digital signal processing techniques

Correction of near-field acoustic measurements made with arbitrary measuring transducers

Reflection of spherical sound wave from a rigid sphere

Statistical properties of the poisson reverberation envelope

On an extension of A. N. Krylov's numerical method for determining the frequencies of small vibrations of systems with damping

ICASSP '77. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '76. IEEE International Conference on Acoustics, Speech, and Signal Processing

Extraction of speaker-specific features from spoken code sentences

Content Provider	IEEE Xplore Digital Library
Author	Jesorsky, P. Hofker, U. Talmi, M.
Copyright Year	1978
Description	Author affiliation: Heinrich-Hertz-Institut für Nachrichtentechnik GmbH, Einsteinufer, Berlin, Germany (Jesorsky, P.)
Abstract	In the speaker-verification task speakers are assumed to be cooperative and are therefore willing to pronounce a pre-arranged code sentence. This paper deals with the extraction of speaker-specific features from a time-normalized parametric description of the code sentence. A new mml (minimum-maximum-locating) method for time-normalization is presented together with an overview of techniques used for feature-extraction. Results of recognition experiments are discussed, which have been obtained with the AUROS system (Automatic Recognition of Speakers by Computers).
Starting Page	279
Ending Page	282
File Size	78991
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1978.1170396
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1978-04-10
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Feature extraction Speech analysis Speech coding Pattern recognition Speech processing Production systems Hardware Computer science Stress Bandwidth
Content Type	Text
Resource Type	Article

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