NDLI: Analysis of techniques for processing parallel signal outputs

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2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2013 IEEE International Conference on Acoustics, Speech and Signal Processing

2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2011 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2010 IEEE International Conference on Acoustics, Speech and Signal Processing

2009 IEEE International Conference on Acoustics, Speech and Signal Processing

2008 IEEE International Conference on Acoustics, Speech and Signal Processing

2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07

2006 IEEE International Conference on Acoustics Speech and Signal Processing Proceedings

Proceedings. (ICASSP '05). IEEE International Conference on Acoustics, Speech, and Signal Processing, 2005.

2004 IEEE International Conference on Acoustics, Speech, and Signal Processing

2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03).

2002 IEEE International Conference on Acoustics, Speech, and Signal Processing

2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)

2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)

1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258)

Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181)

1997 IEEE International Conference on Acoustics, Speech, and Signal Processing

1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings

1995 International Conference on Acoustics, Speech, and Signal Processing

Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing

[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing

[Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing

International Conference on Acoustics, Speech, and Signal Processing

International Conference on Acoustics, Speech, and Signal Processing,

ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '85. IEEE International Conference on Acoustics, Speech, and Signal Processing

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

ICASSP '83. IEEE International Conference on Acoustics, Speech, and Signal Processing

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ICASSP '81. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '80. IEEE International Conference on Acoustics, Speech, and Signal Processing

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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

Volume 1 - 1976

Acoustic-phonetic experiment facility for the study of continuous speech

A dynamic processing approach to extraction and categorization of phonemic information

The 1976 modular acoustic processor (MAP) : Signal analysis and phonemic segmentation

The 1976 modular acoustic processor (MAP) : Diadic segment classification and final phonemic string estimation

Acoustic-phonetic recognition in BBN SPEECHLIS

A Markov model acoustic phonetic component for automatic speech recognition

Speech segmentation and feature normalization based on area functions

The syntax of acoustic segments

An algorithm using linguistic information and its application to the analysis of speech in the spectral domain

A channel adapted vocoder and its applications to continuous speech recognition

Formant excitation before and after glottal closure

A study on laryngeal control for pitch change by use of anatomical structure model

A voice source taking account of coupling with the supraglottal cavities

A hardware vocal source simulator

A technique for the automatic location and description of pitch contours

Computer modeling and estimation of linguistic stress patterns

Automation of the measurement of laryngeal vibration patterns from high speed film

An experimental investigation of the optimal filter as an area function perdictor

A technique for converting the linear prediction areas model of speech to a simple articulatory model

Vocal tract area function measurements: Two time-domain methods

Detection of the closed glottis interval

Analysis and areas modelling of Nasalised speech by a multivariable identification technique

Research of the pitch contour rules for an Italian language speech understanding system

Automatic generation of pitch contour for speech synthesis

An adaptive inverse digital filter for formant analysis of speech

Methods for nonlinear spectral distortion of speech signals

600 bps voice digitizer

Speech digitalization with channel vocoders

Output spectrum contour scaling for an all digital channel vocoder

A framework for the objective evaluation of vocoder speech quality

Quality comparison measure for linear predictive systems

Discrete Hilbert transform filtering

New results in fixed-point fast Fourier transform error analysis

A new algorithm for doing the finite discrete Fourier transformation in the frequency domain imposing uniform and Gaussian boundary conditions

The odd discrete Fourier transform

Discrete convolution of complex integer sequences

Orthogonal transforms for digital signal processing

Discrete Fourier transform based on a double-sampling and its applications

Design technique for a class of stable two-dimensional recursive digital filters

Harmonic differential calculus and filtering in Galois fields

A novel pattern learning and classification procedure applied to the learning of vowels

Analysis, recognition and perception of voiceless fricative consonants in Japanese

Acoustic discrimination between [f] and [θ] in a single speaker

Automatic acoustic-phonetic analysis of vowels and sonorants

A comparative study of the use of zero-crossing analysis methods for vowel recognition

Recognition of vowels in connected speech by use of the characteristics on perception of vowel

Bengali speech : Formant structures of single vowels and initial vowels of words

Automatic acoustic-phonetic analysis of fricatives and plosives

Fusion and identification of synthetic vowels in dichotic listening

Source data entry using voice input

Study of an on-line, adaptive speaker-independent word recognition system based on acoustic-phonetic analysis and statistical pattern recognition techniques

Recognition of coded speech (phonocodes)

Speaker independent recognition of connected digits

An improved isolation word recognition system based upon the linear prediction residual

VICI - A speaker independent word recognition system

Spoken word recognition system for unlimited adult male speakers

Computers applied for the recognition of Hindi syllables

Application of discrete word recognition and response to multiuser tactical communications: WRS

An attempt of automatic recognition of some Russian words

Spoken word recognition using the restricted number of learning samples

Digital coding of speech in sub-bands

LPC Synthesis starting from white noise corrupted or differentially quantized speech

A split-band predictive coding system at 16 kb/s

High quality 16 kb/s voice transmission

Single-integration, adaptive delta modulation

Enhancement of speech by adaptive filtering

Digital encoding of variable-length vectors with application to pitch extraction and pitch-synchronous speech analysis and synthesis

A pitch compensating quantizer

Performance analysis of DPCM speech-transmission systems using Kalman predictors

Subjective evaluation of PCM coded speech

Advanced voice modems for aero communications

A method for filling gaps in a speech signal left by the excision of impulsive noise

Digital lattice filters with reduced number of multipliers

Filtering for code conversion in digital telephone exchanges

Design of nonrecursive digital filters to meet maximum and minimum frequency response constraints

An algorithm for designing constrained least-squares filters

Recent developments in the design and implementation of digital decimators, interpolators, and narrow band filters

The autocorrelation function and spectra of a signal that has been randomly sampled

Covariance-invariant signal processing

Graphical determination of the group delay characteristics of digital filters

Analysis-synthesis using pole-zero approximations to speech spectra

Automatic formant extraction utilizing mel scale and equal loudness contour

Measuring pitch and formant frequencies for a speech understanding system

Direct linear prediction for fundamental frequency analysis

An algorithm for digital time-domain pitch period determination of speech signals and its application to detect F0dynamics in VCV utterances

A pattern classification algorithm for the voiced/Unvoiced decision

Dynamic energy tracking for responsive voicing

Some comparisons among several pitch detection algorithms

Effect of noise and distortion in speech on parametric extraction

Computer analysis of time jitter in vowel sounds

Real-time information reduction in digital sound spectograms of speech

Perceptual and acoustic cues of female voice

An adaptive speech analysis system

Non-prosodic pitch variations in continouos speech

Deconvolution by poisson transformation

Sectioned digital filtering for nonlinear Bayesian signal deconvolution

Time delay estimation

Signal processing by function elimination filters

Sectioned spectrum processing for wideband signals

Crossfrequency estimation for deconvolution

A Monte Carlo approach to numerical deconvolution

Quadratic residues: Application to chirp filters and discrete Fourier transforms

Comparison of transit detectors in ocean-like environments

Localized variation in the ocean's transmission properties: Its drastic effect on a sonar display

A novel method for measuring phase and sensitivity of long focused acoustic array

A fast Fourier transform (FFT) based sonar signal processor

An adaptive thresholding system for functioning in nonstationary noise backgrounds

Modeling hydrophone array directivity effects for sonar system performance prediction

Applications of network synthesis to loudspeaker system theory

Synthesis of loudspeaker driver parameters

Non-linear effects in direct radiator loudspeaker systems

Application of recent Australian loudspeaker research to producible loudspeaker systems

Theory and applications of electrically tapered electro-acoustic arrays

Focus of attention in a distributed-logic speech understanding system

Syntax and semantics in a distributed speech understanding system

Preliminary results on the performance of a system for the automatic recognition of continuous speech

Organization and operation of a connected speech understanding system at lexical, syntactic and semantic levels

A speech recognition system for connected word sequences

Uses of higher level knowledge in a speech understanding system: A progress report

Speech recognition in the question-answering system operated by conversational speech

Generative grammars and dynamic programming in speech recognition with learning

Phoneme-by-phoneme recognition of speech composed of the words of given vocabulary

A syntactic analyzer adapted to speech recognition

A multi-purpose speech recognition system

"A new method for accurate analysis of voiced speech"

New lattice methods for linear prediction

LPCW: An LPC vocoder with linear predictive spectral warping

Piecewise linear predictive coding (PLPC)

Linear predictive coding systems

A direct method for sequentially updating linear predictor coefficients for the covariance method

A comparison of three methods for coefficient quantization and bit allocation

Towards perceptually consistent measures of spectral distance

The factorial linear modelling: A Karhumen-Loeve approach to speech analysis

Linear estimation filters in spectral analysis

Evaluation of the performance of piecewise-linear-prediction coding (PLPC) of speech signals

Roundoff noise minimization in state-space digital filtering

Random rounding: Some principles and applications

Filter structures which minimize roundoff noise in fixed point digital filters

An equivalent network theory for a class of discrete-time networks

Optimization of recursive cascade filters

An algorithm for optimally ordering the sections of a cascade digital filter

A general realization method for wave digital filters

An efficient parallel algorithm for digital IIR filters

Digital fixed-point multiplication error structure and some consequences

Error estimation of digital filters with arbitrary structure and arithmetic by simulation

Distortion in microphones

Distortion in tape recording systems

Distortion in disc recording systems

Distortion in audio amplifiers

Loudspeaker distortion

On the audibility of distortion

Speech timing of coreference

Word hypothesization in the hearsay II speech system

Word verification in a speech understanding system

Word spotting in continuous speech using linear predictive coding

Dictionary expansion via phonological rules for a speech understanding system

Lexical classification in a speech understanding system using fuzzy relations

Duration as a syntactic boundary cue in ambiguous sentences

A digital filter bank for spectral matching

Phonological rule testing of conversational speech

Probabilistic lexical retrieval component with embedded phonological word boundary rules

Intermodulation distortion in hearing aids: The need for measurement standards and inherent complications

Protocol for prescriptive fitting of a wearable master hearing aid

Realtime recognition of unvoiced fricatives in continuous speech to aid the deaf

A real time spectrograph with implications for speech training for the deaf

Speech perception via the tactile mode: Progress report

Tactile stimulation as an aid for the deaf in production and reception of speech: Preliminary studies

Analysis of consonant recognition scores of congenital sensorineural hearing impaired

Two sensory aids having profound effects on the blind

An instrumentation for facilitating easy onset patterns for stutterers

Acoustical analysis for voice disorders

Some properties of formant frequencies of vowels uttered by deaf and hard of hearing children

Hardware considerations in FFT processors

Automatic generation of time efficient digital signal processing software

Programmable communications terminal optimization using systematic digital signal compression procedures

Simple hybrid systems for accurate synthesis and analysis of harmonic spectra

A digital loop communication system applied to signal processing and speech research

A digital signal processing system

The programmable array processor

A new portable stand alone digital processor

A two's complement pipeline multiplier

GASP: A programmable signal processor

Sonar design in the real ocean: Multipath limitations on sonar performance

Statistical properties of underwater acoustic ambient noise fields

Motion induced coherence degradation in passive systems

Dispersive properties of the underwater acoustic channel

Normal mode theory of underwater sound propagation in a range dependent environment

Sonar design in the real ocean: Target, background, and own ship limitations on sonar performance

The passive sonar equation - effects of additive interference

Wideband target strength measurements

An automated solution for the wide band sonar equation

A text-to-speech system based entirely on rules

Speech synthesis from english text: A progress report

Automatic generation of control signals for a parallel formant speech synthesizer

Speech synthesis by dyads and automatic intonation processing

Computer synthesis of Mandarin

A comprehensive model for fundamental frequency generation

Improving synthetic speech quality using binaural reverberation

Multiplex-vocoder for voice response

Speech processing by splicing of autocorrelation function

A multi-channel digital audio laboratory facility

Speech resynthesis with a hardware synthesizer

An application of the linear prediction technique to efficient coding of speech segments

Cascade realization of digital inverse filter for extracting speaker dependent features

Text independent speaker recognition using orthogonal linear prediction

On the connection of some time characteristics of speech signal with the individuality of voice

Feature evaluation and selection for an on-line, adaptive speaker verification system

Automatic speaker recognition by computers

Speaker identification by multivariable linear prediction analysis

Speech - a possible indicator of physical stress

Analysis of techniques for processing parallel signal outputs

Design procedure for improving the usable bandwidth of an MTI radar signal processor

Signal processing for feature extraction and pattern recognition

Frequency-variant optical systems for signal analysis and processing

Performance of a range-ambiguous MTI and doppler filter system

Digital signal processing techniques in truck tire vibration and sound analysis

The application of cepstrum technique in power cable fault detection

Plane averaging signal processing on radar using DFT technology

"Is four channel a quadrifizzle?"

Noise measurement to ensure compatible land use

Theoretical aspect of electromechanical transducers

A meteor infrasound recording system

The visual component of speech reception in conference centers

Vibration pick-up type ear microphone

Analysis of techniques for processing parallel signal outputs

Content Provider	IEEE Xplore Digital Library
Author	Cantwell, T. Wilmot, R.
Copyright Year	1976
Description	Author affiliation: Hughes Aircraft Company, Fullerton, California (Cantwell, T.)
Abstract	Three techniques for processing parallel sonar outputs are individual processors for each output, summing all parallel outputs for use in one processor, and selecting the maximum (peak) value of the parallel outputs for use in one processor. An analysis is made of the detection sensitivity of each technique as a function of the number of parallel outputs and the amount of signal integration. The results show that the peak selection technique provides a substantial savings in processing equipment and reduces sonar display requirements compared to the individual processor technique and provides enhanced detection sensitivity compared to the summation technique.
Starting Page	741
Ending Page	744
File Size	90980
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1976.1169996
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1976-04-12
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Signal analysis Signal processing Gaussian noise Envelope detectors Signal to noise ratio Sonar equipment Gaussian distribution Sonar detection Computer displays Computer simulation
Content Type	Text
Resource Type	Article

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