NDLI: Improved voice identification using a nearest-neighbor distance measure

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

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

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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

Volume v - 1994

Volume i - 1994

Genones: optimizing the degree of mixture tying in a large vocabulary hidden Markov model based speech recognizer

On the fuzzy vector quantization based hidden Markov model

Which model for future speech recognition systems: hidden Markov models or finite-state automata?

Model topology selection for isolated word recognition

Discriminative training of high performance speech recognizer using N best candidates

The influence of speech coding algorithms on automatic speech recognition

Task independent and dependent training: performance comparison of HMM and hybrid HMM/MLP approaches

Using Gaussian mixture modeling in speech recognition

Pseudo-segment based speech recognition using neural recurrent whole-word recognizers

HMM with global path constraint in Viterbi decoding for isolated word recognition

Speech modelling using cepstral-time feature matrices and hidden Markov models

A MRF-based parallel processing algorithm for speech recognition using linear predictive HMM

Automatic generation of prosodic rules for speech synthesis

Data-driven joint f/sub 0/ and duration modeling in text to speech conversion for Spanish

Articulatory speech synthesis based on fractional delay waveguide filters

Hybrid time- and frequency-domain speech synthesis with extended glottal source generation

A new waveform speech synthesis approach based on the COC speech spectrum

New algorithm for spectral smoothing and envelope modification for LP-PSOLA synthesis

Development of a text-to-speech system for Japanese based on waveform splicing

High quality text-to-speech synthesis: a comparison of four candidate algorithms

Comparative experiments on large vocabulary speech recognition

The LIMSI continuous speech dictation system: evaluation on the ARPA Wall Street Journal task

New graph search techniques for speech recognition

Improving speech recognition performance via phone-dependent VQ codebooks and adaptive language models in SPHINX-II

Towards large vocabulary Mandarin Chinese speech recognition

A comparison of phoneme decision tree (PDT) and context adaptive phone (CAP) based approaches to vocabulary-independent speech recognition

Robust methods for using context-dependent features and models in a continuous speech recognizer

High-order allpole modelling of the spectral envelope

Single stage spectral quantization at 20 bits

Low-complexity encoding of speech LSF parameters using constrained-storage TSVQ

Segmental quantization of speech spectral information

Vector quantization-lattice vector quantization of speech LPC coefficients

Spectral quantization of cepstral coefficients

Variable dimension vector quantization of linear predictive coefficients of speech

Efficient interblock noiseless coding of speech LPC parameters

Efficient vector quantisation of LPC parameters for noisy channels

A novel split residual vector quantization scheme for low bit rate speech coding

Vector quantization with hyper-columnar clusters

Filter bank design based on discriminative feature extraction

Analysis of spectral interpolation with weighting dependent on frame energy

Determination of human vocal-tract dynamic geometry from formant trajectories using spatial and temporal Fourier analysis

Voiced-speech analysis based on the residual interfering signal canceler (RISC) algorithm

Voice conversion based on piecewise linear conversion rules of formant frequency and spectrum tilt

A novel approach for classifying continuous speech into visible mouth-shape related classes

Speech spectrum transformation by speaker interpolation

A robust sequential parameter estimation for time-varying speech signal analysis

Time-scale modification of speech based on a nonlinear oscillator model

Adaptive noise immunity learning for word spotting

Signal bias removal for robust telephone based speech recognition in adverse environments

Root adaptive homomorphic deconvolution schemes for speech recognition in noise

Noise independent speech recognition for a variety of noise types

Adaptation to new microphones using tied-mixture normalization

On the importance of the microphone position for speech recognition in the car

Degraded word recognition based on segmental signal-to-noise ratio weighting

Integrating RASTA-PLP into speech recognition

Probabilistic optimum filtering for robust speech recognition

Duration and spectral based stress token generation for HMM speech recognition under stress

Continuous speech recognition in noise using spectral subtraction and HMM adaptation

Parallel distributed binary mapping models for speech recognition

New ways to use LVQ-codebooks together with hidden Markov models

The performance prediction method on sentence recognition system using a finite state automaton

Rejection for connected digit recognition based on GPD segmental discrimination

Wordspotter training using figure-of-merit back propagation

Approaches to topic identification on the switchboard corpus

Spotting events in continuous speech

A fast lattice-based approach to vocabulary independent wordspotting

Optimizing recognition and rejection performance in wordspotting systems

Sentence spotting applied to partial sentences and unknown words

Learning complex output representations in connectionist parsing of spoken language

A robust language model incorporating a substring parser and extended n-grams

Ergodic hidden Markov models and polygrams for language modeling

Reducing the computational complexity for inferring stochastic context-free grammar rules from example text

Correcting complex false starts in spontaneous speech

JANUS 93: towards spontaneous speech translation

Unanswerable queries in a spontaneous speech task

Demonstrations and applications of spoken language technology: highlights and perspectives from the 1993 ARPA Spoken Language Technology and Applications Day

Perceptual benchmarks for automatic language identification

Speaker recognition in tactical communications

Speaker recognition based on minimum error discriminative training

Improved voice identification using a nearest-neighbor distance measure

A model distance measure for talker clustering and identification

Discriminating semi-continuous HMM for speaker verification

Distance measures for text-independent speaker recognition based on MAR model

Automatic language identification using sub-word models

Language identification using phone-based acoustic likelihoods

Analysis of phoneme-based features for language identification

Reed-Solomon coding for CELP EDAC in land mobile radio

A VR-CELP codec implementation for CDMA mobile communications

A 5.6 kb/s speech codec using a pulse codebook and improved Viterbi decoding

Comparison of ARMA modelling methods for low bit rate speech coding

M-LCELP speech coding at 4 kbps

Speech and channel codec candidate for the half rate digital cellular channel

Complexity reduction for FS-1016 with multistage search

A codec candidate for the GSM half rate speech channel

Phoneme recognition improvement by restricting training section in concatenated HMM training

All-phoneme ergodic hidden Markov network for unsupervised speaker adaptation

Tree-structured speaker clustering for fast speaker adaptation

Automatic training of phoneme dictionary based on mutual information criterion

An evaluation of cross-language adaptation for rapid HMM development in a new language

Discriminative training for improved neural prediction systems

Using MAP estimated parameters to improve HMM speech recognition performance

Non-linear input transformations for discriminative HMMs

Bayesian learning of the SCHMM parameters for speech recognition

Learning state-dependent stream weights for multi-codebook HMM speech recognition systems

SLHMM: a continuous speech recognition system based on Alphanet-HMM

Speech compression using ARMA model and wavelet transform

Mixed-phase AR models for voiced speech and perceptual cost functions

Spectral entropy: an alternative indicator for rate allocation?

Speech coding based on adaptive mel-cepstral analysis

High quality coding of wideband audio signals using transform coded excitation (TCX)

Variable bit rate ADPCM via arithmetic coding

Non-linear short-term prediction in speech coding

High-quality harmonic coding at very low bit rates

Transform trellis coded quantization of speech using small frame sizes

Low-bit-rate speech coding using a two-dimensional transform of residual signals and waveform interpolation

Detecting an imposter in telephone speech

Speaker identification using neural tree networks

Segmentation of speech using speaker identification

Hierarchical pattern classification for high performance text-independent speaker verification systems

A hybrid HMM-MLP speaker verification algorithm for telephone speech

Investigations on speaker characterization from Orphee system techniques

A robust, segmental method for text independent speaker identification

A distributed decision approach to speaker verification

Text-dependent speaker recognition using the information in the higher frequency band

A quantitative assessment of the relative speaker discriminating properties of phonemes

Robust cepstral features for speaker identification

Speaker adaptation of tied-mixture-based phoneme models for text-prompted speaker recognition

On the use of data-driven clustering technique for identification of poly- and mono-phonemes for four European languages

Toward vocabulary independent telephone speech recognition

The development of file formats for very large speech corpora: SPHERE and SHORTEN

Sources of degradation of speech recognition in the telephone network

Application of vector quantized hidden Markov modeling to telephone network based connected digit recognition

A comparative analysis of Japanese and English digit recognition

The Australian National Database of Spoken Language

Towards automatic collection of the US census

The voice across Japan database-the Japanese language contribution to Polyphone

Constructing telephone acoustic models from a high-quality speech corpus

Macrophone: an American English telephone speech corpus for the Polyphone project

Knowledge based approach to consonant recognition

Phonemic segmentation of fluent speech

Word accent patterns modelling by concatenation of mora hidden Markov models

An efficient combination of acoustic and supra-segmental informations in a speech recognition system

Using multiple vector quantization and semicontinuous hidden Markov models for speech recognition

Stochastic trajectory modeling for speech recognition

A new method for segmenting continuous speech

Segmental phoneme recognition using piecewise linear regression

Phonetic classification and recognition using HMM representation of overlapping articulatory features for all classes of English sounds

Phoneme recognition in continuous speech using large inhomogeneous hidden Markov models

IPA: improved phone modelling with recurrent neural networks

Acoustic modeling for speech recognition based on spotting of phonetic units

A new speech enhancement technique with application to speaker identification

Co-channel speaker separation based on maximum-likelihood deconvolution

Adaptation techniques for ambience and microphone compensation in the IBM Tangora speech recognition system

Noise suppression using a wavelet model

Comparative performance of spectral subtraction and HMM-based speech enhancement strategies with application to hearing and design

Cepstrum based deconvolution for speech dereverberation

Low residual noise speech enhancement utilizing time-frequency filtering

Speech enhancement based on a new set of auditory constrained parameters

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

Automatic language identification using syllabic spectral features

Automatic language identification of telephone speech messages using phoneme recognition and N-gram modeling

Volume iii - 1994

Volume vi - 1994

Volume iv - 1994

Volume ii - 1994

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

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

Improved voice identification using a nearest-neighbor distance measure

Content Provider	IEEE Xplore Digital Library
Author	Bahler, L.G. Porter, J.E. Higgins, A.L.
Copyright Year	1994
Description	Author affiliation: ITT Aerosp./Commun. Div, San Diego, CA, USA (Bahler, L.G.; Porter, J.E.; Higgins, A.L.)
Abstract	This paper describes recent improvements to an algorithm for identifying an unknown voice from a set of known voices using unconstrained speech material. These algorithms compare the underlying probability distributions of speech utterances using a method that is free of assumptions regarding the form of the distributions (e.g., Gaussian, etc.). In comparing two utterances, the algorithms accumulate minimum inter-frame distances between frames of the utterances. In recognition tests on the Switchboard database, using a closed population of speakers, we show that the new algorithm performs substantially better than the baseline algorithm. The modifications are segment-based scoring, limiting likelihood ratio estimates for robustness and estimating biases associated with reference files.<>
File Size	185684
File Format	PDF
ISBN	0780317750
ISSN	15206149
DOI	10.1109/ICASSP.1994.389291
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1994-04-19
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Testing Speech Aerospace materials Performance evaluation Mood Stress Distortion measurement Speaker recognition Equations Iterative algorithms
Content Type	Text
Resource Type	Article

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