NDLI: Language identification using phone-based acoustic likelihoods

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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

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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

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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

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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

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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

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Complexity reduction for FS-1016 with multistage search

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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

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Using MAP estimated parameters to improve HMM speech recognition performance

Non-linear input transformations for discriminative HMMs

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

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

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

Variable bit rate ADPCM via arithmetic coding

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High-quality harmonic coding at very low bit rates

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A distributed decision approach to speaker verification

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Robust cepstral features for speaker identification

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

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Toward vocabulary independent telephone speech recognition

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

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

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Language identification using phone-based acoustic likelihoods

Content Provider	IEEE Xplore Digital Library
Author	Lamel, L.F. Gauvain, J.L.
Copyright Year	1994
Description	Author affiliation: LIMSI-CNRS, Orsay, France (Lamel, L.F.; Gauvain, J.L.)
Abstract	Applies the technique of phone-based acoustic likelihoods to the problem of language identification. The basic idea is to process the unknown speech signal by language-specific phone model sets in parallel, and to hypothesize the language associated with the model set having the highest likelihood. Using laboratory quality speech the language can be identified as French or English with better than 99% accuracy with only as little as 2 seconds of speech. On spontaneous telephone speech from the OGI corpus, the language can be identified as French or English with 82% accuracy with 10 seconds of speech. The 10 language identification rate using the OGI corpus is 59.7% with 10 seconds of signal.<>
File Size	386099
File Format	PDF
ISBN	0780317750
ISSN	15206149
DOI	10.1109/ICASSP.1994.389297
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	Natural languages Hidden Markov models Signal processing Speech processing Laboratories Neural networks Speech recognition Character recognition Telephony Airports
Content Type	Text
Resource Type	Article

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