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

Speech enhancement based on a new set of auditory constrained parameters (Abstract)

Low residual noise speech enhancement utilizing time-frequency filtering (Abstract)

Cepstrum based deconvolution for speech dereverberation (Abstract)

A comparative analysis of Japanese and English digit recognition (Abstract)

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

Sources of degradation of speech recognition in the telephone network (Abstract)

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

Toward vocabulary independent telephone speech recognition (Abstract)

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

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

Robust cepstral features for speaker identification (Abstract)

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

A quantitative assessment of the relative speaker discriminating properties of phonemes (Abstract)

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

A distributed decision approach to speaker verification (Abstract)

A robust, segmental method for text independent speaker identification (Abstract)

Investigations on speaker characterization from Orphee system techniques (Abstract)

A hybrid HMM-MLP speaker verification algorithm for telephone speech (Abstract)

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

Segmentation of speech using speaker identification (Abstract)

Speaker identification using neural tree networks (Abstract)

Detecting an imposter in telephone speech (Abstract)

Noise suppression using a wavelet model (Abstract)

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

Transform trellis coded quantization of speech using small frame sizes (Abstract)

High-quality harmonic coding at very low bit rates (Abstract)

Non-linear short-term prediction in speech coding (Abstract)

Variable bit rate ADPCM via arithmetic coding (Abstract)

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

Speech coding based on adaptive mel-cepstral analysis (Abstract)

Spectral entropy: an alternative indicator for rate allocation? (Abstract)

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

Speech compression using ARMA model and wavelet transform (Abstract)

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

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

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

Bayesian learning of the SCHMM parameters for speech recognition (Abstract)

Non-linear input transformations for discriminative HMMs (Abstract)

Using MAP estimated parameters to improve HMM speech recognition performance (Abstract)

Discriminative training for improved neural prediction systems (Abstract)

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

Automatic training of phoneme dictionary based on mutual information criterion (Abstract)

Tree-structured speaker clustering for fast speaker adaptation (Abstract)

All-phoneme ergodic hidden Markov network for unsupervised speaker adaptation (Abstract)

Co-channel speaker separation based on maximum-likelihood deconvolution (Abstract)

Phoneme recognition improvement by restricting training section in concatenated HMM training (Abstract)

A codec candidate for the GSM half rate speech channel (Abstract)

Complexity reduction for FS-1016 with multistage search (Abstract)

Speech and channel codec candidate for the half rate digital cellular channel (Abstract)

M-LCELP speech coding at 4 kbps (Abstract)

Comparison of ARMA modelling methods for low bit rate speech coding (Abstract)

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

A VR-CELP codec implementation for CDMA mobile communications (Abstract)

Reed-Solomon coding for CELP EDAC in land mobile radio (Abstract)

Analysis of phoneme-based features for language identification (Abstract)

A new speech enhancement technique with application to speaker identification (Abstract)

Language identification using phone-based acoustic likelihoods (Abstract)

Automatic language identification using syllabic spectral features (Abstract)

Automatic language identification using sub-word models (Abstract)

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

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

Discriminating semi-continuous HMM for speaker verification (Abstract)

A model distance measure for talker clustering and identification (Abstract)

Improved voice identification using a nearest-neighbor distance measure (Abstract)

Speaker recognition based on minimum error discriminative training (Abstract)

Speaker recognition in tactical communications (Abstract)

Acoustic modeling for speech recognition based on spotting of phonetic units (Abstract)

Perceptual benchmarks for automatic language identification (Abstract)

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

Unanswerable queries in a spontaneous speech task (Abstract)

JANUS 93: towards spontaneous speech translation (Abstract)

Correcting complex false starts in spontaneous speech (Abstract)

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

Ergodic hidden Markov models and polygrams for language modeling (Abstract)

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

Learning complex output representations in connectionist parsing of spoken language (Abstract)

Sentence spotting applied to partial sentences and unknown words (Abstract)

IPA: improved phone modelling with recurrent neural networks (Abstract)

Optimizing recognition and rejection performance in wordspotting systems (Abstract)

A fast lattice-based approach to vocabulary independent wordspotting (Abstract)

Spotting events in continuous speech (Abstract)

Approaches to topic identification on the switchboard corpus (Abstract)

Wordspotter training using figure-of-merit back propagation (Abstract)

Rejection for connected digit recognition based on GPD segmental discrimination (Abstract)

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

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

Parallel distributed binary mapping models for speech recognition (Abstract)

Continuous speech recognition in noise using spectral subtraction and HMM adaptation (Abstract)

Phoneme recognition in continuous speech using large inhomogeneous hidden Markov models (Abstract)

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

Probabilistic optimum filtering for robust speech recognition (Abstract)

Integrating RASTA-PLP into speech recognition (Abstract)

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

On the importance of the microphone position for speech recognition in the car (Abstract)

Adaptation to new microphones using tied-mixture normalization (Abstract)

Noise independent speech recognition for a variety of noise types (Abstract)

Root adaptive homomorphic deconvolution schemes for speech recognition in noise (Abstract)

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

Adaptive noise immunity learning for word spotting (Abstract)

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

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

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

Speech spectrum transformation by speaker interpolation (Abstract)

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

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

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

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

Analysis of spectral interpolation with weighting dependent on frame energy (Abstract)

Filter bank design based on discriminative feature extraction (Abstract)

Vector quantization with hyper-columnar clusters (Abstract)

Segmental phoneme recognition using piecewise linear regression (Abstract)

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

Efficient vector quantisation of LPC parameters for noisy channels (Abstract)

Efficient interblock noiseless coding of speech LPC parameters (Abstract)

Variable dimension vector quantization of linear predictive coefficients of speech (Abstract)

Spectral quantization of cepstral coefficients (Abstract)

Vector quantization-lattice vector quantization of speech LPC coefficients (Abstract)

Segmental quantization of speech spectral information (Abstract)

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

Single stage spectral quantization at 20 bits (Abstract)

High-order allpole modelling of the spectral envelope (Abstract)

A new method for segmenting continuous speech (Abstract)

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

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

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

Towards large vocabulary Mandarin Chinese speech recognition (Abstract)

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

New graph search techniques for speech recognition (Abstract)

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

Comparative experiments on large vocabulary speech recognition (Abstract)

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

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

Stochastic trajectory modeling for speech recognition (Abstract)

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

A new waveform speech synthesis approach based on the COC speech spectrum (Abstract)

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

Articulatory speech synthesis based on fractional delay waveguide filters (Abstract)

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

Automatic generation of prosodic rules for speech synthesis (Abstract)

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

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

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

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

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

Using Gaussian mixture modeling in speech recognition (Abstract)

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

The influence of speech coding algorithms on automatic speech recognition (Abstract)

Discriminative training of high performance speech recognizer using N best candidates (Abstract)

Model topology selection for isolated word recognition (Abstract)

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

On the fuzzy vector quantization based hidden Markov model (Abstract)

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

Word accent patterns modelling by concatenation of mora hidden Markov models (Abstract)

Phonemic segmentation of fluent speech (Abstract)

Knowledge based approach to consonant recognition (Abstract)

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

Constructing telephone acoustic models from a high-quality speech corpus (Abstract)

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

Towards automatic collection of the US census (Abstract)

The Australian National Database of Spoken Language (Abstract)