NDLI: A time domain reformulation of linear prediction equivalent to the LSP decomposition

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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

Volume 4 - 2002

Volume 3 - 2002

Volume 2 - 2002

Volume 1 - 2002

Automatic indexing of lecture speech by extracting topic-independent discourse markers

Audio Indexing of Arabic broadcast news

Automatic speech summarization applied to English broadcast news speech

Transcribing audio-video archives

Learning new user formulations in automatic Directory Assistance

A scalable architecture for Directory Assistance automation

Combination of boosting and discriminative training for natural language call steering systems

Combining prior knowledge and boosting for call classification in spoken language dialogue

Extracting descriptive noun phrases from conversational speech

Adaptive language models for spoken dialogue systems

Evaluation of spoken language grammar learning in the ATIS domain

ASR system modeling for automatic evaluation and optimization of dialogue systems

Digit recognition in noisy environments via a sequential GMM/SVM system

Robust speech recognition in Noisy Environments: The 2001 IBM spine evaluation system

Uncertainty decoding with SPLICE for noise robust speech recognition

Accounting for uncertainity in observations: A new paradigm for Robust Automatic Speech Recognition

Missing data speech recognition in reverberant conditions

Combining the union model and missing feature method to improve noise robustness in ASR

Asymmetrical Support Vector Machines and applications in speech processing

Using SVMS and discriminative models for speech recognition

Maximum mutual information based acoustic-features representation of phonological features for speech recognition

Improved cross-task recognition using MMIE training

Direct models for phoneme recognition

Structurally discriminative graphical models for automatic speech recognition - results from the 2001 Johns Hopkins Summer Workshop

A new algorithm for fast discriminative training

An alternative approach of finding competing hypotheses for better minimum classification error training

Minimum Phone Error and I-smoothing for improved discriminative training

Utterance-level boosting of HMM speech recognizers

A dynamic in-search discriminative training approach for large vocabulary speech recognition

Classifier design for verification of multi-class recognition decision

Speaker indexing for retrieval of voicemail messages

Co-channel speaker segment separation

Speaker detection and tracking for telephone transactions

Noise-robust open-set speaker recognition using noise-dependent Gaussian mixture classifier

Automatic estimation of one's age with his/her speech based upon acoustic modeling techniques of speakers

Using prosodic and lexical information for speaker identification

Speaker identification using multilingual phone strings

Gender-dependent phonetic refraction for speaker recognition

Speaker identification using online, frame dependent, and diffusive variance adaptation

Speaker recognition using adaptively boosted decision tree classifier

Generalized linear discriminant sequence kernels for speaker recognition

Mode adaptive Unequal Error Protection for transform predictive speech and audio coders

A packet loss recovery method using packet arrived behind the playout time for CELP decoding

Packet loss concealment based on sinusoidal extrapolation

Multiple description speech coding with diversities

Pyramid CELP: Embedded speech coding for packet communications

Context-based error recovery technique for GSM AMR speech codec

Noise adaptive speech recognition in time-varying noise based on sequential kullback proximal algorithm

Comparing Jacobian adaptation with cepstral mean normalization and parallel model combination for noise robust speech recognition

Jacobian joint adaptation to noise, channel and vocal tract length

Dynamic estimation of a noise over estimation factor for Jacobian-based adaptation

Efficient second-order adaptation for large vocabulary distributed speech recognition

Cepstrum-domain model combination based on decomposition of speech and noise for noisy speech recognition

Automatic confidence score mapping for adapted speech recognition systems

Confidence scoring for accurate HMM-based word recognition by using SM-based monophone score normalization

Confidence scoring based on backward language models

A concept graph based confidence measure

Probabilistic integration of multiple confidence measures and context information for concept verification

Estimating semantic confidence for spoken dialogue systems

An upper bound on the quality of artificial bandwidth extension of narrowband speech signals

Enhancing quality of CELP coded speech via wideband extension by using voicing GMM interpolation and HNM re-synthesis

Speech signal band width extension and noise removal using subband HMN

Gender-dependent and speaker-dependent speech enhancement

Speech enhancement using MMSE short time spectral estimation with gamma distributed speech priors

Speech enhancement based on auditory spectral change

Improved spectral tracking using interpolated linear prediction parameters

Improvement of the restricted temporal decomposition method for line spectral frequency parameters

Circular LPC modeling and constant pitch transform for accurate speech analysis and high quality speech synthesis

The relation between speech segment selectivity and source localization accuracy

Joint optimization of model and excitation in parametric speech coders

Robust formant tracking in noise

Noise-robust cellular phone speech recognition using codec-adapted speech and noise models

On the joint use of noise reduction and MLLR adaptation for in-car hands-free speech recognition

Unified speech recognition for the landline and wireless environments

Robust HMM-based speech/music segmentation

Porting an audio partitioner across domains

Continuous audio-visual digit recognition using decision fusion

Robust bi-modal speech recognition based on state synchronous modeling and stream weight optimization

Enhanced posteriors bias prediction for robust multi-stream ASR combining voicing and estimate reliabilities

Robust phoneme discrimination using acoustic waveforms

Phone based voice activity detection using online Bayesian adaptation with conjugate normal distributions

Discriminative training of language models for speech recognition

Fundamental frequency estimation based on instantaneous frequency amplitude spectrum

Pitch determination and voice quality analysis using Subharmonic-to-Harmonic Ratio

Spline-based continuous-time pitch estimation

Pitch estimation based on Circular AMDF

Robust pitch determination using nonlinear state-space embedding

The DYPSA algorithm for estimation of glottal closure instants in voiced speech

Robust pitch tracking in the car environment

A statistical pitch detection algorithm

Yet Another Algorithm for Pitch Tracking

Recovery of the harmonic fundamental from the mixed partial derivatives of the STFT phase

A multi-pitch tracking algorithm for noisy speech

Data-driven temporal filters for robust features in speech recognition obtained via Minimum Classification Error (MCE)

Modulation features for speech recognition

Discriminative auditory features for robust speech recognition

Perceptual speech modeling for noisy speech recognition

Acoustic analysis and recognition of whispered speech

Blind channel estimation based on speech correlation structure

Noise-robust speech recognition using a new spectral estimation method “PHASOR”

Non-linear transformations of the feature space for robust Speech Recognition

Noise robust speech recognition using Gaussian basis functions for non-linear likelihood function approximation

VTS residual noise compensation

A comparative analysis of UK and US English accents in recognition and synthesis

Improved duration control for speech synthesis using a multigram language model

Towards a multilingual prosody model for text-to-speech

Pitch and speech-rate conversion using envelope modulation modeling

A phase generation method for speech reconstruction from spectral envelope and pitch intervals

Real-time speech synthesis on an ultra low-resource, programmable DSP system

Determination of articulatory movements from speech acoustics using an HMM-based speech production model

Similarity structure in perceptual and physical measures for visual Consonants across talkers

Articulatory speech synthesis based upon fluid dynamic principles

Analysis by synthesis of FM modulation and aspiration noise components in pathological voices

Domain adaptation for TTS systems

The synthesis unit generation algorithm for Mandarin TTS

Robust splicing costs and efficient search with BMM Models for concatenative speech synthesis

Unit selection algorithm for Japanese speech synthesis based on both phoneme unit and diphone unit

Synthesis of unseen context and spectral and pitch contour smoothing in concatenated text to speech synthesis

Speech segment selection for concatenative synthesis based on prosody-aligned distance measure

A data-driven method for input feature selection within neural prosody generation

Corpus based extraction of quantitative prosodic parameters of stress groups in Spanish

Assigning phrase accent to Chinese Text-to-Speech system

Generation of Chinese prosodic phrasing rules by an extension matrix algorithm

Autoregressive acoustical modelling of free field cough sound

Analysis of syllable duration models for Mandarin speech

Study on prosodic boundary location in Chinaese mandarin

Analysis of pitch contour of Thai tone using Fujisaki's model

A method for automatic extraction of model parameters from fundamental frequency contours of speech

Robust parameters for automatic segmentation of speech

A simple approach to non-uniform vowel normalization

DCT based pseudo complex cepstrum

Gaussian mixture model based mutual information estimation between frequency bands in speech

On the approximate W-disjoint orthogonality of speech

Speech analysis and feature extraction using chaotic models

Signal subspace speech enhancement with perceptual post-filtering

Speech enhancement using excitation source information

Noise suppression with synthesis windowing and pseudo noise injection

Speech enhancement by harmonic modeling via map pitch tracking

Monaural speech segregation based on pitch tracking and amplitude modulation

Acoustic diversity for improved speech recognition in reverberant environments

Speech enhancement for non-stationary noise environment by adaptive wavelet packet

Noise suppression based on approximate KLT with wavelet packet expansion

A perceptual signal subspace approach for speech enhancement in colored noise

A subspace approach for enhancing speech corrupted by colored noise

Fast model adaptation and complexity selection for nonnative English speakers

Rapid adaptation with linear combinations of rank-one matrices

Structuring linear transforms for adaptation using training time information

Non-uniform scaling based speaker normalization

Acoustic-phonetic speech parameters for speaker-independent speech recognition

Piecewise linear constraints for model space adaptation

On maximum mutual information speaker-adapted training

Supervised adaptation of MCE-trained CDHMMS using minimum classification error linear regression

Speaker selection training for large vocabulary continuous speech recognition

Efficient adaptation text design based on the Kullback-Leibler measure

Unsupervised incremental online adaptation to unknown environment and speaker

Perception-based selective encryption of G.729 speech

A methodology for speech codec FEC design and optimisation based upon the ITU-T Rec. P.861 objective speech quality measure

A 4 kb/s hybrid MELP/CELP speech coding candidate for ITU standardization

Generalized analysis-by-synthesis based on system identification

Low complexity recursive coding of spectrum parameters

Spectral quantization by companding

KLT-based classified VQ for the speech signal

A novel quantization scheme for the noise-like component in waveform interpolation speech coding

Scalable innovation coding

A new bandwidth scalable wideband speech/audio coder

A time domain reformulation of linear prediction equivalent to the LSP decomposition

All-pole modeling technique based on the Weighted Sum of the LSP polynomials

Evaluation of kernel methods for speaker verification and identification

Using nonstandard SVM for combination of Speaker Verification and Verbal Information Verification in speaker authentication system

Speaker verification using text-constrained Gaussian Mixture Models

Short-time Gaussianization for robust speaker verification

Automatic speechreading with application to speaker verification

A Monte-Carlo method for score normalization in Automatic Speaker Verification using Kullback-Leibler distances

Improving speaker verification with figure of merit training

The sphericity measure for cellular speaker verification

Combining stochastic feature transformation and handset identification for telephone-based speaker verification

Fast LM look-ahead for large vocabulary continuous speech recognition using perfect hashing

Efficient language model lookahead through polymorphic linguistic context assignment

Recent advances in efficient decoding combining on-line transducer composition and smoothed language model incorporation

Doing away with the Viterbi approximation

The 2001 BYBLOS English large vocabulary conversational speech recognition system

Speaking-rate dependent decoding and adaptation for spontaneous lecture speech recognition

Analysis on individual differences in automatic transcription of spontaneous presentations

Towards automatic corpus preparation for a German broadcast news transcription system

Speech labeling and the most frequent phrase extraction using same section in a presentation speech

Recognition of Chinese names in continuous speech for directory assistance applications

Dynamic generation of proper name pronunciations for directory assistance

Flemish accent identification based on formant and duration features

Merging segmental and rhythmic features for Automatic Language Identification

Language identification using Gaussian mixture model tokenization

Mixed-memory Markov models for Automatic Language Identification

Connectionist language modeling for large vocabulary continuous speech recognition

Fast update of latent semantic spaces using a linear transform framework

Language model adaptation through topic decomposition and MDI estimation

Building a topic-dependent maximum entropy model for very large corpora

Turn-Based Language Modeling for spoken dialog systems

Rescoring effectiveness of language models using different levels of knowledge and their integration

Text normalization with varied data sources for conversational speech language modeling

A low-power, fixed-point, front-end feature extraction for a distributed speech recognition system

A comparison of front-end configurations for robust speech recognition

Increased mfcc filter bandwidth for noise-robust phoneme recognition

Including detailed information feature in MFCC for large vocabulary contious speech recornition

Hierarchical tandem feature extraction

Optimizing feature extraction for English word recognition

Towards knowledge-based features for HMM based large vocabulary automatic speech recognition

On feature extraction by mutual information maximization

Application of variational Bayesian PCA for speech feature extraction

A Bayesian approach to speech feature enhancement using the dynamic cepstral prior

Mandarin verbal information verification

Auditory-based acoustic distinctive features and spectral cues for automatic speech recognition using a multi-stream paradigm

Multi-dialectal Spanish speech recognition

Context-dependent acoustic modeling using graphemes for large vocabulary speech recognition

Training of across-word phoneme models for large vocabulary continuous speech recognition

Maximum entropy and MCE based HMM stream weight estimation for audio-visual ASR

Implementation of automatic capitalisation generation systems for speech input

Transcription of out-of-vocabulary words in large vocabulary speech recognition based on phoneme-to-grapheme conversion

Automatic new word extraction method

Efficient reduction of Gaussian components using MDL criterion for HMM-based speech recognition

Compact decision trees with cluster validity for speech recognition

Unsupervised acoustic model training

A beamforming approach to permutation alignment for multichannel frequency-domain blind speech separation

Soft constrained subband beamforming for hands-free speech enhancement

Hands-free continuous speech recognition in noise using a speaker beam-former based on spectrum-entropy

Talker localization in a real acoustic environment based on DOA estimation and statistical sound source identification

Speech recognizer-based microphone array processing for robust hands-free speech recognition

Microphone array post-filtering for non-stationary noise suppression

Microphone array post-filter for diffuse noise field

Noise reduction using a small-scale microphone array in multi noise source environment

Blind separation of non-linear convolved speech mixtures

Bund source separation based on Multi-Stage ICA combining frequency-domain ICA and time-domain ICA

Blind source separation based on fast-convergence algorithm using ICA and beamforming for real convolutive mixture

Adaptation experiments on the SPINE database with the Extended Maximum Likelihood Linear Transformation (EMLLT) model

Increasing speech recognition robustness with HMM2

Two dissimilarity measures for HMMS and their application in phoneme model clustering

The HMM error model

On application of the Faragó-Lugosi algorithm in speech recognition

Modeling inverse covariance matrices by basis expansion

Factor analysed hidden Markov models

On the use of high order derivatives for high performance alphabet recognition

A recognition method with parametric trajectory synthesized using direct relations between static and dynamic feature vector time series

A mixture linear model with target-directed dynamics for spontaneous speech recognition

A new combined model of statics-dynamics of speech

A sequence-based generalization of mean-field annealing using the Forward/Backward algorithm: Application to image segmentation

Universal approximation of fully complex feed-forward neural networks

A normalised complex backpropagation algorithm

Robust on-line Principal Component Analysis based on a fixed-point approach

Time series prediction based on the Relevance Vector Machine with adaptive kernels

A Bayesian model selection criterion for HMM topology optimization

Blind source separation of time-varying instantaneous mixtures using an on-line algorithm

Robust blind source separation and dispersing algorithms

Polar coordinate based nonlinear function for frequency-domain blind source separation

Is gradient descent appropriate for entropy-based source separation?

On complex infomax applied to functional MRI data

Location-based sound segregation

On the training of DS-CDMA neural-network receivers

Extracting noise-robust features from audio data

A new subband non linear prediction coding algorithm for narrowband speech signal: The nADPCMB⊥MLT coding scheme

An artificial neural network-based scheme for robust watermarking of audio signals

An RNN-based channel classification for Mandarin speech recognition over GSM/PSTN transmission environments

A new nonlinear prediction model based on the Recurrent Neural Predictive Hidden Markov Model for speech enhancement

Hierarchical cluster model for perceptual image processing

A human-perception-like image recognition system based on PAP vector representation with multi resolution concept

Outlier estimation and detection application to skin lesion classification

Offline recognition of handwritten Chinese characters using Gabor features, CDHMM modeling and MCE training

Natural gradient learning neural networks for modeling and identification of nonlinear systems with memory

On sensitivity of neural adaptive filters with respect to the slope parameter of a neuron activation function

A discriminative training criterion and an associated EM learning algorithm

Simultaneous extraction of Principal Components using givens rotations and output variances

Fast algorithm for adaptive estimation of principal and minor components

Lip segmentation by fuzzy clustering incorporating with shape function

A recurrent RBF network for non-linear channel with time-varying characteristic

FFT coefficient memory reduction technique for OFDM applications

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

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

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