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

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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

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

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ASR system modeling for automatic evaluation and optimization of dialogue systems

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Utterance-level boosting of HMM speech recognizers

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Classifier design for verification of multi-class recognition decision

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Using prosodic and lexical information for speaker identification

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Packet loss concealment based on sinusoidal extrapolation

Multiple description speech coding with diversities

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Context-based error recovery technique for GSM AMR speech codec

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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

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Confidence scoring based on backward language models

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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

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Speech enhancement based on auditory spectral change

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Robust HMM-based speech/music segmentation

Porting an audio partitioner across domains

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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

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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

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

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

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Short-time Gaussianization for robust speaker verification

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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

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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

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Language model adaptation through topic decomposition and MDI estimation

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

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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

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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

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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

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

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

The HMM error model

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Modeling inverse covariance matrices by basis expansion

Factor analysed hidden Markov models

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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

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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

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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

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

Content Provider	IEEE Xplore Digital Library
Author	Kawahara, Tatsuya Hasegawa, Masahiro
Copyright Year	2002
Description	Author affiliation: School of Informatics, Kyoto University, Sakyo-ku, 606-8501, Japan (Kawahara, Tatsuya; Hasegawa, Masahiro)
Abstract	Automatic detection of section (sub-topic) boundaries in lecture speech is addressed. The method makes use of the characteristic expressions used in initial utterances of sections defined as discourse makers, as well as pause and language model information. The discourse markers are derived in a totally unsupervised manner based on word statistics used in the information retrieval technique. The statistics is used to select candidates picked up by other information. Experimental results show that the proposed method realizes better indexing performance (better precision at high recall rates) than the simple baseline method using pause information only. Moreover, it is shown to be robust against speech recognition errors.
File Size	1637241
File Format	PDF
ISBN	0780374029
ISSN	15206149
DOI	10.1109/ICASSP.2002.5743639
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	2002-05-13
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Speech Manuals Machine assisted indexing Radio access networks Soil Computational modeling Switches
Content Type	Text

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