NDLI: A Noise Robust Front-End with Low Computational Cost for Embedded In-Car Speech Recognition

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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

Volume 4 - 2007

2007 IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP 2006 Conference Committee

Technical Program Committee

Review Committee

General Chair's Welcome

Future Signal Processing Society Meetings

Table of contents

Statistical Sentence Extraction for Information Distillation

A Discriminative Training Framework using N-Best Speech Recognition Transcriptions and Scores for Spoken Utterance Classification

Spoken Language Understanding Strategies on the France Telecom 3000 Voice Agency Corpus

Dynamic Bayesian Networks and Discriminative Classifiers for Multi-Stage Semantic Interpretation

The Relevance of Voice Quality Features in Speaker Independent Emotion Recognition

Detection and Analysis of Abnormal Situations Through Fear-Type Acoustic Manifestations

Trigger-Based Language Modeling using a Loss-Sensitive Perceptron Algorithm

Gaussian Mixture Language Models for Speech Recognition

Topic-Independent Speaking-Style Transformation of Language Model for Spontaneous Speech Recognition

Large-Scale Distributed Language Modeling

Correlated Latent Semantic Model for Unsupervised LM Adaptation

Discriminative Training of Decoding Graphs for Large Vocabulary Continuous Speech Recognition

Robust Speaker Recognition with Cross-Channel Data: MIT-LL Results on the 2006 NIST SRE Auxiliary Microphone Task

Constrained MLLR for Speaker Recognition

Session Variability Subspace Projection Based Model Compensation for Speaker Verification

A Generalized Feature Transformation Approach for Channel Robust Speaker Verification

Improved Methods for Characterizing the Alternative Hypothesis using Minimum Verification Error Training for LLR-Based Speaker Verification

Triage Framework for Resource Conservation in a Speaker Identification System

Open-Vocabulary Spoken Utterance Retrieval using Confusion Networks

Fast Unconstrained Audio Search in Numerous Human Languages

Optimal Estimation of Rejection Thresholds for Topic Spotting

Two-Stage Method for Specific Audio Retrieval

Divergence-Based Similarity Measure for Spoken Document Retrieval

Castsearch - Context Based Spoken Document Retrieval

Improving Speech Transcription for Mandarin-English Translation

On Efficient Coupling of ASR and SMT for Speech Translation

Consensus Network Decoding for Statistical Machine Translation System Combination

Phonetic Transcription using Speech Recognition Technique Considering Variations in Pronunciation

Speech Translation with Phrase Based Stochastic Finite-State Transducers

Language Model Adaptation in Machine Translation from Speech

Spoken Query Processing for Information Retrieval

Automatic Detection of Sentence and Clause Units using Local Syntactic Dependency

Improving Automatic Call Classification using Machine Translation

DBN Based Joint Dialogue Act Recognition of Multiparty Meetings

Semi-Supervised Learning for Part-of-Speech Tagging of Mandarin Transcribed Speech

Structures for Spoken Language Understanding: A Two-Step Approach

Speech-Based Interactive Information Guidance System using Question-Answering Technique

The Hidden Information State Approach to Dialog Management

Confidence Measures for Semi-Automatic Labeling of Dialog Acts

Insights Gained from Development and Long-Term Operation of a Real-Environment Speech-Oriented Guidance System

Directory Retrieval using Voice Form-Filling

Learning to Ground in Spoken Dialogue Systems

Word Topical Mixture Models for Dynamic Language Model Adaptation

Unsupervised Languagemodel Adaptation for Meeting Recognition

Data Driven Approach for Language Model Adaptation using Stepwise Relative Entropy Minimization

Joint Morphological-Lexical Language Modeling (JMLLM) for Arabic

Comparing Evaluation Metrics for Sentence Boundary Detection

Entropy Based Classifier Combination for Sentence Segmentation

Efficient and Robust Language Modeling in an Automatic Children's Reading Tutor System

Boosting of Prosodic and Pronunciation Features to Detect Mispronunciations of Non-Native Children

Generalized Segment Posterior Probability for Automatic Mandarin Pronunciation Evaluation

CDF-Matching for Automatic Tone Error Detection in Mandarin Call System

Development of Articulation Assessment and Training System with Speech Recognition and Articulation Training Strategies Selection

Capturing Spontaneous Conversation and Social Dynamics: A Privacy-Sensitive Data Collection Effort

The MIT-LL/IBM 2006 Speaker Recognition System: High-Performance Reduced-Complexity Recognition

STBU System for the NIST 2006 Speaker Recognition Evaluation

A Generative-Discriminative Framework using Ensemble Methods for Text-Dependent Speaker Verification

Parameterization of Prosodic Feature Distributions for SVM Modeling in Speaker Recognition

An Evaluation of Audio-Visual Person Recognition on the XM2VTS Corpus using the Lausanne Protocols

Automatic Weighting for the Combination of TDOA and Acoustic Features in Speaker Diarization for Meetings

Noise Robust Speaker Identification for Spontaneous Arabic Speech

NAP and WCCN: Comparison of Approaches using MLLR-SVM Speaker Verification System

Word-Conditioned Phone N-Grams for Speaker Recognition

Language Normalization for Bilingual Speaker Recognition Systems

Finding Speaker Identities with a Conditional Maximum Entropy Model

A Statistical Approach to Performance Evaluation of Speaker Recognition Systems

P-Value Segment Selection Technique for Speaker Verification

Model Complexity Selection and Cross-Validation EM Training for Robust Speaker Diarization

Incorporating Auditory Feature Uncertainties in Robust Speaker Identification

Svm-Based Speaker Verification by Location in the Space of Reference Speakers

Complementary Features for Speaker Verification Based on Genetic Algorithms

Acoustic Model Enhancement: An Adaptation Technique for Speaker Verification Under Noisy Environments

Text-Independent Speaker Verification in Embedded Environments

Development of a Femininity Estimator using Speaker Recognition Techniques for Voice Therapy of Gender Identity Disorder Clients

Effects of Device Mismatch, Language Mismatch and Environmental Mismatch on Speaker Verification

A New Segmentation Algorithm Combined with Transient Frames Power for Text Independent Speaker Verification

Combined Inter-Frame and Intra-Frame Fast Scoring Methods for Efficient Implementation of GMM-Based Speaker Verification Systems

Comparison of Large Margin Training to Other Discriminative Methods for Phonetic Recognition by Hidden Markov Models

Approximating the Kullback Leibler Divergence Between Gaussian Mixture Models

Evaluation of Proposed Modifications to MPE for Large Scale Discriminative Training

Integration of Multiple Feature Sets for Reducing Ambiguity in ASR

A Constrained Line Search Optimization for Discriminative Training in Speech Recognition

Generalization of Linear Discriminant Analysis used in Segmental Unit Input HMM for Speech Recognition

Complementary System Generation using Directed Decision Trees

System Combination by Driven Decoding

A Generalized Dynamic Composition Algorithm of Weighted Finite State Transducers for Large Vocabulary Speech Recognition

The IBM 2006 Gale Arabic ASR System

Unsupervised Training for Mandarin Broadcast News and Conversation Transcription

The AMI System for the Transcription of Speech in Meetings

Feature Compensation using More Accurate Statistics of Modeling Error

Unsupervised Stream Weight Estimation using Anti-Models

A New Concept for Feature-Domain Dereverberation for Robust Distant-Talking ASR

Word Graph Based Feature Enhancement for Noisy Speech Recognition

Stereo-Based Stochastic Mapping for Robust Speech Recognition

Missing Feature Speech Recognition using Dereverberation and Echo Suppression in Reverberant Environments

Two-Microphone Voice Activity Detection Based on the Homogeneity of the Direction of Arrival Estimates

Adaptive Training with Joint Uncertainty Decoding for Robust Recognition of Noisy Data

Segmental Modeling for Audio Segmentation

Fundamental Frequency Estimation Based on Pitch-Scaled Harmonic Filtering

Continuous Electromyographic Speech Recognition with a Multi-Stream Decoding Architecture

Mel-Spectrographic Mask Estimation for Missing Data Speech Recognition using Short-Time-Fourier-Transform Ratio Estimators

Semi-Synchronous Speech and Pen Input

GMM Supervector Based SVM with Spectral Features for Speech Emotion Recognition

State Synchronous Modeling on Phone Boundary for Audio Visual Speech Recognition and Application to Muti-View Face Images

Unsupervised Lexicon Acquisition from Speech and Text

A Novel Phone-State Matrix Based Vocabulary-Indenendent Keyword Spotting Method for Spontaneous Speech

Profile View Lip Reading

Outlier Correction for Local Distance Measures in Example Based Speech Recognition

Cross-Validation EM Training for Robust Parameter Estimation

Further Experiments with Detector-Based Conditional Random Fields in Phonetic Recognition

Use of Differential Cepstra as Acoustic Features in Hidden Trajectory Modeling for Phonetic Recognition

N-Best Rescoring for Speech Recognition using Penalized Logistic Regression Machines with Garbage Class

Estimation of General Identifiable Linear Dynamic Models with an Application in Speech Recognition

Information Theoretic Analysis of Direct Articulatory Measurements for Phonetic Discrimination

Physiological Feature Extraction for Text Independent Speaker Identification using Non-Uniform Subband Processing

An Investigation into the Correlation and Prediction of Acoustic Speech Features from MFCC Vectors

An Approach to Formant Frequency Estimation at Low Signal-to-Noise Ratio

The Effect of Spectral Space Reduction in Spontaneous Speech on Recognition Performances

Joint Estimation of Formant Trajectories via Spectro-Temporal Smoothing and Bayesian Techniques

Segregation of Speakers for Speaker Adaptation in TV News Audio

Speaker Clustering Based on Minimum Rand Index

Speaker Diarization: Towards a More Robust and Portable System

Classification of Acoustic Maps to Determine Speaker Position and Orientation from a Distributed Microphone Network

Robust Speaker Localization in Meeting Room Domain

Efficient Synthesis of Approximately Consistent Graphs for Acoustic Multi-Source Localization

Conditional Vector Quantization for Voice Conversion

A Novel Method for Prosody Prediction in Voice Conversion

Towards a Voice Conversion System Based on Frame Selection

Full HMM Training for Minimizing Generation Error in Synthesis

LSM-Based Unit Pruning for Concatenative Speech Synthesis

A New Method for Speech Synthesis and Transformation Based on an ARX-LF Source-Filter Decomposition and HNM Modeling

ITU-T G.729.1: AN 8-32 Kbit/S Scalable Coder Interoperable with G.729 for Wideband Telephony and Voice Over IP

Backwards Compatible Wideband Telephony in Mobile Networks: CELP Watermarking and Bandwidth Extension

The Broadvoice Speech Coding Algorithm

An Optimal Unit-Selection Algorithm for Ultra Low Bit-Rate Speech Coding

Transform Audio Coding with Arithmetic-Coded Scalar Quantization and Model-Based Bit Allocation

A Candidate for the ITU-T G.722 Packet Loss Concealment Standard

Enhancement of Speech Signals Under Multiple Hypotheses using an Indicator for Transient Noise Presence

Overcoming the Vector Taylor Series Approximation in Speech Feature Enhancement - A Particle Filter Approach

A Comparative Intelligibility Study of Speech Enhancement Algorithms

Binaural Cue Preservation for Hearing Aids using an Interaural Transfer Function Multichannel Wiener Filter

ICA-Based MAP Algorithm for Speech Signal Enhancement

Noise Eigenvalue Modification Methods for Spatial Subspace Based Multi-Channel Speech Enhancement

Multisensor Dynamic Waveform Fusion

Speech Noise Estimation using Enhanced Minima Controlled Recursive Averaging

Visually-Derived Wiener Filters for Speech Enhancement

Multi-Sensor Spectro-Temporal Comb Filtering for Speech Enhancement

Speech Bandwidth Extension by Data Hiding and Phonetic Classification

Bandwidth Expansionwith a pólya URN Model

A Scalable Bandwidth Extension Algorithm

Feasibility of Single Channel Speaker Separation Based on Modulation Frequency Analysis

High-Presence Hearing-Aid System using DSP-Based Real-Time Blind Source Separation Module

Speech Intelligibility Enhancement using Tunable Equalization Filter

Voice-Melody Transcription Under a Speech Recognition Framework

Articulatory Feature-Based Methods for Acoustic and Audio-Visual Speech Recognition: Summary from the 2006 JHU Summer workshop

Latent Prosody Model of Continuous Mandarin Speech

Incorporating Training Errors for Large Margin HMMS Under Semi-Definite Programming Framework

Combining Discriminative Feature, Transform, and Model Training for Large Vocabulary Speech Recognition

Mandarin Accent Analysis Based on Formant Frequencies

Modeling Duration via Lattice Rescoring

An Articulatory Feature-Based Tandem Approach and Factored Observation Modeling

Gammatone Features and Feature Combination for Large Vocabulary Speech Recognition

Approximate Test Risk Minimization Through Soft Margin Estimation

An Acoustic Model Based on Kullback-Leibler Divergence for Posterior Features

A Bayesian Approach for Phonetic Decision Tree State Tying in Conversational Speech Recognition

Word-Level Tone Modeling for Mandarin Speech Recognition

Varying Time Constants and Gain Adaptation in Feature Extraction for Speech Processing

Pronunciation Modeling for Spontaneous Speech Recognition using Latent Pronunciation Analysis (LPA) and Prior Knowledge

A New Minimum Divergence Approach to Discriminative Training

Improving High Quality TTS using Circular Linear Prediction and Constant Pitch Transform

DCT-Based Amplitude and Frequency Modulated Harmonic-Plus-Noise Modelling for Text-to-Speech Synthesis

Conversion Function Clustering and Selection for Expressive Voice Conversion

Cost Reduction of Training Mapping Function Based on Multistep Voice Conversion

Database Mining for Flexible Concatenative Text-to-Speech

The Application and Computer Simulation of Multi-Channel Cochlear Implant Based on All Phase DFT Filter

Inequality Maximum Entropy Classifier with Character Features for Polyphone Disambiguation in Mandarin TTS Systems

A Novel HMM-Based TTS System using Both Continuous HMMS and Discrete HMMS

Script Design Based on Decision Tree with Context Vector and Acoustic Distance for Mandarin TTS

Extraction of Linguistic Information with the AID of Acoustic Data to Build Speech Systems

Acoustic Monitoring of Singing Insects

Musical Instrument Spectral Envelope Sensitivity

Importance of the Dynamic Range of an Analysis Windowfunction for Phase-Only and Magnitude-Only Reconstruction of Speech

A Duplex Theory of Spike Coding in the Early Stages of the Auditory System

Proposal of Synchronization Indexes of Single Neuron Activity on Periodic Stimulus

Identifying Impaired Cochlear Implant Channels via Speech-Token Confusion Matrix Analysis

A Statistical Acoustic Confusability Metric Between Hidden Markov Models

Dual Parameters for Voiced-Unvoiced Speech Signal Determination

Latent Correlation Analysis of HMM Parameters for Speech Recognition

Probabilistic and Bottle-Neck Features for LVCSR of Meetings

Constraint Induction of Phonetic-Acoustic Decision Trees for Crosslingual Acoustic Modelling

Representation of Phonemes in Primary Auditory Cortex: How the Brain Analyzes Speech

Incremental Adaptation Based on a Macroscopic Time Evolution System

A Maximum Likelihood Approach to Unsupervised Online Adaptation of Stochastic Vector Mapping Function for Robust Speech Recognition

Efficient, Low Latency Adaptation for Speech Recognition

Modelling Pronunciation Variation using Multi-Path HMMS for Syllables

A Hidden-State Maximum Entropy Model Forword Confidence Estimation

Exploiting Uncertainties for Binaural Speech Recognition

An Auditory Neural Feature Extraction Method for Robust Speech Recognition

Noise Robust Voice Activity Detection Based on Statistical Model and Parallel Non-Linear Kalman Filtering

Revised Contextual LRT for Voice Activity Detection

Optimization of Temporal Filters in the Modulation Frequency Domain via Constrained Linear Discriminant Analysis (C-LDA) for Constructing Robust Features in Speech Recognition

Maximum Entropy Confidence Estimation for Speech Recognition

A Segmentation Posterior Based Endpointing Algorithm

Robust Distant Speech Recognition by Combining Position-Dependent CMN with Conventional CMN

Prosody Modelling of Spanish for Expressive Speech Synthesis

Use of Poisson Processes to Generate Fundamental Frequency Contours

Agreement Learning for Automatic Accent Annotation

A Speaker Adaptation Technique for MRHSMM-Based Style Control of Synthetic Speech

Head Movement Synthesis Based on Semantic and Prosodic Features for a Chinese Expressive Avatar

A Hybrid Signal-and-Link-Parametric Approach to Single-Ended Quality Measurement of Packetized Speech

Long-Term Quantization of Speech LSF Parameters

A Canonical Representation of Speech

A New Modulation Concept for Mixed Pseudo Analogue-Digital Speech and Audio Transmissioin

An Integrated Scheme for Robust Distributed Speech Recognition Over Lossy Packet Networks

Spoken Language Recognition with Relevance Feedback

Multi-Pass Pronunciation Adaptation

High-Accuracy Phone Recognition By Combining High-Performance Lattice Generation and Knowledge Based Rescoring

Improved Speech Recognition using Acoustic and Lexical Correlates of Pitch Accent in a N-Best Rescoring Framework

Three-Stage Error Concealment for Distributed Speech Recognition (DSR) with Histogram-Based Quantization (HQ) Under Noisy Environment

Noise Robust Phonetic Classificationwith Linear Regularized Least Squares and Second-Order Features

Self-Normalizing Dual Systems for Minor and Principal Component Extraction

Perceptual Gain Function for Eigenspectral Domain Speech Enhancement

A Soft-Decision Adaptation Mode Controller for an Efficient Frequency-Domain Generalized Sidelobe Canceller

Speech Enhancement Based on the Decomposition of Speech Into Deterministic and Stochastic Components and Psychoacoustic Model

Dual-Microphone Speech Dereverberation using a Reference Signal

Variational Bayesian Learning of Speech GMMS for Feature Enhancement Based on Algonquin

Echo Detection and Delay Estimation using a Pattern Recogntion Approach and Cepstral Correlation

Autoregressive Parameter Estimation for Kalman Filtering Speech Enhancement

Subband-Based Blind Signal Processing for Source Separation in Convolutive Mixtures of Speech

A Supervised Learning Approach to Monaural Segregation of Reverberant Speech

Improving Pronunciation Inference using N-Best List, Acoustics and Orthography

Keyword Search using Modified Minimum Edit Distance Measure

Phonetic Boundary Refinement using Support Vector Machine

Finding Maximum Margin Segments in Speech

Towards More Reality in the Recognition of Emotional Speech

Dynamic Stream Weight Modeling for Audio-Visual Speech Recognition

On Compressing N-Gram Language Models

Manual Transcription of Conversational Speech at the Articulatory Feature Level

Speech Emotion Recognition using Gaussian Mixture Vector Autoregressive Models

iLBC-Based Transparametrization: A Real Alternative to DSR for Speech Recognition Over Packet Networks

An Approach to Low Footprint Pronunciation Models for Embedded Speaker Independent Name Recognition

A Framework for Secure Speech Recognition

Spotting using Durational Entropy

Data-Driven Subvector Clustering using the Cross-Entropy Method

Multilingual Acoustic Models for Speech Recognition in Low-Resource Devices

Mobile CPU Based Optimization of Fast Likelihood Computation for Continuous Speech Recognition

Language Recognition with Word Lattices and Support Vector Machines

Dialect Classification on Printed Text using Perplexity Measure and Conditional Random Fields

The LIMSI 2006 TC-STAR EPPS Transcription Systems

Discriminative Vector for Spoken Language Recognition

N-Best Tokenization in a GMM-SVM Language Identification System

Acoustic Model Interpolation for Non-Native Speech Recognition

Language Identification using Acoustic Models and Speaker Compensated Cepstral-Time Matrices

Phone Set Generation Based on Acoustic and Contextual Analysis for Multilingual Speech Recognition

Normalizing the Speech Modulation Spectrum for Robust Speech Recognition

Enhanced SVM Training for Robust Speech Activity Detection

Speech Recognition using FHMMS Robust Against Nonstationary Noise

Effect of Speech and Noise Cross Correlation on AMFCC Speech Recognition Features

Unsupervised Speech/Non-Speech Detection for Automatic Speech Recognition in Meeting Rooms

Combination of Recognizers and Fusion of Features Approach to Missing Data ASR Under Non-Stationary Noise Conditions

A Noise Robust Front-End with Low Computational Cost for Embedded In-Car Speech Recognition

Considering Uncertainty by Particle Filter Enhanced Speech Features in Large Vocabulary Continuous Speech Recognition

Harmonic-Temporal Clustering of Speech for Single and Multiple F0 Contour Estimation in Noisy Environments

Evaluation of Pitch Detection Algorithms Under Real Conditions

AM-FM Demodulation of Spectrograms using Localized 2D Max-Gabor Analysis

Speech Segmentation using Extrema-Based Signal Track Length Measure

Spectral Estimation of Voiced Speech using a Family of MVDR Estimates

A Robust Pitch Estimation Algorithm in Noise

Pitch Estimation using Models of Voiced Speech on Three Levels

Stress and Emotion Classification using Jitter and Shimmer Features

Support Vector Regression for Automatic Recognition of Spontaneous Emotions in Speech

Comparison of Four Approaches to Age and Gender Recognition for Telephone Applications

Composite Permutation Coding with Simple Indexing for Speech/Audio Codecs

GMM-Based Entropy-Constrained Vector Quantization

Sequential Split Vector Quantization of LSF Parameters using Conditional Pdf

An 8-12 Kbit/S Embedded CELP Coder Interoperable with ITU-T G.729 CIDER: First Stage of the New G.729.1 Standard

Closed Loop Dynamic Bit Allocation for Excitation Parameters in Analysis-by-Synthesis Speech Codec

Efficient Frame Erasure Concealment in Predictive Speech Codecs using Glottal Pulse Resynchronisation

An Adaptive, Scalable Packet Loss Recovery Method

Stochastic Modeling and Quantization of Harmonic Phases in Speech using Wrapped Gaussian Mixture Models

Reconstructing Medical Dictations from Automatically Recognized and Non-Literal Transcripts with Phonetic Similarity Matching

Combination of Acoustic Classifiers Based on Dempster-Shafer Theory of Evidence

Novel Lookahead Decision Tree State Tying for Acoustic Modeling

Large-Margin Minimum Classification Error Training for Large-Scale Speech Recognition Tasks

A Temporal Auditory Model with Adaptation for Automatic Speech Recognition

Cross-Site and Intra-Site ASR System Combination: Comparisons on Lattice and 1-Best Methods

An Evaluation of Lattice Scoring using a Smoothed Estimate of Word Accuracy

Adequacy Analysis of Simulation-Based Assessment of Speech Recognition System

Applying a Hybrid Genetic Algorithm in the Design of a Self-Paced Brain Interface with a Low False Positive Rate

EEG Windowed Statisticalwavelet Deviation for Estimation of Muscular Artifacts

A Novel Measure for Synchrony and its Application to Neural Signals

A Time-Varying Phase Coherence Measure for Quantifying Functional Integration in the Brain

Multi-Scale EEG Brain Dynamics During Sustained Attention Tasks

Source Localization of EEG/MEG Data by Correlating Columns of ICA Solution with Lead Field Matrix

Survey of Parallel MATLAB Techniques and Applications to Signal and Image Processing

The Mathworks Distributed and Parallel Computing Tools for Signal Processing Applications

pMATLAB: Parallel MATLAB Library for Signal Processing Applications

Compiling Parallel MATLAB for General Distributions using Telescoping Languages

The Star-P High Performance Computing Platform

An Interactive Environment to Manipulate Large Graphs

Recent Advances and Challenges of Semantic Image/Video Search

Improving Image Search with PHETCH

The Mediamill Semantic Video Search Engine

PLSA on Large Scale Image Databases

Web Image Clustering

A Multimodal and Multilevel Ranking Framework for Content-Based Video Retrieval

Statistical Parametric Speech Synthesis

Model Adaptation Approach to Speech Synthesis with Diverse Voices and Styles

A Statistical Approach for Modeling Prosody Features using POS Tags for Emotional Speech Synthesis

Combining Gaussian Mixture Model with Global Variance Term to Improve the Quality of an HMM-Based Polyglot Speech Synthesizer

HMM-Based Hierarchical Unit Selection Combining Kullback-Leibler Divergence with Likelihood Criterion

One-to-Many and Many-to-One Voice Conversion Based on Eigenvoices

Experiences with Reproducible Research in Various Facets of Signal Processing Research

Reproducible Computational Experiments using Scons

Putting Reproducible Signal Processing into Practice: A Case Study in Watermarking

Reproducible Research: A Case Study of Sampling Signals with Finite Rate of Innovation

State-of-the-Art and Evolution in Public Data Sets and Competitions for System Identification, Time Series Prediction and Pattern Recognition

How to Encourage and Publish Reproducible Research

Speech Recognition System Combination for Machine Translation

Integrating Speech Recognition and Machine Translation

Arabic ASR and MT Integration for GALE

The Impact of ASR on Speech-to-Speech Translation Performance

Speech Translation Enhanced ASR for European Parliament Speeches - On the Influence of ASR Performance on Speech Translation

Speech Translation by Confusion Network Decoding

Equilibria of Adaptive Wavetable Oscillators with Applications to Beat Tracking

On the Use of Entropy for Beat Tracking Evaluation

Evaluating Low-Level Features for Beat Classification and Tracking

Beat Tracking using an Autocorrelation Phase Matrix

Rhythm and Tempo Analysis Toward Automatic Music Transcription

Sequential Inference of Rhythmic Structure in Musical Audio

Distributed Totally Asynchronous Iterative Waterfilling for Wideband Interference Channel with Time/Frequency Offset

Efficient Implementation of a Quasi-Maximum-Likelihood Detector Based on Semi-Definite Relaxation

Modeling file-sharing with BitTorrent-like incentives

Optimal Strategies for Efficient Peer-to-Peer File Sharing

Multipath Routing, Congestion Control and Dynamic Load Balancing

Fast Coper for Broadband Access: An Overview

A Survey of Dynamic Spectrum Access: Signal Processing and Networking Perspectives

On the Impact of Reconfigurable Antenna Arrays in Cognitive Radio

Compressed Sensing for Wideband Cognitive Radios

Squeezing the Most Out of Cognitive Radio: A Joint MAC/PHY Perspective

Coordination of Allocation Measurements in OFDM Based Ad Hoc Overlay Systems

Cognitive Dynamic Systems

Architecture and Applications of wireless Smart Cameras (Networks)

Layered and Collaborative Gesture Analysis in Multi-Camera Networks

Embedded Middleware on Distributed Smart Cameras

S3: The IBM Smart Surveillance System: From Transactional Systems to Observational Systems

Real-Time Distributed Tracking

A Sensor Network for Event Retrieval in a Home Like Ubiquitous Environment

Geometry and Manifolds for Independent Component Analysis

On the Design of Gradient Algorithms Employing Orthogonal Matrix Constraints

An Intrinsic CG Algorithm for Computing Dominant Subspaces

Generalised Fastica for Independent Subspace Analysis

Geometric Optimization Methods for Independent Component Analysis Applied on Gene Expression Data

Flag Manifolds for Subspace ICA Problems

Robust Search Methods for Music Signals Based on Simple Representation

Fast Recognition of Remixed Music Audio

Identifying `Cover Songs' with Chroma Features and Dynamic Programming Beat Tracking

A Comparison of Timbral and Harmonic Music Segmentation Algorithms

Automatic Bass Line Transcription from Streaming Polyphonic Audio

Active Music Listening Interfaces Based on Signal Processing

Volume 3 - 2007

Volume 2 - 2007

Volume 1 - 2007

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

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

A Noise Robust Front-End with Low Computational Cost for Embedded In-Car Speech Recognition

Content Provider	IEEE Xplore Digital Library
Author	Pei Ding Lei He Xiang Yan Rui Zhao Jie Hao
Copyright Year	2007
Description	Author affiliation: Toshiba Res. & Dev. Center, Beijing, China (Pei Ding; Lei He; Xiang Yan; Rui Zhao; Jie Hao)
Abstract	This paper proposes a noise robust front-end with low computational cost for embedded in-car speech recognition. The minimum mean-square error (MMSE) estimation algorithm is adopted to suppress the background noise, and in the gain function calculation a suitable piece-wise linear function is used to substitute the traditional Taylor series accumulation method to simplify the computation complexity. After speech enhancement, spectrum smoothing is implemented in both time and frequency index with geometric sequence weights to further compensate the spectral components distorted by noise over-reduction. Experiments on Chinese isolated phrase recognition show that the proposed front-end significantly improves the recognition robustness in car environments while the computational load is extremely reduced. Compared with the ETSI advanced front-end, the average error reduction rate (ERR) of 12.2% and 4.5% is obtained in artificial car noisy speech and real in-car speech, respectively.
File Size	5005911
File Format	PDF
ISBN	1424407273
ISSN	15206149
DOI	10.1109/ICASSP.2007.367252
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	2007-04-15
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Noise robustness Computational efficiency Speech recognition Working environment noise Estimation error Background noise Piecewise linear techniques Taylor series Speech enhancement Smoothing methods robustness speech recognition acoustic noise speech enhancement
Content Type	Text
Resource Type	Article

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7	National Digital Library of India Club (NDLI Club)	Queries related to NDLI Club formation, support, user awareness program, seminar/symposium, collaboration, social media, promotion, and outreach	clubsupport@ndl.gov.in
8	Digital Preservation Centre (DPC)	Assistance with digitizing and archiving copyright-free printed books	dpc@ndl.gov.in
9	IDR Setup or Support	Queries related to establishment and support of Institutional Digital Repository (IDR) and IDR workshops	idr@ndl.gov.in