Springer Topics in Signal Processing
|
|
- Merryl Eaton
- 6 years ago
- Views:
Transcription
1 Springer Topics in Signal Processing Volume 3 Series Editors J. Benesty, Montreal, Québec, Canada W. Kellermann, Erlangen, Germany
2 Springer Topics in Signal Processing Edited by J. Benesty and W. Kellermann Vol. 1: Benesty, J.; Chen, J.; Huang, Y. Microphone Array Signal Processing 250 p [ ] Vol. 2: Benesty, J.; Chen, J.; Huang, Y.; Cohen, I. Noise Reduction in Speech Processing 240 p [ ] Vol. 3: Cohen, I.; Benesty, J.; Gannot, S. (Eds.) Speech Processing in Modern Communication 360 p [ ]
3 Israel Cohen Jacob Benesty Sharon Gannot (Eds.) Speech Processing in Modern Communication Challenges and Perspectives ABC
4 Prof. Israel Cohen Technion - Israel Institute of Technology Dept. Electrical Engineering Haifa Technion City Israel icohen@ee.technion.ac.il Dr. Sharon Gannot Bar-Ilan University School of Engineering Ramat-Gan Bdg Israel gannot@eng.biu.ac.il Prof. Dr. Jacob Benesty Université de Quebec Inst. National de la Recherche Scientifique (INRS) 800 de la Gauchetiere Ouest Montreal QC H5A 1K6 Canada benesty@emt.inrs.ca ISBN e-isbn DOI / Springer Topics in Signal Processing ISSN Library of Congress Control Number: c 2010 Springer-Verlag Berlin Heidelberg e-issn This work is subject to copyright. All rights are reserved, whether the whole or part of the material is concerned, specifically the rights of translation, reprinting, reuse of illustrations, recitation, broadcasting, reproduction on microfilm or in any other way, and storage in data banks. Duplication of this publication or parts thereof is permitted only under the provisions of the German Copyright Law of September 9, 1965, in its current version, and permission for use must always be obtained from Springer. Violations are liable to prosecution under the German Copyright Law. The use of general descriptive names, registered names, trademarks, etc. in this publication does not imply, even in the absence of a specific statement, that such names are exempt from the relevant protective laws and regulations and therefore free for general use. Cover Design: WMXDesign GmbH, Heidelberg Printed in acid-free paper springer.com
5 Preface More and more devices for human-to-human and human-to-machine communications, where sound pickup and rendering is necessary, require some sophisticated algorithms. This is due to the fact that the acoustic environment in which we live in and communicate is extremely challenging. The difficult problems encountered in this environment are very well known and they are mainly acoustic echo cancellation, interference and noise suppression, and dereverberation. More than ever, these fundamental problems need to be tackled rigorously. This is the objective of this edited book, which contains twelve chapters that are briefly summarized below. Chapter 1 addresses the problem of linear system identification in the short-time Fourier transform (STFT) domain. Identification of linear systems is of major importance in diverse applications of signal processing, including acoustic echo cancellation, relative transfer function (RTF) identification, dereverberation, blind source separation, and beamforming in reverberant environments. In this chapter, the authors introduce three models for linear system identification and investigate the influence of model order on the estimation accuracy. The three models are based on either crossband filters between subbands, multiplicative transfer functions, or cross-multiplicative transfer functions. It is shown both analytically and experimentally that the estimation accuracy does not necessarily improve by increasing the model order. The problem of RTF identification between sensors is addressed in Chapter 2. This transfer function represents the coupling between two sensors with respect to a desired or interfering source. The authors describe an alternative representation of time domain convolution with convolutive transfer functions in the STFT domain, and show improved results compared to existing RTF identification methods. In low-cost hands-free telecommunication systems the loudspeaker signal may contain a certain level of nonlinear distortions, which necessitate nonlinear modeling of the acoustic echo path. Chapter 3 describes a novel approach for nonlinear system identification in the STFT domain. It introv
6 vi duces Volterra filters in the STFT domain and considers the identification of quadratically nonlinear systems. It shows that a significant reduction in computational cost as well as substantial improvement in estimation accuracy can be achieved over a time-domain Volterra model, particularly when long-memory systems are considered. Chapter 4 presents a family of non-parametric variable step-size (VSS) algorithms, which are particularly suitable for realistic acoustic echo cancellation (AEC) scenarios. The VSS algorithms are developed based on another objective of AEC application, i.e., to recover the near-end signal from the error signal of the adaptive filter. As a consequence, these algorithms are equipped with good robustness features against near-end signal variations, like double-talk. Speech enhancement in transient noise environments is addressed in Chapter 5. An estimate of the desired signal is obtained under signal presence uncertainty using a simultaneous detection and estimation approach. This method facilitates suppression of transient noise with a controlled level of speech distortion. Cost parameters control the tradeoff between speech distortion, caused by missed detection of speech components, and residual musical noise resulting from false-detection. Chapter 6 describes a model-based approach for combined dereverberation and denoising of speech signals. This approach is developed by using a multichannel autoregressive model of room acoustics and a time-varying power spectrum model of clean speech signals. Chapter 7 investigates separation of speech and music signals from singlesensor audio mixtures. It describes codebook approaches and a Bayesian probabilistic framework for source modeling and source estimation. The source models include Gaussian scaled mixture models, codebooks of auto regressive models, and Bayesian non negative matrix factorization (BNMF). Microphone arrays are becoming increasingly more common in the acquisition and denoising of acoustic signals. Additional microphones allow us to apply spatiotemporal filtering methods, which are significantly more powerful than conventional temporal filtering techniques. Chapter 8 is concerned with beamformer designs tailored to the specific nature of microphone array environments, i.e., broadband signals and reverberant channels. A distinction is made between wideband and narrowband metrics, and the relationships between broadband performance measures and the corresponding component narrowband measures are analyzed. Chapter 9 presents some new insights into the minimum variance distortionless response (MVDR) beamformer. It analyzes the tradeoff between dereverberation and noise reduction achieved by using the MVDR beamformer, and discusses relations between the MVDR and other optimal beamformers. Chapter 10 addresses the problem of extracting several desired speech signals from multi-microphone measurements, which are contaminated by nonstationary and stationary interfering signals. A linearly constrained minimum variance (LCMV) beamformer is designed with two sets of linear constraints:
7 one for maintaining the desired signals and one for mitigating both the stationary and non-stationary interferences. Spherical microphone arrays have been recently studied for spatial sound recording, speech communication, and sound field analysis for room acoustics and noise control. Complementary studies presented progress in beamforming methods. Chapter 11 reviews beamforming methods recently developed for spherical arrays, from the widely used delay-and-sum and Dolph-Chebyshev, to the more advanced optimal methods, typically performed in the spherical harmonics domain. Finally, Chapter 12 presents a family of broadband source localization algorithms based on parameterized spatiotemporal correlation, including the popular and robust steered response power (SRP) algorithm. It develops source localization methods based on minimum information entropy and temporally constrained minimum variance. This book has been edited for engineers, researchers, and graduate students who work on speech processing for communication applications. We hope that the readers will find many new and interesting concepts that are presented in this text useful and inspiring. We deeply appreciate the efforts, willingness, and enthusiasm of all the contributing authors. Without their commitment, this book would not have been possible. We would like to take this opportunity to thank again Christoph Baumann, Carmen Wolf, and Petra Jantzen from Springer (Germany) for their wonderful help in the preparation and publication of this manuscript. Working with them is always a pleasure and a wonderful experience. Finally, we would like to dedicate this edited book to our parents. vii Haifa/ Montreal/ Ramat-Gan Nov Israel Cohen Jacob Benesty Sharon Gannot
8 Contents 1 Linear System Identification in the Short-Time Fourier Transform Domain... 1 Yekutiel Avargel and Israel Cohen 1.1 Introduction Problem Formulation System Identification Using Crossband Filters Crossband Filters Representation Batch Estimation of Crossband Filters Selecting the Optimal Number of Crossband Filters System Identification Using the MTF Approximation The MTF Approximation Optimal Window Length The Cross-MTF Approximation Adaptive Estimation of Cross-Terms Adaptive Control Algorithm ExperimentalResults Crossband Filters Estimation Comparison of the Crossband Filters and MTF Approaches CMTF Adaptation for Acoustic Echo Cancellation Conclusions Appendix References Identification of the Relative Transfer Function between Sensors in the Short-Time Fourier Transform Domain Ronen Talmon, Israel Cohen, and Sharon Gannot 2.1 Introduction Identification of the RTF Using Multiplicative Transfer Function Approximation xi
9 x Contents Problem Formulation and the Multiplicative Transfer Function Approximation RTF Identification Using Non-Stationarity RTF Identification Using Speech Signals Identification of the RTF Using Convolutive Transfer Function Approximation The Convolutive Transfer Function Approximation RTF Identification Using the Convolutive Transfer Function Approximation Relative Transfer Function Identification in Speech Enhancement Applications Blocking Matrix The Transfer Function Generalized Sidelobe Canceler Conclusions References Representation and Identification of Nonlinear Systems in the Short-Time Fourier Transform Domain Yekutiel Avargel and Israel Cohen 3.1 Introduction Volterra System Identification Representation of Volterra Filters in the STFT Domain Second-Order Volterra Filters High-Order Volterra Filters A New STFT Model For Nonlinear Systems Quadratically Nonlinear Model High-Order Nonlinear Models Quadratically Nonlinear System Identification Batch Estimation Scheme Adaptive Estimation Scheme ExperimentalResults Performance Evaluation for White Gaussian Inputs Nonlinear Undermodeling in Adaptive System Identification Nonlinear Acoustic Echo Cancellation Application Conclusions Appendix References Variable Step-Size Adaptive Filters for Echo Cancellation 89 Constantin Paleologu, Jacob Benesty, and Silviu Ciochină 4.1 Introduction Non-Parametric VSS-NLMS Algorithm VSS-NLMS Algorithms for Echo Cancellation VSS-APA for Echo Cancellation VFF-RLS for System Identification
10 Contents xi 4.6 Simulations VSS-NLMS Algorithms for AEC VSS-APA for AEC VFF-RLS for System Identification Conclusions References Simultaneous Detection and Estimation Approach for Speech Enhancement and Interference Suppression Ari Abramson and Israel Cohen 5.1 Introduction Classical Speech Enhancement in Nonstationary Noise Environments Simultaneous Detection and Estimation for Speech Enhancement Quadratic Distortion Measure Quadratic Spectral Amplitude Distortion Measure Spectral Estimation Under a Transient Noise Indication A Priori SNR Estimation ExperimentalResults Simultaneous Detection and Estimation Spectral Estimation Under a Transient Noise Indication Conclusions References Speech Dereverberation and Denoising Based on Time Varying Speech Model and Autoregressive Reverberation Model Takuya Yoshioka, Tomohiro Nakatani, Keisuke Kinoshita, and Masato Miyoshi 6.1 Introduction Goal Technological Background Minimum Mean-Squared Error Signal Estimation andmodel-basedapproach Dereverberation Method Heuristic Derivation of Weighted Prediction Error Method Reverberation Model Clean Speech Model Clean Speech Signal Estimator and Parameter Optimization Combined Dereverberation and Denoising Method Room Acoustics Model Clean Speech Model
11 xii Contents Clean Speech Signal Estimator Parameter Optimization Experiments Conclusions References Codebook Approaches for Single Sensor Speech/Music Separation Raphaël Blouet and Israel Cohen 7.1 Introduction Single Sensor Source Separation Problem Formulation GSMM-Based Source Separation AR-Based Source Separation Bayesian Non-Negative Matrix Factorization Learning the Codebook Multi-Window Source Separation General Description of the Algorithm Choice of a Confidence Measure Practical Choice of the Thresholds Estimation of the Expansion Coefficients Median Filter Smoothing Prior GMM Modeling of the Amplitude Coefficients ExperimentalStudy Evaluation Criteria Experimental Setup and Results Conclusions References Microphone Arrays: Fundamental Concepts Jacek P. Dmochowski and Jacob Benesty 8.1 Introduction SignalModel ArrayModel Signal-to-Noise Ratio ArrayGain Noise Rejection and Desired Signal Cancellation Beampattern Anechoic Plane Wave Model Directivity Superdirective Beamforming WhiteNoiseGain Spatial Aliasing Monochromatic Signal Broadband Signal
12 Contents xiii 8.11 Mean-Squared Error Wiener Filter Minimum Variance Distortionless Response Conclusions References The MVDR Beamformer for Speech Enhancement Emanuël A. P. Habets, Jacob Benesty, Sharon Gannot, and Israel Cohen 9.1 Introduction Problem Formulation From Speech Distortion Weighted Multichannel Wiener Filter to Minimum Variance Distortionless Response Filter Speech Distortion Weighted Multichannel Wiener Filter Minimum Variance Distortionless Response Filter Decomposition of the Speech Distortion Weighted Multichannel Wiener Filter Equivalence of MVDR and Maximum SNR Beamformer PerformanceMeasures PerformanceAnalysis On the Comparison of Different MVDR Beamformers Local Analyzes Global Analyzes Non-Coherent Noise Field Coherent plus Non-Coherent Noise Field Performance Evaluation Influence of the Number of Microphones Influence of the Reverberation Time Influence of the Noise Field Example Using Speech Signals Conclusions Appendix References Extraction of Desired Speech Signals in Multiple-Speaker Reverberant Noisy Environments Shmulik Markovich, Sharon Gannot, and Israel Cohen 10.1 Introduction Problem Formulation Proposed Method The LCMV and MVDR Beamformers The Constraints Set Equivalent Constraints Set Modified Constraints Set
13 xiv Contents 10.4 Estimation of the Constraints Matrix Interferences Subspace Estimation Desired Sources RTF Estimation Algorithm Summary Experimental Study The Test Scenario Simulated Environment Real Environment Conclusions References Spherical Microphone Array Beamforming Boaz Rafaely, Yotam Peled, Morag Agmon, Dima Khaykin, and Etan Fisher 11.1 Introduction Spherical Array Processing Regular Beam Pattern Delay-and-Sum Beam Pattern Dolph-Chebyshev Beam Pattern Optimal Beamforming Beam Pattern with Desired Multiple Nulls D Beam Pattern and its Steering Near-Field Beamforming Direction-of-Arrival Estimation Conclusions References Steered Beamforming Approaches for Acoustic Source Localization Jacek P. Dmochowski and Jacob Benesty 12.1 Introduction Signal Model Spatial and Spatiotemporal Filtering Parameterized Spatial Correlation Matrix (PSCM) Source Localization Using Parameterized Spatial Correlation Steered Response Power Minimum Variance Distortionless Response Maximum Eigenvalue Broadband MUSIC Minimum Entropy Sparse Representation of the PSCM Linearly Constrained Minimum Variance Autoregressive Modeling Challenges Conclusions References
14 Contents xv Index
15 List of Contributors Ari Abramson Technion Israel Institute of Technology, Israel Morag Agmon Ben-Gurion University of the Negev, Israel Yekutiel Avargel Technion Israel Institute of Technology, Israel Jacob Benesty INRS-EMT, QC, Canada Raphaël Blouet Audionamix, France Silviu Ciochină University Politehnica of Bucharest, Romania Israel Cohen Technion Israel Institute of Technology, Israel Jacek P. Dmochowski City College of New York, NY, USA xvii
16 xviii List of Contributors Etan Fisher Ben-Gurion University of the Negev, Israel Sharon Gannot Bar-Ilan University, Israel Emanuël A. P. Habets Imperial College, UK Dima Khaykin Ben-Gurion University of the Negev, Israel Keisuke Kinoshita NTT Communication Science Laboratories, Japan Shmulik Markovich Bar-Ilan University, Israel Masato Miyoshi NTT Communication Science Laboratories, Japan Tomohiro Nakatani NTT Communication Science Laboratories, Japan Constantin Paleologu University Politehnica of Bucharest, Romania Yotam Peled Ben-Gurion University of the Negev, Israel Boaz Rafaely Ben-Gurion University of the Negev, Israel Ronen Talmon Technion Israel Institute of Technology, Israel Takuya Yoshioka NTT Communication Science Laboratories, Japan
Emanuël A. P. Habets, Jacob Benesty, and Patrick A. Naylor. Presented by Amir Kiperwas
Emanuël A. P. Habets, Jacob Benesty, and Patrick A. Naylor Presented by Amir Kiperwas 1 M-element microphone array One desired source One undesired source Ambient noise field Signals: Broadband Mutually
More informationRecent Advances in Acoustic Signal Extraction and Dereverberation
Recent Advances in Acoustic Signal Extraction and Dereverberation Emanuël Habets Erlangen Colloquium 2016 Scenario Spatial Filtering Estimated Desired Signal Undesired sound components: Sensor noise Competing
More information546 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 17, NO. 4, MAY /$ IEEE
546 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL 17, NO 4, MAY 2009 Relative Transfer Function Identification Using Convolutive Transfer Function Approximation Ronen Talmon, Israel
More information/$ IEEE
IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 17, NO. 6, AUGUST 2009 1071 Multichannel Eigenspace Beamforming in a Reverberant Noisy Environment With Multiple Interfering Speech Signals
More informationMichael Brandstein Darren Ward (Eds.) Microphone Arrays. Signal Processing Techniques and Applications. With 149 Figures. Springer
Michael Brandstein Darren Ward (Eds.) Microphone Arrays Signal Processing Techniques and Applications With 149 Figures Springer Contents Part I. Speech Enhancement 1 Constant Directivity Beamforming Darren
More informationDual Transfer Function GSC and Application to Joint Noise Reduction and Acoustic Echo Cancellation
Dual Transfer Function GSC and Application to Joint Noise Reduction and Acoustic Echo Cancellation Gal Reuven Under supervision of Sharon Gannot 1 and Israel Cohen 2 1 School of Engineering, Bar-Ilan University,
More informationDISTANT or hands-free audio acquisition is required in
158 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 18, NO. 1, JANUARY 2010 New Insights Into the MVDR Beamformer in Room Acoustics E. A. P. Habets, Member, IEEE, J. Benesty, Senior Member,
More informationA BROADBAND BEAMFORMER USING CONTROLLABLE CONSTRAINTS AND MINIMUM VARIANCE
A BROADBAND BEAMFORMER USING CONTROLLABLE CONSTRAINTS AND MINIMUM VARIANCE Sam Karimian-Azari, Jacob Benesty,, Jesper Rindom Jensen, and Mads Græsbøll Christensen Audio Analysis Lab, AD:MT, Aalborg University,
More informationSpeech Enhancement Using Beamforming Dr. G. Ramesh Babu 1, D. Lavanya 2, B. Yamuna 2, H. Divya 2, B. Shiva Kumar 2, B.
www.ijecs.in International Journal Of Engineering And Computer Science ISSN:2319-7242 Volume 4 Issue 4 April 2015, Page No. 11143-11147 Speech Enhancement Using Beamforming Dr. G. Ramesh Babu 1, D. Lavanya
More informationIEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 21, NO. 5, MAY
IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 21, NO. 5, MAY 2013 945 A Two-Stage Beamforming Approach for Noise Reduction Dereverberation Emanuël A. P. Habets, Senior Member, IEEE,
More informationROBUST SUPERDIRECTIVE BEAMFORMER WITH OPTIMAL REGULARIZATION
ROBUST SUPERDIRECTIVE BEAMFORMER WITH OPTIMAL REGULARIZATION Aviva Atkins, Yuval Ben-Hur, Israel Cohen Department of Electrical Engineering Technion - Israel Institute of Technology Technion City, Haifa
More informationOn Regularization in Adaptive Filtering Jacob Benesty, Constantin Paleologu, Member, IEEE, and Silviu Ciochină, Member, IEEE
1734 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 6, AUGUST 2011 On Regularization in Adaptive Filtering Jacob Benesty, Constantin Paleologu, Member, IEEE, and Silviu Ciochină,
More information260 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 18, NO. 2, FEBRUARY /$ IEEE
260 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 18, NO. 2, FEBRUARY 2010 On Optimal Frequency-Domain Multichannel Linear Filtering for Noise Reduction Mehrez Souden, Student Member,
More informationJoint dereverberation and residual echo suppression of speech signals in noisy environments Habets, E.A.P.; Gannot, S.; Cohen, I.; Sommen, P.C.W.
Joint dereverberation and residual echo suppression of speech signals in noisy environments Habets, E.A.P.; Gannot, S.; Cohen, I.; Sommen, P.C.W. Published in: IEEE Transactions on Audio, Speech, and Language
More informationIN REVERBERANT and noisy environments, multi-channel
684 IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL. 11, NO. 6, NOVEMBER 2003 Analysis of Two-Channel Generalized Sidelobe Canceller (GSC) With Post-Filtering Israel Cohen, Senior Member, IEEE Abstract
More informationMicrophone Array Design and Beamforming
Microphone Array Design and Beamforming Heinrich Löllmann Multimedia Communications and Signal Processing heinrich.loellmann@fau.de with contributions from Vladi Tourbabin and Hendrik Barfuss EUSIPCO Tutorial
More informationMATLAB Guide to Finite Elements
MATLAB Guide to Finite Elements Peter I. Kattan MATLAB Guide to Finite Elements An Interactive Approach Second Edition With 108 Figures and 25 Tables Peter I. Kattan, PhD P.O. BOX 1392 Amman 11118 Jordan
More informationReal-time Adaptive Concepts in Acoustics
Real-time Adaptive Concepts in Acoustics Real-time Adaptive Concepts in Acoustics Blind Signal Separation and Multichannel Echo Cancellation by Daniel W.E. Schobben, Ph. D. Philips Research Laboratories
More informationHUMAN speech is frequently encountered in several
1948 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 20, NO. 7, SEPTEMBER 2012 Enhancement of Single-Channel Periodic Signals in the Time-Domain Jesper Rindom Jensen, Student Member,
More informationAdvances in Direction-of-Arrival Estimation
Advances in Direction-of-Arrival Estimation Sathish Chandran Editor ARTECH HOUSE BOSTON LONDON artechhouse.com Contents Preface xvii Acknowledgments xix Overview CHAPTER 1 Antenna Arrays for Direction-of-Arrival
More informationSpeech and Audio Processing Recognition and Audio Effects Part 3: Beamforming
Speech and Audio Processing Recognition and Audio Effects Part 3: Beamforming Gerhard Schmidt Christian-Albrechts-Universität zu Kiel Faculty of Engineering Electrical Engineering and Information Engineering
More informationarxiv: v1 [cs.sd] 4 Dec 2018
LOCALIZATION AND TRACKING OF AN ACOUSTIC SOURCE USING A DIAGONAL UNLOADING BEAMFORMING AND A KALMAN FILTER Daniele Salvati, Carlo Drioli, Gian Luca Foresti Department of Mathematics, Computer Science and
More informationInformed Spatial Filtering for Sound Extraction Using Distributed Microphone Arrays
IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 22, NO. 7, JULY 2014 1195 Informed Spatial Filtering for Sound Extraction Using Distributed Microphone Arrays Maja Taseska, Student
More informationANALOG CIRCUITS AND SIGNAL PROCESSING
ANALOG CIRCUITS AND SIGNAL PROCESSING Series Editors Mohammed Ismail, The Ohio State University Mohamad Sawan, École Polytechnique de Montréal For further volumes: http://www.springer.com/series/7381 Yongjian
More informationAdvanced Signal Processing and Digital Noise Reduction
Advanced Signal Processing and Digital Noise Reduction Advanced Signal Processing and Digital Noise Reduction Saeed V. Vaseghi Queen's University of Belfast UK ~ W I lilteubner L E Y A Partnership between
More informationStudy of the General Kalman Filter for Echo Cancellation
IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 21, NO. 8, AUGUST 2013 1539 Study of the General Kalman Filter for Echo Cancellation Constantin Paleologu, Member, IEEE, Jacob Benesty,
More informationMULTICHANNEL systems are often used for
IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 52, NO. 5, MAY 2004 1149 Multichannel Post-Filtering in Nonstationary Noise Environments Israel Cohen, Senior Member, IEEE Abstract In this paper, we present
More informationCognitive Systems Monographs
Cognitive Systems Monographs Volume 9 Editors: Rüdiger Dillmann Yoshihiko Nakamura Stefan Schaal David Vernon Heiko Hamann Space-Time Continuous Models of Swarm Robotic Systems Supporting Global-to-Local
More informationStudents: Avihay Barazany Royi Levy Supervisor: Kuti Avargel In Association with: Zoran, Haifa
Students: Avihay Barazany Royi Levy Supervisor: Kuti Avargel In Association with: Zoran, Haifa Spring 2008 Introduction Problem Formulation Possible Solutions Proposed Algorithm Experimental Results Conclusions
More informationAdvances in Computer Vision and Pattern Recognition
Advances in Computer Vision and Pattern Recognition For further volumes: http://www.springer.com/series/4205 Marco Alexander Treiber Optimization for Computer Vision An Introduction to Core Concepts and
More informationDesign for Innovative Value Towards a Sustainable Society
Design for Innovative Value Towards a Sustainable Society Mitsutaka Matsumoto Yasushi Umeda Keijiro Masui Shinichi Fukushige Editors Design for Innovative Value Towards a Sustainable Society Proceedings
More informationMULTICHANNEL ACOUSTIC ECHO SUPPRESSION
MULTICHANNEL ACOUSTIC ECHO SUPPRESSION Karim Helwani 1, Herbert Buchner 2, Jacob Benesty 3, and Jingdong Chen 4 1 Quality and Usability Lab, Telekom Innovation Laboratories, 2 Machine Learning Group 1,2
More informationArchitecture Design and Validation Methods
Architecture Design and Validation Methods Springer-Verlag Berlin Heidelberg GmbH Egon Börger (Ed.) Architecture Design and Validation Methods With 175 Figures, Springer Editor Prof. Dr. Egon Börger Universita
More informationDual-Microphone Speech Dereverberation in a Noisy Environment
Dual-Microphone Speech Dereverberation in a Noisy Environment Emanuël A. P. Habets Dept. of Electrical Engineering Technische Universiteit Eindhoven Eindhoven, The Netherlands Email: e.a.p.habets@tue.nl
More informationModeling Manufacturing Systems. From Aggregate Planning to Real-Time Control
Modeling Manufacturing Systems From Aggregate Planning to Real-Time Control Springer-Verlag Berlin Heidelberg GmbH Paolo Brandimarte. Agostino Villa (Eds.) Modeling Manufacturing Systems From Aggregate
More informationDesign of Robust Differential Microphone Arrays
IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 22, NO. 10, OCTOBER 2014 1455 Design of Robust Differential Microphone Arrays Liheng Zhao, Jacob Benesty, Jingdong Chen, Senior Member,
More informationClustered Multi-channel Dereverberation for Ad-hoc Microphone Arrays
Clustered Multi-channel Dereverberation for Ad-hoc Microphone Arrays Shahab Pasha and Christian Ritz School of Electrical, Computer and Telecommunications Engineering, University of Wollongong, Wollongong,
More informationApplied Technology and Innovation Management
Applied Technology and Innovation Management Heinrich Arnold Michael Erner Peter Möckel Christopher Schläffer Editors Applied Technology and Innovation Management Insights and Experiences from an Industry-Leading
More informationEl-Kébir Boukas and Fouad M. AL-Sunni. Mechatronic Systems. Analysis, Design and Implementation ABC
Mechatronic Systems El-Kébir Boukas and Fouad M. AL-Sunni Mechatronic Systems Analysis, Design and Implementation ABC Authors Prof. El-Kébir Boukas Mechanical Engineering Department Ecole Polytechnique
More informationSpeech Enhancement Using Microphone Arrays
Friedrich-Alexander-Universität Erlangen-Nürnberg Lab Course Speech Enhancement Using Microphone Arrays International Audio Laboratories Erlangen Prof. Dr. ir. Emanuël A. P. Habets Friedrich-Alexander
More informationFuture-Oriented Technology Analysis
Future-Oriented Technology Analysis Cristiano Cagnin Michael Keenan Ron Johnston Fabiana Scapolo Rémi Barré Editors Future-Oriented Technology Analysis Strategic Intelligence for an Innovative Economy
More informationDigital Signal Processing
Digital Signal Processing Fourth Edition John G. Proakis Department of Electrical and Computer Engineering Northeastern University Boston, Massachusetts Dimitris G. Manolakis MIT Lincoln Laboratory Lexington,
More informationAn analysis of blind signal separation for real time application
University of Wollongong Research Online University of Wollongong Thesis Collection 1954-2016 University of Wollongong Thesis Collections 2006 An analysis of blind signal separation for real time application
More informationHealth Information Technology Standards. Series Editor: Tim Benson
Health Information Technology Standards Series Editor: Tim Benson Tim Benson Principles of Health Interoperability HL7 and SNOMED Second Edition Tim Benson Abies Ltd Hermitage, Thatcham Berkshire UK ISBN
More informationAiro Interantional Research Journal September, 2013 Volume II, ISSN:
Airo Interantional Research Journal September, 2013 Volume II, ISSN: 2320-3714 Name of author- Navin Kumar Research scholar Department of Electronics BR Ambedkar Bihar University Muzaffarpur ABSTRACT Direction
More informationICT for the Next Five Billion People
ICT for the Next Five Billion People Arnold Picot Josef Lorenz Editors ICT for the Next F Five Billion People Information and Communication for Sustainable Development Editors Prof. Dr. Dr. Arnold Picot
More informationApplication of Evolutionary Algorithms for Multi-objective Optimization in VLSI and Embedded Systems
Application of Evolutionary Algorithms for Multi-objective Optimization in VLSI and Embedded Systems M.C. Bhuvaneswari Editor Application of Evolutionary Algorithms for Multi-objective Optimization in
More informationDigital Image Processing
Digital Image Processing D. Sundararajan Digital Image Processing A Signal Processing and Algorithmic Approach 123 D. Sundararajan Formerly at Concordia University Montreal Canada Additional material to
More informationAdvances in Multirate Systems
Advances in Multirate Systems Editor Advances in Multirate Systems Editor Department of Electronics Institute National INAOE Tonantzintla, Puebla Mexico ISBN 978-3-319-59273-2 ISBN 978-3-319-59274-9 (ebook)
More informationSpringer Series on. Signals and Communication Technology
Springer Series on Signals and Communication Technology Signals and Communication Technology Functional Structures in Networks AMLn A Language for Model Driven Development of Telecom Systems T. Muth ISBN
More informationSpectral estimation using higher-lag autocorrelation coefficients with applications to speech recognition
Spectral estimation using higher-lag autocorrelation coefficients with applications to speech recognition Author Shannon, Ben, Paliwal, Kuldip Published 25 Conference Title The 8th International Symposium
More informationRisk-Based Ship Design
Risk-Based Ship Design Apostolos Papanikolaou (Ed.) Risk-Based Ship Design Methods, Tools and Applications Authored by Carlos Guedes Soares, Andrzej Jasionowski, Jørgen Jensen, Dag McGeorge, Apostolos
More informationIntroduction to distributed speech enhancement algorithms for ad hoc microphone arrays and wireless acoustic sensor networks
Introduction to distributed speech enhancement algorithms for ad hoc microphone arrays and wireless acoustic sensor networks Part I: Array Processing in Acoustic Environments Sharon Gannot 1 and Alexander
More informationFoundations in Signal Processing, Communications and Networking
Foundations in Signal Processing, Communications and Networking Series Editors: W. Utschick, H. Boche, R. Mathar For other titles published in this series, go to www.springer.com/series/7603 Meik Dörpinghaus
More informationTowards an intelligent binaural spee enhancement system by integrating me signal extraction. Author(s)Chau, Duc Thanh; Li, Junfeng; Akagi,
JAIST Reposi https://dspace.j Title Towards an intelligent binaural spee enhancement system by integrating me signal extraction Author(s)Chau, Duc Thanh; Li, Junfeng; Akagi, Citation 2011 International
More informationAcoustic Emission Testing
Acoustic Emission Testing Christian U. Grosse (Eds.) Acoustic Emission Testing 123 Christian U. Grosse Department of Non-destructive Testing and Monitoring Techniques Material Testing Institute MPA University
More informationSpeech Enhancement for Nonstationary Noise Environments
Signal & Image Processing : An International Journal (SIPIJ) Vol., No.4, December Speech Enhancement for Nonstationary Noise Environments Sandhya Hawaldar and Manasi Dixit Department of Electronics, KIT
More informationStudy Of Sound Source Localization Using Music Method In Real Acoustic Environment
International Journal of Electronics Engineering Research. ISSN 975-645 Volume 9, Number 4 (27) pp. 545-556 Research India Publications http://www.ripublication.com Study Of Sound Source Localization Using
More informationLecture Notes in Applied and Computational Mechanics
Lecture Notes in Applied and Computational Mechanics Volume 28 Series Editors Prof. Dr.-Ing. Friedrich Pfeiffer Prof. Dr.-Ing. Peter Wriggers Lecture Notes in Applied and Computational Mechanics Edited
More informationSpeech Enhancement Using Robust Generalized Sidelobe Canceller with Multi-Channel Post-Filtering in Adverse Environments
Chinese Journal of Electronics Vol.21, No.1, Jan. 2012 Speech Enhancement Using Robust Generalized Sidelobe Canceller with Multi-Channel Post-Filtering in Adverse Environments LI Kai, FU Qiang and YAN
More informationSpeech Enhancement Techniques using Wiener Filter and Subspace Filter
IJSTE - International Journal of Science Technology & Engineering Volume 3 Issue 05 November 2016 ISSN (online): 2349-784X Speech Enhancement Techniques using Wiener Filter and Subspace Filter Ankeeta
More informationK-Best Decoders for 5G+ Wireless Communication
K-Best Decoders for 5G+ Wireless Communication Mehnaz Rahman Gwan S. Choi K-Best Decoders for 5G+ Wireless Communication Mehnaz Rahman Department of Electrical and Computer Engineering Texas A&M University
More informationACOUSTIC SIGNAL PROCESSING FOR TELECOMMUNICATION
ACOUSTIC SIGNAL PROCESSING FOR TELECOMMUNICATION THE KLUWER INTERNATIONAL SERIES IN ENGINEERING AND COMPUTER SCIENCE ACOUSTIC SIGNAL PROCESSING FOR TELECOMMUNICATION Edited by STEVEN L. GAY Bell Laboratories,
More informationBEAMFORMING WITHIN THE MODAL SOUND FIELD OF A VEHICLE INTERIOR
BeBeC-2016-S9 BEAMFORMING WITHIN THE MODAL SOUND FIELD OF A VEHICLE INTERIOR Clemens Nau Daimler AG Béla-Barényi-Straße 1, 71063 Sindelfingen, Germany ABSTRACT Physically the conventional beamforming method
More informationOnline Version Only. Book made by this file is ILLEGAL. 2. Mathematical Description
Vol.9, No.9, (216), pp.317-324 http://dx.doi.org/1.14257/ijsip.216.9.9.29 Speech Enhancement Using Iterative Kalman Filter with Time and Frequency Mask in Different Noisy Environment G. Manmadha Rao 1
More informationAuditory System For a Mobile Robot
Auditory System For a Mobile Robot PhD Thesis Jean-Marc Valin Department of Electrical Engineering and Computer Engineering Université de Sherbrooke, Québec, Canada Jean-Marc.Valin@USherbrooke.ca Motivations
More informationNOISE reduction, sometimes also referred to as speech enhancement,
2034 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 22, NO. 12, DECEMBER 2014 A Family of Maximum SNR Filters for Noise Reduction Gongping Huang, Student Member, IEEE, Jacob Benesty,
More informationA MULTI-CHANNEL POSTFILTER BASED ON THE DIFFUSE NOISE SOUND FIELD. Lukas Pfeifenberger 1 and Franz Pernkopf 1
A MULTI-CHANNEL POSTFILTER BASED ON THE DIFFUSE NOISE SOUND FIELD Lukas Pfeifenberger 1 and Franz Pernkopf 1 1 Signal Processing and Speech Communication Laboratory Graz University of Technology, Graz,
More informationAdaptive Wireless. Communications. gl CAMBRIDGE UNIVERSITY PRESS. MIMO Channels and Networks SIDDHARTAN GOVJNDASAMY DANIEL W.
Adaptive Wireless Communications MIMO Channels and Networks DANIEL W. BLISS Arizona State University SIDDHARTAN GOVJNDASAMY Franklin W. Olin College of Engineering, Massachusetts gl CAMBRIDGE UNIVERSITY
More informationA Comparison of the Convolutive Model and Real Recording for Using in Acoustic Echo Cancellation
A Comparison of the Convolutive Model and Real Recording for Using in Acoustic Echo Cancellation SEPTIMIU MISCHIE Faculty of Electronics and Telecommunications Politehnica University of Timisoara Vasile
More informationIntroduction to Computational Optimization Models for Production Planning in a Supply Chain
Introduction to Computational Optimization Models for Production Planning in a Supply Chain Stefan Voß David L.Woodruff Introduction to Computational Optimization Models for Production Planning in a Supply
More informationLecture Notes in Artificial Intelligence. Lecture Notes in Computer Science
Lecture Notes in Artificial Intelligence 897 Subseries of Lecture Notes in Computer Science Edited by J. G. Carbonell and J. Siekmann Lecture Notes in Computer Science Edited by G. Goos, J. Hartmanis and
More informationSpeech and Audio Processing for Coding, Enhancement and Recognition
Speech and Audio Processing for Coding, Enhancement and Recognition Tokunbo Ogunfunmi Roberto Togneri Madihally (Sim) Narasimha Editors Speech and Audio Processing for Coding, Enhancement and Recognition
More informationAdaptive Antenna Array Processing for GPS Receivers
Adaptive Antenna Array Processing for GPS Receivers By Yaohua Zheng Thesis submitted for the degree of Master of Engineering Science School of Electrical & Electronic Engineering Faculty of Engineering,
More informationMULTICHANNEL AUDIO DATABASE IN VARIOUS ACOUSTIC ENVIRONMENTS
MULTICHANNEL AUDIO DATABASE IN VARIOUS ACOUSTIC ENVIRONMENTS Elior Hadad 1, Florian Heese, Peter Vary, and Sharon Gannot 1 1 Faculty of Engineering, Bar-Ilan University, Ramat-Gan, Israel Institute of
More informationSpeech Enhancement Based On Spectral Subtraction For Speech Recognition System With Dpcm
International OPEN ACCESS Journal Of Modern Engineering Research (IJMER) Speech Enhancement Based On Spectral Subtraction For Speech Recognition System With Dpcm A.T. Rajamanickam, N.P.Subiramaniyam, A.Balamurugan*,
More informationSource Separation and Echo Cancellation Using Independent Component Analysis and DWT
Source Separation and Echo Cancellation Using Independent Component Analysis and DWT Shweta Yadav 1, Meena Chavan 2 PG Student [VLSI], Dept. of Electronics, BVDUCOEP Pune,India 1 Assistant Professor, Dept.
More informationComparison of LMS Adaptive Beamforming Techniques in Microphone Arrays
SERBIAN JOURNAL OF ELECTRICAL ENGINEERING Vol. 12, No. 1, February 2015, 1-16 UDC: 621.395.61/.616:621.3.072.9 DOI: 10.2298/SJEE1501001B Comparison of LMS Adaptive Beamforming Techniques in Microphone
More informationAbout Multichannel Speech Signal Extraction and Separation Techniques
Journal of Signal and Information Processing, 2012, *, **-** doi:10.4236/jsip.2012.***** Published Online *** 2012 (http://www.scirp.org/journal/jsip) About Multichannel Speech Signal Extraction and Separation
More informationThe Hybrid Simplified Kalman Filter for Adaptive Feedback Cancellation
The Hybrid Simplified Kalman Filter for Adaptive Feedback Cancellation Felix Albu Department of ETEE Valahia University of Targoviste Targoviste, Romania felix.albu@valahia.ro Linh T.T. Tran, Sven Nordholm
More informationComparison of LMS and NLMS algorithm with the using of 4 Linear Microphone Array for Speech Enhancement
Comparison of LMS and NLMS algorithm with the using of 4 Linear Microphone Array for Speech Enhancement Mamun Ahmed, Nasimul Hyder Maruf Bhuyan Abstract In this paper, we have presented the design, implementation
More informationRequirements Engineering for Digital Health
Requirements Engineering for Digital Health Samuel A. Fricker Christoph Thümmler Anastasius Gavras Editors Requirements Engineering for Digital Health Editors Samuel A. Fricker Blekinge Institute of Technology
More informationEnhancement of Speech Signal Based on Improved Minima Controlled Recursive Averaging and Independent Component Analysis
Enhancement of Speech Signal Based on Improved Minima Controlled Recursive Averaging and Independent Component Analysis Mohini Avatade & S.L. Sahare Electronics & Telecommunication Department, Cummins
More informationComputational Intelligence for Network Structure Analytics
Computational Intelligence for Network Structure Analytics Maoguo Gong Qing Cai Lijia Ma Shanfeng Wang Yu Lei Computational Intelligence for Network Structure Analytics 123 Maoguo Gong Xidian University
More informationBinaural Beamforming with Spatial Cues Preservation
Binaural Beamforming with Spatial Cues Preservation By Hala As ad Thesis submitted to the Faculty of Graduate and Postdoctoral Studies in partial fulfillment of the requirements for the degree of Master
More informationZEW Economic Studies. Publication Series of the Centre for European Economic Research (ZEW), Mannheim, Germany
ZEW Economic Studies Publication Series of the Centre for European Economic Research (ZEW), Mannheim, Germany ZEW Economic Studies Vol. 1: O. Hohmeyer, K. Rennings (Eds.) Man-Made Climate Change Economic
More informationSingle channel noise reduction
Single channel noise reduction Basics and processing used for ETSI STF 94 ETSI Workshop on Speech and Noise in Wideband Communication Claude Marro France Telecom ETSI 007. All rights reserved Outline Scope
More informationPrinciples of Space- Time Adaptive Processing 3rd Edition. By Richard Klemm. The Institution of Engineering and Technology
Principles of Space- Time Adaptive Processing 3rd Edition By Richard Klemm The Institution of Engineering and Technology Contents Biography Preface to the first edition Preface to the second edition Preface
More informationMultiple Antenna Processing for WiMAX
Multiple Antenna Processing for WiMAX Overview Wireless operators face a myriad of obstacles, but fundamental to the performance of any system are the propagation characteristics that restrict delivery
More informationSpringerBriefs in Computer Science
SpringerBriefs in Computer Science Series Editors Stan Zdonik Shashi Shekhar Jonathan Katz Xindong Wu Lakhmi C. Jain David Padua Xuemin (Sherman) Shen Borko Furht V.S. Subrahmanian Martial Hebert Katsushi
More informationScientific Data Mining and Knowledge Discovery
Scientific Data Mining and Knowledge Discovery Mohamed Medhat Gaber Editor Scientific Data Mining and Knowledge Discovery Principles and Foundations ABC Editor Mohamed Medhat Gaber Caulfield School of
More informationAdaptive f-xy Hankel matrix rank reduction filter to attenuate coherent noise Nirupama (Pam) Nagarajappa*, CGGVeritas
Adaptive f-xy Hankel matrix rank reduction filter to attenuate coherent noise Nirupama (Pam) Nagarajappa*, CGGVeritas Summary The reliability of seismic attribute estimation depends on reliable signal.
More informationTECHNOLOGY, INNOVATION, and POLICY 3. Series of the Fraunhofer Institute for Systems and Innovation Research (lsi)
TECHNOLOGY, INNOVATION, and POLICY 3 Series of the Fraunhofer Institute for Systems and Innovation Research (lsi) Guido Reger Ulrich Schmoch (Eds.) Organisation of Science and Technology at the Watershed
More informationAdaptive selective sidelobe canceller beamformer with applications in radio astronomy
Adaptive selective sidelobe canceller beamformer with applications in radio astronomy Ronny Levanda and Amir Leshem 1 Abstract arxiv:1008.5066v1 [astro-ph.im] 30 Aug 2010 We propose a new algorithm, for
More informationGROUP SPARSITY FOR MIMO SPEECH DEREVERBERATION. and the Cluster of Excellence Hearing4All, Oldenburg, Germany.
0 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics October 8-, 0, New Paltz, NY GROUP SPARSITY FOR MIMO SPEECH DEREVERBERATION Ante Jukić, Toon van Waterschoot, Timo Gerkmann,
More informationCalibration of Microphone Arrays for Improved Speech Recognition
MITSUBISHI ELECTRIC RESEARCH LABORATORIES http://www.merl.com Calibration of Microphone Arrays for Improved Speech Recognition Michael L. Seltzer, Bhiksha Raj TR-2001-43 December 2001 Abstract We present
More informationDesign and Implementation on a Sub-band based Acoustic Echo Cancellation Approach
Vol., No. 6, 0 Design and Implementation on a Sub-band based Acoustic Echo Cancellation Approach Zhixin Chen ILX Lightwave Corporation Bozeman, Montana, USA chen.zhixin.mt@gmail.com Abstract This paper
More informationData Assimilation: Tools for Modelling the Ocean in a Global Change Perspective
Data Assimilation: Tools for Modelling the Ocean in a Global Change Perspective NATO ASI Series Advanced Science Institutes Series A series presenting the results of activities sponsored by the NA TO Science
More informationTechnology Roadmapping for Strategy and Innovation
Technology Roadmapping for Strategy and Innovation Martin G. Moehrle, Ralf Isenmann, and Robert Phaal (Eds.) Technology Roadmapping for Strategy and Innovation Charting the Route to Success ABC Editors
More informationDry Etching Technology for Semiconductors. Translation supervised by Kazuo Nojiri Translation by Yuki Ikezi
Dry Etching Technology for Semiconductors Translation supervised by Kazuo Nojiri Translation by Yuki Ikezi Kazuo Nojiri Dry Etching Technology for Semiconductors Kazuo Nojiri Lam Research Co., Ltd. Tokyo,
More information