Imperial College London

Dr Patrick A. Naylor

Faculty of EngineeringDepartment of Electrical and Electronic Engineering

Professor of Speech & Acoustic Signal Processing
 
 
 
//

Contact

 

+44 (0)20 7594 6235p.naylor Website

 
 
//

Location

 

803Electrical EngineeringSouth Kensington Campus

//

Summary

 

Publications

Publication Type
Year
to

328 results found

Antonacci F, Filos J, Thomas M, Habets EAP, Sarti A, Naylor PAet al., 2012, Inference of room geometry from acoustic impulse responses, IEEE Trans. Audio Speech Language Proc., Vol: 20, Pages: 2683-2695

Journal article

Annibale P, Antonacci F, Bestagini P, Brutti A, Canclini A, Cristoforetti L, Habets EAP, Filos J, Kellermann W, Kowalczyk K, Lombard A, Mabande E, Markovic D, Naylor PA, Omologo M, Rabenstein R, Sarti A, Svaizer P, Thomas MRPet al., 2011, The SCENIC Project: Space-Time Audio Processing for Environment-Aware Acoustic Sensing and Rendering

Conference paper

Slaney M, Naylor PA, 2011, Audio and Acoustic Signal Processing, IEEE SIGNAL PROCESSING MAGAZINE, Vol: 28, Pages: 160-U26, ISSN: 1053-5888

Journal article

Jarrett DP, Thomas MR, Habets EAP, Naylor PAet al., 2011, Simulating Room Impulse Responses for Spherical Microphone Arrays

Conference paper

Loganathan P, Habets EAP, Naylor PA, 2011, A Proportionate Adaptive Algorithm with Variable Partitioned Block Length for Acoustic Echo Cancellation

Conference paper

Gaubitch ND, Brookes M, Naylor PA, Sharma Det al., 2011, Bayesian Adaptive method for estimating Speech Intelligibility in noise, Pages: 169-174

We present the Bayesian Adaptive Speech Intelligibility Estimation (BASIE) method - a tool for rapid estimation of a given speech reception threshold (SRT) and the slope at that threshold of multiple psychometric functions for speech intelligibility in noise. The core of this tool is an adaptive Bayesian procedure, which adjusts the signal-to-noise ratio at each subsequent stimulus such that the expected variance of the threshold and slope estimates are minimised. Simulation results show that the algorithm is able to achieve SRT estimates accurate to within ±1 dB in under 30 iterations. Furthermore, we discuss strategies for using BASIE to evaluate the effects of speech processing algorithms on intelligibility and we give two illustrative examples for different noise reduction methods with supporting listening experiments.

Conference paper

Sharma D, Hilkhuysen G, Gaubitch ND, Brookes M, Naylor PAet al., 2011, C-Qual - A validation of PESQ using degradations encountered in forensic and law enforcement audio, Pages: 177-181

Assessment of speech quality of law-enforcement audio recordings is important as degradations introduced by non-ideal recording conditions can reduce the intelligence value of such recordings. Furthermore a model that predicts speech quality could be beneficial for assessing the performance of audio collection and enhancement systems. The Perceptual Evaluation of Speech Quality (PESQ) algorithm (ITU-T P.862) has been validated for degradations common in telecommunications. In this paper we apply PESQ to degradations typically encountered in law-enforcement. Also we present a subjectively labeled database (C-Qual) containing distortions encountered in law enforcement scenarios. Comparing the prediction by PESQ and the observed opinions provided by the listeners shows that PESQ is less suitable for estimating the speech quality in this context.

Conference paper

Loganathan P, Habets EAP, Naylor PA, 2011, A PROPORTIONATE ADAPTIVE ALGORITHM WITH VARIABLE PARTITIONED BLOCK LENGTH FOR ACOUSTIC ECHO CANCELLATION, IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), Publisher: IEEE, Pages: 73-76, ISSN: 1520-6149

Conference paper

Annibale P, Antonacci F, Bestagini P, Brutti A, Canclini A, Cristoforetti L, Habets E, Kellermann W, Kowalczyk K, Lombard A, Mabande E, Markovic D, Naylor P, Omologo M, Rabenstein R, Sarti A, Svaizer P, Thomas Met al., 2011, The SCENIC Project: Environment-aware Sound Sensing and Rendering, PROCEEDINGS OF THE 2ND EUROPEAN FUTURE TECHNOLOGIES CONFERENCE AND EXHIBITION 2011 (FET 11), Vol: 7, Pages: 150-152, ISSN: 1877-0509

Journal article

Jarrett DP, Habets EAP, Thomas MRP, Gaubitch ND, Naylor PAet al., 2011, Dereverberation performance of rigid and open spherical microphone arrays: theory & simulation

Conference paper

Canclini A, Antonacci F, Thomas MRP, Filos J, Sarti A, Naylor PA, Tubaro Set al., 2011, Exact Localization of Acoustic Reflectors from Quadratic Constraints

Conference paper

Filos J, Canclini A, Thomas MRP, Antonacci F, Sarti A, Naylor PAet al., 2011, Robust Inference of Room Geometry From Acoustic Measurements Using the Hough Transform

Conference paper

Gaubitch ND, Brookes M, Naylor PA, Sharma Det al., 2011, Single-Microphone Blind Channel Identification in Speech Using Spectrum Classification

Conference paper

Gudnason J, Thomas MRP, Ellis DPW, Naylor PAet al., 2011, Data-Driven Voice Source Waveform Analysis and Synthesis, Speech Communication, Vol: to appear

Journal article

Habets EAP, Benesty J, Naylor PA, 2011, A Cross-Relation Based Affine Projection Algorithm for Blind SIMO System Identification

Conference paper

Sharma D, Naylor PA, Gaubitch ND, Miet al., 2011, Short-Time Objective Assessment of Speech Quality

Conference paper

Thomas MRP, Gaubitch N, Naylor PA, 2011, Application of Channel Shortening to Acoustic Channel Equalization in the Presence of Noise and Estimation Error

Conference paper

Thomas MRP, Gudnason J, Naylor PA, 2011, Estimation of Glottal Closing and Opening Instants in Voiced Speech using the YAGA Algorithm, IEEE Trans. Audio Speech Language Proc., Vol: to appear

Journal article

Loganathan P, Habets EAP, Naylor PA, 2010, A Partitioned Block Proportionate Adaptive Algorithm for Acoustic Echo Cancellation

Conference paper

Rashobh R, Khong AWH, Naylor PA, 2010, Adaptive blind system identification for speech dereverberation using a priori estimates

Conference paper

Nakatani T, Kellermann W, Naylor P, Miyoshi M, Juang BHFet al., 2010, Introduction to the Special Issue on Processing Reverberant Speech: Methodologies and Applications, IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING, Vol: 18, Pages: 1673-1675, ISSN: 1558-7916

Journal article

Tsakiris MC, Lopes CG, Naylor PA, 2010, An Alternative Criterion for Regularization in Recursive Least-Squares Problems, York, UK

Conference paper

Jarrett DP, Habets EAP, Naylor PA, 2010, Source Localization in the Spherical Harmonic Domain Using a Pseudointensity Vector

Conference paper

Jarrett DP, Habets EAP, Naylor PA, 2010, 3D Source Localization in the Spherical Harmonic Domain Using a Pseudointensity Vector, Aalborg, Denmark

Conference paper

Naylor PA, Gaubitch ND, 2010, Speech Dereverberation, Publisher: Springer, ISBN: 978-1-84996-056-4

Book

Nordholm S, Abhayapala T, Doclo S, Gannot S, Naylor P, Tashev Iet al., 2010, Microphone Array Speech Processing, EURASIP JOURNAL ON ADVANCES IN SIGNAL PROCESSING, ISSN: 1687-6180

Journal article

Filos J, Habets EAP, Naylor PA, 2010, A Two-Step Approach to Blindly Infer Room Geometries

Conference paper

Naylor PA, 2010, P. A. Naylor, “Providing a plurality of audio files with consistent loudness levels but different audio characteristics,” International Patent Patent No. WO/1910/005 823, 2010. [Online]. Available: http: //www.wipo.int/pctdb/en/wo.jsp?WO=2010005823, WO/1910/005 823

Patent

Naylor PA, Evers C, Eman, 2010, Speech Dereverberation

Conference paper

Jarrett DP, Habets EAP, Naylor PA, 2010, Eigenbeam-based acoustic source tracking in noisy reverberant environments

Conference paper

This data is extracted from the Web of Science and reproduced under a licence from Thomson Reuters. You may not copy or re-distribute this data in whole or in part without the written consent of the Science business of Thomson Reuters.

Request URL: http://wlsprd.imperial.ac.uk:80/respub/WEB-INF/jsp/search-html.jsp Request URI: /respub/WEB-INF/jsp/search-html.jsp Query String: id=00004259&limit=30&person=true&page=6&respub-action=search.html