Imperial College London

Dr Patrick A. Naylor

Faculty of EngineeringDepartment of Electrical and Electronic Engineering

Professor of Speech & Acoustic Signal Processing
 
 
 
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Contact

 

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

 
 
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Location

 

803Electrical EngineeringSouth Kensington Campus

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Summary

 

Publications

Publication Type
Year
to

331 results found

Gaubitch ND, Löllmann HW, Jeub M, Falk TH, Naylor PA, Vary P, Brookes Met al., 2012, Performance comparison of algorithms for blind reverberation time estimation from speech

© 2012, Institute of Electrical and Electronics Engineers Inc. All rights reserved. The reverberation time, T60, is one of the key parameters used to quantify room acoustics. It can provide information about the quality and intelligibility of speech recorded in a reverberant environment, and it can be used to increase robustness to reverberation of speech processing algorithms. T60 can be determined directly from a measurement of the acoustic impulse response, but in situations where this is unavailable it must be estimated blindly from reverberant speech. In this contribution, we provide a study of three state-of-the-art methods for blind T60 estimation. Experimental results with a large number of talkers, simulated and measured acoustic impulse responses, and various levels of additive white Gaussian noise are presented. The relative merits of the three methods in terms of computational time, estimation accuracy, noise sensitivity and inter-talker variance are discussed. In general, all three methods are able to estimate the reverberation time to within 0.2 s for T60 ≤ 0.8 s and SNR ≥ 30 dB, while increasing the noise level causes overestimation. The relative computational speed of the three methods is also assessed.

Conference paper

Naylor PA, Gaubitch ND, 2012, Acoustic signal processing in noise: It's not getting any quieter

© 2012, Institute of Electrical and Electronics Engineers Inc. All rights reserved. Acoustic signal processing research has been addressing the issues associated with additive noise and other degradations in speech for many years and several significant technical advances are now embedded in the state-of-the-art. Nevertheless, the problems are not solved and may actually be worsening. The philosophy advocated in this paper is that further improvements in acoustic signal processing for noise reduction and robustness are, of course, important but are unlikely to be sufficient on their own. Alongside the signal processing, successful systems are likely going to need to include two further factors: an element of matching to the human perception system and also an element of sensing and adaptation to the local environment, giving systems acoustic awareness. Examples of current research on human perception and acoustic signal processing are discussed. These include some aspects of auditory cognition and signal processing methods for building acoustic awareness. A new initiative for benchmarking is also highlighted.

Conference paper

Jarrett D, Habets EAP, Thomas M, Naylor PAet al., 2012, Rigid sphere room impulse response simulation: algorithm and applications, J. Acoust. Soc. America, Vol: 132

Journal article

Annibale P, Filos J, Naylor PA, Rabenstein Ret al., 2012, TDOA-based speed of sound estimation for air temperature and room geometry inference, IEEE Trans. Audio, Speech, Lang. Process.

Journal article

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

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

Conference paper

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

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

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

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

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

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

Conference paper

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

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

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

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

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

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

Conference paper

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

Conference paper

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

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

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