Patrick A. Naylor

Orcid: 0000-0001-8546-8013

Affiliations:
  • Imperial College London, UK


According to our database1, Patrick A. Naylor authored at least 319 papers between 1988 and 2024.

Collaborative distances:
  • Dijkstra number2 of four.
  • Erdős number3 of four.

Timeline

Legend:

Book 
In proceedings 
Article 
PhD thesis 
Dataset
Other 

Links

Online presence:

On csauthors.net:

Bibliography

2024
Steered Response Power for Sound Source Localization: a tutorial review.
EURASIP J. Audio Speech Music. Process., December, 2024

XANE: eXplainable Acoustic Neural Embeddings.
CoRR, 2024

XANE Background Acoustic Embeddings: Ablation and Clustering Analysis.
Proceedings of the 18th International Workshop on Acoustic Signal Enhancement, 2024

LASER: Language-Queried Speech Enhancer.
Proceedings of the 18th International Workshop on Acoustic Signal Enhancement, 2024

Latency-Agnostic Speech Enhancement for Wireless Acoustic Sensor Networks Using Polynomial Eigenvalue Decomposition.
Proceedings of the 18th International Workshop on Acoustic Signal Enhancement, 2024

Binaural Speech Enhancement Using Deep Complex Convolutional Transformer Networks.
Proceedings of the IEEE International Conference on Acoustics, 2024

Speech Enhancement in Hearing Aids Using Target Speech Presence Estimation Based on a Delayed Remote Microphone Signal.
Proceedings of the IEEE International Conference on Acoustics, 2024

Microphone Pair Selection for Sound Source Localization in Massive Arrays of Spatially Distributed Microphones.
Proceedings of the 32nd European Signal Processing Conference, 2024

2023
Dual input neural networks for positional sound source localization.
EURASIP J. Audio Speech Music. Process., December, 2023

Polynomial Eigenvalue Decomposition for Multichannel Broadband Signal Processing: A mathematical technique offering new insights and solutions.
IEEE Signal Process. Mag., November, 2023

Audio Signal Processing in the 21st Century: The important outcomes of the past 25 years.
IEEE Signal Process. Mag., July, 2023

Using a single-channel reference with the MBSTOI binaural intelligibility metric.
Speech Commun., April, 2023

Signal Compaction Using Polynomial EVD for Spherical Array Processing With Applications.
IEEE ACM Trans. Audio Speech Lang. Process., 2023

Uncertainty Quantification in Machine Learning for Joint Speaker Diarization and Identification.
CoRR, 2023

Subspace Hybrid MVDR Beamforming for Augmented Hearing.
CoRR, 2023

Long-term Conversation Analysis: Exploring Utility and Privacy.
CoRR, 2023

Two-Stage Voice Anonymization for Enhanced Privacy.
Proceedings of the 24th Annual Conference of the International Speech Communication Association, 2023

Speech enhancement using binary estimator selection applied to hearing aids with a remote microphone.
Proceedings of the 8th International Conference on Frontiers of Signal Processing, 2023

Subspace Hybrid Beamforming for Head-Worn Microphone Arrays.
Proceedings of the IEEE International Conference on Acoustics, 2023

The MBSTOI Binaural Intelligibility Metric Using a Close-Talking Microphone Reference.
Proceedings of the IEEE International Conference on Acoustics, 2023

Graph Neural Networks for Sound Source Localization on Distributed Microphone Networks.
Proceedings of the IEEE International Conference on Acoustics, 2023

Canonical Voice Conversion and Dual-Channel Processing for Improved Voice Privacy of Speech Recognition Data.
Proceedings of the 31st European Signal Processing Conference, 2023

Room Adaptation of Training Data for Distant Speech Recognition.
Proceedings of the 31st European Signal Processing Conference, 2023

Binaural Speech Enhancement Using Complex Convolutional Recurrent Networks.
Proceedings of the 57th Asilomar Conference on Signals, Systems, and Computers, ACSSC 2023, Pacific Grove, CA, USA, October 29, 2023

Distant, Multichannel Speech Recognition Using Microphone Array Coding and Cloud-Based Beamforming with a Self-Attention Channel Combinator.
Proceedings of the 57th Asilomar Conference on Signals, Systems, and Computers, ACSSC 2023, Pacific Grove, CA, USA, October 29, 2023

Zero Shot Text to Speech Augmentation for Automatic Speech Recognition on Low-Resource Accented Speech Corpora.
Proceedings of the 57th Asilomar Conference on Signals, Systems, and Computers, ACSSC 2023, Pacific Grove, CA, USA, October 29, 2023

The Neural-SRP Method for Positional Sound Source Localization.
Proceedings of the 57th Asilomar Conference on Signals, Systems, and Computers, ACSSC 2023, Pacific Grove, CA, USA, October 29, 2023

Uncovering the Potential for a Weakly Supervised End-to-End Model in Recognising Speech from Patient with Post-Stroke Aphasia.
Proceedings of the 5th Clinical Natural Language Processing Workshop, 2023

2022
A Compact Noise Covariance Matrix Model for MVDR Beamforming.
IEEE ACM Trans. Audio Speech Lang. Process., 2022

An audio enhancement system to improve intelligibility for social-awareness in HRI.
Multim. Tools Appl., 2022

Binaural Speech Enhancement Using STOI-optimal Masks.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

A Linear MMSE Filter Using Delayed Remote Microphone Signals for Speech Enhancement in Hearing Aid Applications.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

Fixed Beamformer Design Using Polynomial Eigenvalue Decomposition.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

Polynomial Eigenvalue Decomposition-Based Target Speaker Voice Activity Detection in the Presence of Competing Talkers.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

Machine Learning for Parameter Estimation in the MBSTOI Binaural Intelligibility Metric.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

An Introduction to the Speech Enhancement for Augmented Reality (Spear) Challenge.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

Deep Complex-Valued Convolutional-Recurrent Networks for Single Source DOA Estimation.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

Frame-Based Space-Time Covariance Matrix Estimation for Polynomial Eigenvalue Decomposition-Based Speech Enhancement.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

A Distributed Steered Response Power Approach to Source Localization in Wireless Acoustic Sensor Networks.
Proceedings of the 17th International Workshop on Acoustic Signal Enhancement, 2022

Relative Acoustic Features for Distance Estimation in Smart-Homes.
Proceedings of the 23rd Annual Conference of the International Speech Communication Association, 2022

Spatial Processing Front-End for Distant ASR Exploiting Self-Attention Channel Combinator.
Proceedings of the IEEE International Conference on Acoustics, 2022

Non-Intrusive Signal Analysis for Room Adaptation of ASR Models.
Proceedings of the 30th European Signal Processing Conference, 2022

Microphone Array Coding Preserving Spatial Information for Cloud-based Multichannel Speech Recognition.
Proceedings of the 30th European Signal Processing Conference, 2022

Speech Enhancement in Distributed Microphone Arrays Using Polynomial Eigenvalue Decomposition.
Proceedings of the 30th European Signal Processing Conference, 2022

2021
Time-Frequency Analysis and Parameterisation of Knee Sounds for Non-Invasive Detection of Osteoarthritis.
IEEE Trans. Biomed. Eng., 2021

Speech Enhancement Based on Modulation-Domain Parametric Multichannel Kalman Filtering.
IEEE ACM Trans. Audio Speech Lang. Process., 2021

Enhancement of Noisy Reverberant Speech Using Polynomial Matrix Eigenvalue Decomposition.
IEEE ACM Trans. Audio Speech Lang. Process., 2021

Overlapping Speaker Segmentation Using Multiple Hypothesis Tracking of Fundamental Frequency.
IEEE ACM Trans. Audio Speech Lang. Process., 2021

Polynomial Matrix Eigenvalue Decomposition-Based Source Separation Using Informed Spherical Microphone Arrays.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2021

Spatial Coding for Microphone Arrays Using Ipnlms-Based RTF Estimation.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2021

A Polynomial Eigenvalue Decomposition Music Approach for Broadband Sound Source Localization.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2021

Polynomial Matrix Eigenvalue Decomposition of Spherical Harmonics for Speech Enhancement.
Proceedings of the IEEE International Conference on Acoustics, 2021

Processing Pipelines for Efficient, Physically-Accurate Simulation of Microphone Array Signals in Dynamic Sound Scenes.
Proceedings of the IEEE International Conference on Acoustics, 2021

Multichannel Overlapping Speaker Segmentation Using Multiple Hypothesis Tracking Of Acoustic And Spatial Features.
Proceedings of the IEEE International Conference on Acoustics, 2021

Model-based Beamforming for Wearable Microphone Arrays.
Proceedings of the 29th European Signal Processing Conference, 2021

A Study of Salient Modulation Domain Features for Speaker Identification.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2021

2020
Data Corpus for the IEEE-AASP Challenge on Acoustic Source Localization and Tracking (LOCATA).
Dataset, January, 2020

End-to-End Classification of Reverberant Rooms Using DNNs.
IEEE ACM Trans. Audio Speech Lang. Process., 2020

The LOCATA Challenge: Acoustic Source Localization and Tracking.
IEEE ACM Trans. Audio Speech Lang. Process., 2020

Evaluation of a Multi-speaker System for Socially Assistive HRI in Real Scenarios.
Proceedings of the Advances in Physical Agents II, 2020

PEVD-Based Speech Enhancement in Reverberant Environments.
Proceedings of the 2020 IEEE International Conference on Acoustics, 2020

Non-Intrusive Estimation of Speech Signal Parameters using a Frame-based Machine Learning Approach.
Proceedings of the 28th European Signal Processing Conference, 2020

Speech Dereverberation Performance of a Polynomial-EVD Subspace Approach.
Proceedings of the 28th European Signal Processing Conference, 2020

Analysis of Phonetic Dependence of Segmentation Errors in Speaker Diarization.
Proceedings of the 28th European Signal Processing Conference, 2020

Head Orientation Estimation from Multiple Microphone Arrays.
Proceedings of the 28th European Signal Processing Conference, 2020

2019
OTIMP: The Oticon-Imperial hearing aid impulse response database.
Dataset, May, 2019

OTIMP: The Oticon-Imperial hearing aid impulse response database.
Dataset, May, 2019

Noise Covariance Matrix Estimation for Rotating Microphone Arrays.
IEEE ACM Trans. Audio Speech Lang. Process., 2019

Joint Acoustic Localization and Dereverberation Through Plane Wave Decomposition and Sparse Regularization.
IEEE ACM Trans. Audio Speech Lang. Process., 2019

Highlights From the Audio and Acoustic Signal Processing Technical Committee [In the Spotlight].
IEEE Signal Process. Mag., 2019

Using DNNs to Detect Materials in a Room based on Sound Absorption.
CoRR, 2019

Data Augmentation of Room Classifiers using Generative Adversarial Networks.
CoRR, 2019

Speech Enhancement Using Polynomial Eigenvalue Decomposition.
Proceedings of the 2019 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2019

Multiple Hypothesis Tracking for Overlapping Speaker Segmentation.
Proceedings of the 2019 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2019

Second Order Sequential Best Rotation Algorithm with Householder Reduction for Polynomial Matrix Eigenvalue Decomposition.
Proceedings of the IEEE International Conference on Acoustics, 2019

Speaker Change Detection Using Fundamental Frequency with Application to Multi-talker Segmentation.
Proceedings of the IEEE International Conference on Acoustics, 2019

Non-Intrusive POLQA Estimation of Speech Quality using Recurrent Neural Networks.
Proceedings of the 27th European Signal Processing Conference, 2019

2018
Optimized Self-Localization for SLAM in Dynamic Scenes Using Probability Hypothesis Density Filters.
IEEE Trans. Signal Process., 2018

Modulation-Domain Multichannel Kalman Filtering for Speech Enhancement.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

Acoustic SLAM.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

DoA Reliability for Distributed Acoustic Tracking.
IEEE Signal Process. Lett., 2018

Proceedings of the LOCATA Challenge Workshop - a satellite event of IWAENC 2018.
CoRR, 2018

Robust Feature Extraction from AD-HOC Microphones for Meeting Diarization.
Proceedings of the 16th International Workshop on Acoustic Signal Enhancement, 2018

Binaural Mask-Informed Speech Enhancement for Hearing AIDS with Head Tracking.
Proceedings of the 16th International Workshop on Acoustic Signal Enhancement, 2018

Locata Challenge-Evaluation Tasks and Measures.
Proceedings of the 16th International Workshop on Acoustic Signal Enhancement, 2018

The LOCATA Challenge Data Corpus for Acoustic Source Localization and Tracking.
Proceedings of the 10th IEEE Sensor Array and Multichannel Signal Processing Workshop, 2018

Robust Source Counting and Acoustic DOA Estimation using Density-Based Clustering.
Proceedings of the 10th IEEE Sensor Array and Multichannel Signal Processing Workshop, 2018

Acoustic Analysis and Assessment of the Knee in Osteoarthritis During Walking.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Multichannel Kalman Filtering for Speech Ehnancement.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Room Identification Using Frequency Dependence of Spectral Decay Statistics.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Joint Source Localization and Dereverberation by Sound Field Interpolation Using Sparse Regularization.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Modulation-Domain Parametric Multichannel Kalman Filtering for Speech Enhancement.
Proceedings of the 26th European Signal Processing Conference, 2018

An acoustic image-source characterisation of surface profiles.
Proceedings of the 26th European Signal Processing Conference, 2018

Estimation of the Noise Covariance Matrix for Rotating Sensor Arrays.
Proceedings of the 52nd Asilomar Conference on Signals, Systems, and Computers, 2018

2017
Direction of Arrival Estimation in the Spherical Harmonic Domain Using Subspace Pseudointensity Vectors.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

Augmented Intensity Vectors for Direction of Arrival Estimation in the Spherical Harmonic Domain.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

Single-Channel Online Enhancement of Speech Corrupted by Reverberation and Noise.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

Room Impulse Response Interpolation Using a Sparse Spatio-Temporal Representation of the Sound Field.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

Speech enhancement for robust automatic speech recognition: Evaluation using a baseline system and instrumental measures.
Comput. Speech Lang., 2017

A dynamic programming approach for automatic stride detection and segmentation in acoustic emission from the knee.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Frequency-domain under-modelled blind system identification based on cross power spectrum and sparsity regularization.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Channel estimation for crosstalk cancellation in wireless acoustic networks.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Discriminative feature domains for reverberant acoustic environments.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Robust spherical harmonic domain interpolation of spatially sampled array manifolds.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Improving the perceptual quality of ideal binary masked speech.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Measuring, modelling and predicting perceived reverberation.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Multiple source localization using Estimation Consistency in the Time-Frequency domain.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Source tracking using moving microphone arrays for robot audition.
Proceedings of the 2017 IEEE International Conference on Acoustics, 2017

Microphone array signal processing for robot audition.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

Multi-source estimation consistency for improved multiple direction-of-arrival estimation.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

Audio-visual tracking by density approximation in a sequential Bayesian filtering framework.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

Speaker tracking in reverberant environments using multiple directions of arrival.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

Estimation of the perceived level of reverberation using non-intrusive single-channel variance of decay rates.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

Non-intrusive bit-rate detection of coded speech.
Proceedings of the 25th European Signal Processing Conference, 2017

Robust statistical processing of TDOA estimates for distant speaker diarization.
Proceedings of the 25th European Signal Processing Conference, 2017

Sparse parametric modeling of the early part of acoustic impulse responses.
Proceedings of the 25th European Signal Processing Conference, 2017

Multiple DOA estimation based on estimation consistency and spherical harmonic multiple signal classification.
Proceedings of the 25th European Signal Processing Conference, 2017

2016
Source Coding in Networks With Covariance Distortion Constraints.
IEEE Trans. Signal Process., 2016

A Single-Channel Non-Intrusive C50 Estimator Correlated With Speech Recognition Performance.
IEEE ACM Trans. Audio Speech Lang. Process., 2016

Estimation of Room Acoustic Parameters: The ACE Challenge.
IEEE ACM Trans. Audio Speech Lang. Process., 2016

A data-driven non-intrusive measure of speech quality and intelligibility.
Speech Commun., 2016

Acoustic Characterization of Environments (ACE) Challenge Results Technical Report.
CoRR, 2016

On the evaluation of multichannel blind system identification from the viewpoint of system equalization.
Proceedings of the IEEE International Workshop on Acoustic Signal Enhancement, 2016

An iterative method for equalization of multichannel acoustic systems robust to system identification errors.
Proceedings of the IEEE International Workshop on Acoustic Signal Enhancement, 2016

Under-modelled blind system identification for time delay estimation in reverberant environments.
Proceedings of the IEEE International Workshop on Acoustic Signal Enhancement, 2016

Linear prediction based dereverberation for spherical microphone arrays.
Proceedings of the IEEE International Workshop on Acoustic Signal Enhancement, 2016

Spherical harmonic rake receivers for dereverberation.
Proceedings of the IEEE International Workshop on Acoustic Signal Enhancement, 2016

Spherical microphone array acoustic rake receivers.
Proceedings of the 2016 IEEE International Conference on Acoustics, 2016

3D acoustic source localization in the spherical harmonic domain based on optimized grid search.
Proceedings of the 2016 IEEE International Conference on Acoustics, 2016

Acoustic simultaneous localization and mapping (A-SLAM) of a moving microphone array and its surrounding speakers.
Proceedings of the 2016 IEEE International Conference on Acoustics, 2016

Perceptual and instrumental evaluation of the perceived level of reverberation.
Proceedings of the 2016 IEEE International Conference on Acoustics, 2016

Cross-correlation based under-modelled multichannel blind acoustic system identification with sparsity regularization.
Proceedings of the 24th European Signal Processing Conference, 2016

2D direction of arrival estimation of multiple moving sources using a spherical microphone array.
Proceedings of the 24th European Signal Processing Conference, 2016

Multiple source localization in the spherical harmonic domain using augmented intensity vectors based on grid search.
Proceedings of the 24th European Signal Processing Conference, 2016

Localization of moving microphone arrays from moving sound sources for robot audition.
Proceedings of the 24th European Signal Processing Conference, 2016

Speaker localization with moving microphone arrays.
Proceedings of the 24th European Signal Processing Conference, 2016

On Perceptual Audio Compression with Side Information at the Decoder.
Proceedings of the 2016 Data Compression Conference, 2016

Predicting the quality of processed speech by combining modulation-based features and model trees.
Proceedings of the 12th ITG Symposium on Speech Communication, 2016

2015
Data Corpus for the IEEE-AASP Challenge on the Acoustic Characterization of Environments (ACE).
Dataset, March, 2015

Audio coding in wireless acoustic sensor networks.
Signal Process., 2015

Reverberant speech recognition exploiting clarity index estimation.
EURASIP J. Adv. Signal Process., 2015

On the Covariance Matrix Distortion Constraint for the Gaussian Wyner-Ziv Problem.
CoRR, 2015

Evaluating the Non-Intrusive Room Acoustics Algorithm with the ACE Challenge.
CoRR, 2015

Direct-to-Reverberant Ratio Estimation on the ACE Corpus Using a Two-channel Beamformer.
CoRR, 2015

Reverberation time estimation on the ACE corpus using the SDD method.
CoRR, 2015

Proceedings of the ACE Challenge Workshop - a satellite event of IEEE-WASPAA (2015).
CoRR, 2015

Acoustic blur kernel with sliding window for blind estimation of reverberation time.
Proceedings of the 2015 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2015

Single-channel speaker diarization based on spatial features.
Proceedings of the 2015 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2015

The ACE challenge - Corpus description and performance evaluation.
Proceedings of the 2015 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2015

Multichannel equalisation for high-order spherical microphone arrays using beamformed channels.
Proceedings of the 2015 IEEE International Conference on Digital Signal Processing, 2015

Bearing-only acoustic tracking of moving speakers for robot audition.
Proceedings of the 2015 IEEE International Conference on Digital Signal Processing, 2015

Speaker change detection and speaker diarization using spatial information.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

Direct-to-Reverberant Ratio estimation using a null-steered beamformer.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

Single-channel blind estimation of reverberation parameters.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

The SAS project: Speech signal processing in high school education.
Proceedings of the 23rd European Signal Processing Conference, 2015

Corpus based reconstruction of speech degraded by wind noise.
Proceedings of the 23rd European Signal Processing Conference, 2015

Direction of arrival estimation using pseudo-intensity vectors with direct-path dominance test.
Proceedings of the 23rd European Signal Processing Conference, 2015

An extended reverberation decay tail metric as a measure of perceived late reverberation.
Proceedings of the 23rd European Signal Processing Conference, 2015

Noise robust blind system identification algorithms based on a Rayleigh quotient cost function.
Proceedings of the 23rd European Signal Processing Conference, 2015

Modelling source directivity in room impulse response simulation for spherical microphone arrays.
Proceedings of the 23rd European Signal Processing Conference, 2015

Data-driven statistical modelling of room impulse responses in the power domain.
Proceedings of the 23rd European Signal Processing Conference, 2015

Late reverberant spectral variance estimation using acoustic channel equalization.
Proceedings of the 23rd European Signal Processing Conference, 2015

Coding and Enhancement in Wireless Acoustic Sensor Networks.
Proceedings of the 2015 Data Compression Conference, 2015

2014
Robust Multichannel Dereverberation using Relaxed Multichannel Least Squares.
IEEE ACM Trans. Audio Speech Lang. Process., 2014

Noise Reduction in the Spherical Harmonic Domain Using a Tradeoff Beamformer and Narrowband DOA Estimates.
IEEE ACM Trans. Audio Speech Lang. Process., 2014

Optimal beamforming as a time domain equalization problem with application to room acoustics.
Proceedings of the 14th International Workshop on Acoustic Signal Enhancement, 2014

A quantitative comparison of blind C50 estimators.
Proceedings of the 14th International Workshop on Acoustic Signal Enhancement, 2014

Statistical modelling of multichannel blind system identification errors.
Proceedings of the 14th International Workshop on Acoustic Signal Enhancement, 2014

Multiple source localisation in the spherical harmonic domain.
Proceedings of the 14th International Workshop on Acoustic Signal Enhancement, 2014

Identification of surface acoustic impedances in a reverberant room using the FDTD method.
Proceedings of the 14th International Workshop on Acoustic Signal Enhancement, 2014

Distributed remote vector gaussian source coding with covariance distortion constraints.
Proceedings of the 2014 IEEE International Symposium on Information Theory, Honolulu, HI, USA, June 29, 2014

Non-intrusive estimation of the level of reverberation in speech.
Proceedings of the IEEE International Conference on Acoustics, 2014

Noise-robust detection of peak-clipping in decoded speech.
Proceedings of the IEEE International Conference on Acoustics, 2014

A non-intrusive PESQ measure.
Proceedings of the 2014 IEEE Global Conference on Signal and Information Processing, 2014

Reverberant speech recognition: A phoneme analysis.
Proceedings of the 2014 IEEE Global Conference on Signal and Information Processing, 2014

An analysis of the effect of larynx-synchronous averaging on dereverberation of voiced speech.
Proceedings of the 22nd European Signal Processing Conference, 2014

ILD preservation in the multichannel wiener filter for binaural hearing aid applications.
Proceedings of the 22nd European Signal Processing Conference, 2014

Source localization and signal reconstruction in a reverberant field using the FDTD method.
Proceedings of the 22nd European Signal Processing Conference, 2014

Distributed Remote Vector Gaussian Source Coding for Wireless Acoustic Sensor Networks.
Proceedings of the Data Compression Conference, 2014

Impact of the vent size in the feedback-path and occlusion-effect in hearing aids.
Proceedings of the IEEE Biomedical Circuits and Systems Conference, 2014

2013
Blind Channel Magnitude Response Estimation in Speech Using Spectrum Classification.
IEEE Trans. Speech Audio Process., 2013

TDOA-Based Speed of Sound Estimation for Air Temperature and Room Geometry Inference.
IEEE Trans. Speech Audio Process., 2013

Blind System Identification Using Sparse Learning for TDOA Estimation of Room Reflections.
IEEE Signal Process. Lett., 2013

Roomprints for forensic audio applications.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2013

MINTFormer: A spatially aware channel equalizer.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2013

Robust low-complexity multichannel equalization for dereverberation.
Proceedings of the IEEE International Conference on Acoustics, 2013

Spherical harmonic domain noise reduction using an MVDR beamformer and DOA-based second-order statistics estimation.
Proceedings of the IEEE International Conference on Acoustics, 2013

Noise-robust reverberation time estimation using spectral decay distributions with reduced computational cost.
Proceedings of the IEEE International Conference on Acoustics, 2013

Non-intrusive speech intelligibility assessment.
Proceedings of the 21st European Signal Processing Conference, 2013

Room geometry estimation from a single channel acoustic impulse response.
Proceedings of the 21st European Signal Processing Conference, 2013

Robust speech dereverberation using subband multichannel least squares with variable relaxation.
Proceedings of the 21st European Signal Processing Conference, 2013

Detection of clipping in coded speech signals.
Proceedings of the 21st European Signal Processing Conference, 2013

A comparison of non-intrusive SNR estimation algorithms and the use of mapping functions.
Proceedings of the 21st European Signal Processing Conference, 2013

2012
Estimation of Glottal Closing and Opening Instants in Voiced Speech Using the YAGA Algorithm.
IEEE Trans. Speech Audio Process., 2012

A Forced Spectral Diversity Algorithm for Speech Dereverberation in the Presence of Near-Common Zeros.
IEEE Trans. Speech Audio Process., 2012

A Speech Distortion and Interference Rejection Constraint Beamformer.
IEEE Trans. Speech Audio Process., 2012

Detection of Glottal Closure Instants From Speech Signals: A Quantitative Review.
IEEE Trans. Speech Audio Process., 2012

Inference of Room Geometry From Acoustic Impulse Responses.
IEEE Trans. Speech Audio Process., 2012

Data-driven voice source waveform analysis and synthesis.
Speech Commun., 2012

Acoustic Signal Processing in Noise: It's Not Getting Any Quieter.
Proceedings of the IWAENC 2012 - International Workshop on Acoustic Signal Enhancement, Proceedings, RWTH Aachen University, Germany, September 4th, 2012

Relaxed Multichannel Least Squares with Constrained Initial Taps for Multichannel Dereverberation.
Proceedings of the IWAENC 2012 - International Workshop on Acoustic Signal Enhancement, Proceedings, RWTH Aachen University, Germany, September 4th, 2012

A Tradeoff Beamformer for Noise Reduction in the Spherical Harmonic Domain.
Proceedings of the IWAENC 2012 - International Workshop on Acoustic Signal Enhancement, Proceedings, RWTH Aachen University, Germany, September 4th, 2012

Performance Comparison of Algorithms for Blind Reverberation Time Estimation from Speech.
Proceedings of the IWAENC 2012 - International Workshop on Acoustic Signal Enhancement, Proceedings, RWTH Aachen University, Germany, September 4th, 2012

Exact Localization of Planar Acoustic Reflectors in Three-Dimensional Geometries.
Proceedings of the IWAENC 2012 - International Workshop on Acoustic Signal Enhancement, Proceedings, RWTH Aachen University, Germany, September 4th, 2012

Geometric inference of the room geometry under temperature variations.
Proceedings of the 5th International Symposium on Communications, 2012

Descriptive Vocabulary Development for Degraded Speech.
Proceedings of the 13th Annual Conference of the International Speech Communication Association, 2012

An insight into common filtering in noisy SIMO blind system identification.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Non intrusive codec identification algorithm.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Localization of planar acoustic reflectors from the combination of linear estimates.
Proceedings of the 20th European Signal Processing Conference, 2012

2011
Audio and Acoustic Signal Processing [In the Spotlight].
IEEE Signal Process. Mag., 2011


Application of channel shortening to acoustic channel equalization in the presence of noise and estimation error.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2011

Exact localization of acoustic reflectors from quadratic constraints.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2011

A proportionate adaptive algorithm with variable partitioned block length for acoustic echo cancellation.
Proceedings of the IEEE International Conference on Acoustics, 2011

Simulating room impulse responses for spherical microphone arrays.
Proceedings of the IEEE International Conference on Acoustics, 2011

Short-time objective assessment of speech quality.
Proceedings of the 19th European Signal Processing Conference, 2011

A cross-relation based affine projection algorithm for blind SIMO system identification.
Proceedings of the 19th European Signal Processing Conference, 2011

Single-microphone blind channel identification in speech using spectrum classification.
Proceedings of the 19th European Signal Processing Conference, 2011

Robust inference of room geometry from acoustic measurements using the hough transform.
Proceedings of the 19th European Signal Processing Conference, 2011

2010
Introduction to the Special Issue on Processing Reverberant Speech: Methodologies and Applications.
IEEE Trans. Speech Audio Process., 2010

Signal-Based Performance Evaluation of Dereverberation Algorithms.
J. Electr. Comput. Eng., 2010

Microphone Array Speech Processing.
EURASIP J. Adv. Signal Process., 2010

An alternative criterion for regularization in Recursive Least-Squares problems.
Proceedings of the 2010 7th International Symposium on Wireless Communication Systems, 2010

A System-Identification-Error-Robust Method for equalization of multichannel acoustic systems.
Proceedings of the IEEE International Conference on Acoustics, 2010

Voice source estimation for artificial bandwidth extension of telephone speech.
Proceedings of the IEEE International Conference on Acoustics, 2010

Performance analysis of IPNLMS for identification of time-varying systems.
Proceedings of the IEEE International Conference on Acoustics, 2010

An online quasi-Newton algorithm for blind SIMO identification.
Proceedings of the IEEE International Conference on Acoustics, 2010

Data driven method for non-intrusive speech intelligibility estimation.
Proceedings of the 18th European Signal Processing Conference, 2010

3D source localization in the spherical harmonic domain using a pseudointensity vector.
Proceedings of the 18th European Signal Processing Conference, 2010

Adaptive blind system identification for speech dereverberation using a priori estimates.
Proceedings of the IEEE Asia Pacific Conference on Circuits and Systems, 2010

Intelligibility Estimation in Law Enforcement Speech Processing.
Proceedings of the 9. ITG-Fachtagung Sprachkommunikation 2010, 2010

Models, Measurement and Evaluation.
Proceedings of the Speech Dereverberation., 2010

Introduction.
Proceedings of the Speech Dereverberation., 2010

Adaptive Blind Multichannel System Identification.
Proceedings of the Speech Dereverberation., 2010

Dereverberation Using LPC-based Approaches.
Proceedings of the Speech Dereverberation., 2010

Subband Inversion of Multichannel Acoustic Systems.
Proceedings of the Speech Dereverberation., 2010

2009
The SIGMA Algorithm: A Glottal Activity Detector for Electroglottographic Signals.
IEEE Trans. Speech Audio Process., 2009

A Class of Sparseness-Controlled Algorithms for Echo Cancellation.
IEEE Trans. Speech Audio Process., 2009

Equalization of Multichannel Acoustic Systems in Oversampled Subbands.
IEEE Trans. Speech Audio Process., 2009

On the application of the LCMV beamformer to speech enhancement.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2009

Voice source waveform analysis and synthesis using principal component analysis and Gaussian mixture modelling.
Proceedings of the 10th Annual Conference of the International Speech Communication Association, 2009

Fast exact Affine Projection Algorithm using displacement structure theory.
Proceedings of the 16th International Conference on Digital Signal Processing, 2009

Data-driven voice soruce waveform modelling.
Proceedings of the IEEE International Conference on Acoustics, 2009

Blind system identification for speech dereverberation with Forced Spectral Diversity.
Proceedings of the IEEE International Conference on Acoustics, 2009

An experimental study of the robustness of multichannel inverse filtering systems to near-common zeros.
Proceedings of the 17th European Signal Processing Conference, 2009

Acoustic system equalization using channel shortening techniques for speech dereverberation.
Proceedings of the 17th European Signal Processing Conference, 2009

Blind estimation of a feature-domain reverberation model in non-diffuse environments with variance adjustment.
Proceedings of the 17th European Signal Processing Conference, 2009

Evaluation of pitch estimation in noisy speech for application in non-intrusive speech quality assessment.
Proceedings of the 17th European Signal Processing Conference, 2009

Frequency-domain adaptive multidelay algorithm with sparseness control for acoustic echo cancellation.
Proceedings of the 17th European Signal Processing Conference, 2009

Blind channel identification in speech using the Long-Term Average Speech Spectrum.
Proceedings of the 17th European Signal Processing Conference, 2009

2008
A Class of Frobenius Norm-Based Algorithms Using Penalty Term and Natural Gradient for Blind Signal Separation.
IEEE Trans. Speech Audio Process., 2008

An Algorithm to Generate Representations of System Identification Errors.
J. Electr. Comput. Eng., 2008

Frequency-Domain Adaptive Algorithm for Network Echo Cancellation in VoIP.
EURASIP J. Audio Speech Music. Process., 2008

Multimicrophone speech dereverberation using spatiotemporal and spectral processing.
Proceedings of the International Symposium on Circuits and Systems (ISCAS 2008), 2008

Computationally efficient equalization of room impulse responses robust to system estimation errors.
Proceedings of the IEEE International Conference on Acoustics, 2008

Blind estimation of reverberation time based on the distribution of signal decay rates.
Proceedings of the IEEE International Conference on Acoustics, 2008

Algorithms for identifying clusters of near-common zeros in multichannel blind system identification and equalization.
Proceedings of the IEEE International Conference on Acoustics, 2008

Frequency domain selective tap adaptive algorithms for sparse system identification.
Proceedings of the IEEE International Conference on Acoustics, 2008

A lowcomplexity fast converging partial update adaptive algorithm employing variable step-size for acoustic echo cancellation.
Proceedings of the IEEE International Conference on Acoustics, 2008

Temporal selective dereverberation of noisy speech using one microphone.
Proceedings of the IEEE International Conference on Acoustics, 2008

The sigma algorithm for estimation of reference-quality glottal closure instants from Electroglottograph signals.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Application of the DYPSA algorithm to segmented time scale modification of speech.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

A sparseness controlled proportionate algorithm for acoustic echo cancellation.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Adaptive inverse filtering of room acoustics.
Proceedings of the 42nd Asilomar Conference on Signals, Systems and Computers, 2008

Session MP6: Blind system identification, multi-channel system inversion, and speech dereverberation.
Proceedings of the 42nd Asilomar Conference on Signals, Systems and Computers, 2008

2007
Estimation of Glottal Closure Instants in Voiced Speech Using the DYPSA Algorithm.
IEEE Trans. Speech Audio Process., 2007

Selective-Tap Adaptive Filtering With Performance Analysis for Identification of Time-Varying Systems.
IEEE Trans. Speech Audio Process., 2007

Energy constrained frequency-domain normalized LMS algorithm for blind channel identification.
Signal Image Video Process., 2007

A Low Delay and Fast Converging Improved Proportionate Algorithm for Sparse System Identification.
EURASIP J. Audio Speech Music. Process., 2007

Adaptive Partial-Update and Sparse System Identification.
EURASIP J. Audio Speech Music. Process., 2007

Spatiotemporal Averagingmethod for Enhancement of Reverberant Speech.
Proceedings of the 15th International Conference on Digital Signal Processing, 2007

A Practical Adaptive Blind Multichannel Estimation Algorithm with Application to Acoustic Impulse Responses.
Proceedings of the 15th International Conference on Digital Signal Processing, 2007

Misalignment Performance of Selective Tap Adaptive Algorithms for System Identification of Time-Varying Unknown Systems.
Proceedings of the IEEE International Conference on Acoustics, 2007

Objective measurement of colouration in reverberation.
Proceedings of the 15th European Signal Processing Conference, 2007

Multichannel DYPSA for estimation of glottal closure instants in reverberant speech.
Proceedings of the 15th European Signal Processing Conference, 2007

Enhanced robustness to unvoiced speech and noise in the DYPSA algorithm for identification of glottal closure instants.
Proceedings of the 15th European Signal Processing Conference, 2007

Blind speech dereverberation in the presence of common acoustical zeros.
Proceedings of the 15th European Signal Processing Conference, 2007

A noise-robust dual filter approach to multichannel blind system identification.
Proceedings of the 15th European Signal Processing Conference, 2007

2006
Performance analysis of dynamic acoustic source separation in reverberant rooms.
IEEE Trans. Speech Audio Process., 2006

Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms.
IEEE Trans. Speech Audio Process., 2006

A quantitative assessment of group delay methods for identifying glottal closures in voiced speech.
IEEE Trans. Speech Audio Process., 2006

Stereophonic acoustic echo cancellation: analysis of the misalignment in the frequency domain.
IEEE Signal Process. Lett., 2006

Generalized Optimal Step-Size for Blind Multichannel LMS System Identification.
IEEE Signal Process. Lett., 2006

Adaptive algorithms for sparse echo cancellation.
Signal Process., 2006

Blind Signal Separation Using a Criterion Based on Principle of Minimal Disturbance.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Effect of Interchannel Coherence on Conditioning and Misalignment Performance for Stereo Acoustic ECHO Cancellation.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Noise Robust Adaptive Blind Channel Identification Using Spectral Constraints.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Proportionate Frequency Domain Adaptive Algorithms for Blind Channel Identification.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

An evaluation measure for reverberant speech using decay tail modelling.
Proceedings of the 14th European Signal Processing Conference, 2006

A study of echo in VoIP systems and synchronous convergence of the μ-law PNLMS algorithm.
Proceedings of the 14th European Signal Processing Conference, 2006

Two-stage blind identification of SIMO systems with common zeros.
Proceedings of the 14th European Signal Processing Conference, 2006

Analyzing effect of noise on LMS-type approaches to blind estimation of simo channels: Robustness issue.
Proceedings of the 14th European Signal Processing Conference, 2006

An extended normalized multichannel FLMS algorithm for blind channel identification.
Proceedings of the 14th European Signal Processing Conference, 2006

2005
Corrections to "Selective-Tap Adaptive Algorithms in the Solution of the Nonuniqueness Problem for Stereophonic Acoustic Echo Cancellation".
IEEE Signal Process. Lett., 2005

Selective-tap adaptive algorithms in the solution of the nonuniqueness problem for stereophonic acoustic echo cancellation.
IEEE Signal Process. Lett., 2005

Blind identification using second-order statistics: a nonstationarity and nonwhiteness approach.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

A family of selective-tap algorithms for stereo acoustic echo cancellation.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

An improved proportionate multi-delay block adaptive filter for packet-switched network echo cancellation.
Proceedings of the 13th European Signal Processing Conference, 2005

Improving robustness of blind adaptive multichannel identification algorithms using constraints.
Proceedings of the 13th European Signal Processing Conference, 2005

Adaptive common root estimation and the common zeros problem in blind channel identification.
Proceedings of the 13th European Signal Processing Conference, 2005

Recent advances in partial update and sparse adaptive filters.
Proceedings of the 13th European Signal Processing Conference, 2005

2004
Expected performance of a family of blind source separation algorithms in a reverberant room.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

An improved IPNLMS algorithm for echo cancellation in packet-switched networks.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

Reducing inter-channel coherence in stereophonic acoustic echo cancellation using partial update adaptive filters.
Proceedings of the 2004 12th European Signal Processing Conference, 2004

Multi-microphone speech dereverberation using spatio-temporal averaging.
Proceedings of the 2004 12th European Signal Processing Conference, 2004

2003
I/Q mismatch compensation in zero-IF OFDM receivers with application to DAB.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

A short-sort M-Max NLMS partial-update adaptive filter with applications to echo cancellation.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

2002
Automatic epoch extraction for closed-phase analysis of speech.
Proceedings of the 14th International Conference on Digital Signal Processing, 2002

The DYPSA algorithm for estimation of glottal closure instants in voiced speech.
Proceedings of the IEEE International Conference on Acoustics, 2002

2001
Dynamic structures for non-uniform subband adaptive filtering.
Proceedings of the IEEE International Conference on Acoustics, 2001

1998
Subband adaptive filtering for acoustic echo control using allpass polyphase IIR filterbanks.
IEEE Trans. Speech Audio Process., 1998

Application of the leaky extended LMS (XLMS) algorithm in stereophonic acoustic echo cancellation.
Signal Process., 1998

Voice source parameters for speaker verification.
Proceedings of the 9th European Signal Processing Conference, 1998

Steady-state solutions of the extended LMS algorithm for stereophonic acoustic echo cancellation with leakage or signal conditioning.
Proceedings of the 9th European Signal Processing Conference, 1998

1997
Voice activity detection using source separation techniques.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

1996
Subband acoustic echo control using non-critical frequency sampling.
Proceedings of the 8th European Signal Processing Conference, 1996

1995
Finite-precision design and implementation of all-pass polyphase networks for echo cancellation in sub-bands.
Proceedings of the 1995 International Conference on Acoustics, 1995

1993
Polyphase allpass IIR structures for sub-band acoustic echo cancellation.
Proceedings of the Third European Conference on Speech Communication and Technology, 1993

1988
Speech production modelling with variable glottal reflection coefficient.
Proceedings of the IEEE International Conference on Acoustics, 1988


  Loading...