Publications
443 results found
Nelke CM, Naylor PA, Vary P, 2015, CORPUS BASED RECONSTRUCTION OF SPEECH DEGRADED BY WIND NOISE, 23rd European Signal Processing Conference (EUSIPCO), Publisher: IEEE, Pages: 864-868, ISSN: 2076-1465
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- Citations: 1
Javed HA, Naylor PA, 2015, AN EXTENDED REVERBERATION DECAY TAIL METRIC AS A MEASURE OF PERCEIVED LATE REVERBERATION, 23rd European Signal Processing Conference (EUSIPCO), Publisher: IEEE, Pages: 1063-1067, ISSN: 2076-1465
Cauchi B, Naylor PA, Gerkmann T, et al., 2015, LATE REVERBERANT SPECTRAL VARIANCE ESTIMATION USING ACOUSTIC CHANNEL EQUALIZATION, 23rd European Signal Processing Conference (EUSIPCO), Publisher: IEEE, Pages: 2481-2485, ISSN: 2076-1465
Sharma D, Poddar A, Manna S, et al., 2015, THE SAS PROJECT: SPEECH SIGNAL PROCESSING IN HIGH SCHOOL EDUCATION, 23rd European Signal Processing Conference (EUSIPCO), Publisher: IEEE, Pages: 1781-1785, ISSN: 2076-1465
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- Citations: 2
Lim F, Naylor PA, Thomas MRP, et al., 2015, ACOUSTIC BLUR KERNEL WITH SLIDING WINDOW FOR BLIND ESTIMATION OF REVERBERATION TIME, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), Publisher: IEEE, ISSN: 1931-1168
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- Citations: 1
Hu M, Parada PP, Sharma D, et al., 2015, SINGLE-CHANNEL SPEAKER DIARIZATION BASED ON SPATIAL FEATURES, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), Publisher: IEEE, ISSN: 1931-1168
Lim F, Zhang W, Habets EAP, et al., 2014, Robust multichannel dereverberation using relaxed multichannel least squares, IEEE ACM Transactions on Audio, Speech, and Language Processing, Vol: 22, Pages: 1379-1390, ISSN: 1558-7916
A novel approach is proposed for robust multichannel dereverberation in the presence of system identification error (SIEs), based on channel shortening. A mathematical link is derived between the well known multiple-input/output inverse theorem (MINT) algorithm and channel shortening. The relaxed multichannel least squares (RMCLS) algorithm is then proposed as an efficient realization within the channel shortening paradigm and is shown through experimental results to outperform MINT in the presence of SIEs. While the RMCLS is robust to SIEs, the coloration of the output cannot be controlled. Two extensions to RMCLS are proposed to control the level of coloration and the performances of both extensions are evaluated comparatively. It is shown that both substantially maintain the dereverberation performance and robustness to SIEs obtained from RMCLS while effectively controlling the level of coloration introduced.
Evers C, Moore AH, Naylor PA, 2014, Multiple source localisation in the spherical harmonic domain
Moore AH, Naylor PA, Skoglund J, 2014, An Analysis of the Effect of Larynx-Synchronous Averaging on Dereverberation of Voiced Speech, European Signal Processing Conference, ISSN: 2219-5491
Eaton J, Naylor PA, 2014, Detection of clipping in coded speech signals, 21st European Signal Processing Conference (EUSIPCO), Publisher: IEEE
In order to exploit the full dynamic range of communicationsand recording equipment, and to minimise the effects of noiseand interference, input gain to a recording device is typicallyset as high as possible. This often leads to the signal exceedingthe input limit of the equipment resulting in clipping. Com-munications devices typically rely on codecs such as GSM06.10to compress voice signals into lower bitrates. Althoughdetecting clipping in a hard-clipped speech signal is straight-forward due to the characteristic flattening of the peaks of thewaveform, this is not the case for speech that has subsequentlypassed through a codec. We describe a novel clipping detec-tion algorithm based on amplitude histogram analysis and leastsquares residuals which can estimate the clipped samples andthe original signal level in speech even after the clipped speechhas been perceptually coded.
Jarrett DP, Taseska M, Habets EAP, et al., 2014, Noise Reduction in the Spherical Harmonic Domain Using a Tradeoff Beamformer and Narrowband DOA Estimates, IEEE/ACM Transactions on Audio, Speech, and Language Processing, Vol: 22, Pages: 965-976
Eaton J, Naylor PA, 2014, Noise-robust detection of peak-clipping in decoded speech, Pages: 7019-7023
Clipping is a commonplace problem in voice telecommunications and detection of clipping is useful in a range of speech processing applications. We analyse and evaluate the performance of three previously presented algorithms for clipping detection in decoded speech in high levels of ambient noise. We identify a baseline method which is well known for clipping detection, determine experimentally the optimized operation parameter for the baseline approach, and use this in our experiments. Our results indicate that the new algorithms outperform the baseline except at extreme levels of clipping and negative signal-to-noise ratios.
Stanton R, Gaubitch N, Naylor P, et al., 2014, A Differentiable Approximation to Speech Intelligibility Index with Applications to Listening Enhancement, AES Intl Conf on Audio Forensics
The Speech Intelligibility Index is a standardised objective measure for estimating the intelligibility of speech in noise. It is, however difficult to use it in the iterative optimisation of speech enhancement algorithms because it is a discontinuous function of its input parameters. In this paper, we derive an approximation for the Speech Intelligibility Index that is both continuous and differentiable, which allows for more efficient optimisation procedures. The use of the approximation is demonstrated in an application to near-end speech enhancement.
Parada PP, Sharma D, Naylor PA, 2014, Non-intrusive estimation of the level of reverberation in speech, Pages: 4718-4722, ISSN: 1520-6149
We show corroborating evidence that, among a set of common acoustic parameters, the clarity index C50 provides a measure of reverberation that is well correlated with speech recognition accuracy. We also present a data driven method for non-intrusive C50 parameter estimation from a single channel speech signal. The method extracts a number of features from the speech signal and uses a binary regression tree, trained on appropriate training data, to estimate the C50. Evaluation is carried out using speech utterances convolved with real and simulated room impulse responses, and additive babble noise. The new method outperforms a baseline approach in our evaluation. © 2014 IEEE.
Parada PP, Sharma D, Lainez J, et al., 2014, A QUANTITATIVE COMPARISON OF BLIND C<sub>50</sub> ESTIMATORS, 14th International Workshop on Acoustic Signal Enhancement (IWAENC), Publisher: IEEE, Pages: 298-302
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- Citations: 2
Lim F, Naylor PA, 2014, STATISTICAL MODELLING OF MULTICHANNEL BLIND SYSTEM IDENTIFICATION ERRORS, 14th International Workshop on Acoustic Signal Enhancement (IWAENC), Publisher: IEEE, Pages: 119-123
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- Citations: 4
Antonello N, van Waterschoot T, Moonen M, et al., 2014, IDENTIFICATION OF SURFACE ACOUSTIC IMPEDANCES IN A REVERBERANT ROOM USING THE FDTD METHOD, 14th International Workshop on Acoustic Signal Enhancement (IWAENC), Publisher: IEEE, Pages: 114-118
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- Citations: 5
Thomas MRP, Tashev IJ, Lim F, et al., 2014, OPTIMAL BEAMFORMING AS A TIME DOMAIN EQUALIZATION PROBLEM WITH APPLICATION TO ROOM ACOUSTICS, 14th International Workshop on Acoustic Signal Enhancement (IWAENC), Publisher: IEEE, Pages: 75-79
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- Citations: 3
Zahedi A, Ostergaard J, Jensen SH, et al., 2014, Distributed Remote Vector Gaussian Source Coding for Wireless Acoustic Sensor Networks, Data Compression Conference (DCC), Publisher: IEEE COMPUTER SOC, Pages: 263-272, ISSN: 1068-0314
Costa MH, Naylor PA, 2014, ILD PRESERVATION IN THE MULTICHANNEL WIENER FILTER FOR BINAURAL HEARING AID APPLICATIONS, 22nd European Signal Processing Conference (EUSIPCO), Publisher: IEEE, Pages: 636-640, ISSN: 2076-1465
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- Citations: 7
Antonello N, van Waterschool T, Moonen M, et al., 2014, SOURCE LOCALIZATION AND SIGNAL RECONSTRUCTION IN A REVERBERANT FIELD USING THE FDTD METHOD, 22nd European Signal Processing Conference (EUSIPCO), Publisher: IEEE, Pages: 301-305, ISSN: 2076-1465
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- Citations: 7
Sharma D, Meredith L, Lainez J, et al., 2014, A Non-Intrusive PESQ Measure, IEEE Global Conference on Signal and Information Processing (GlobalSIP), Publisher: IEEE, Pages: 975-978
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- Citations: 10
Parada PP, Sharma D, Naylor PA, et al., 2014, REVERBERANT SPEECH RECOGNITION: A PHONEME ANALYSIS, IEEE Global Conference on Signal and Information Processing (GlobalSIP), Publisher: IEEE, Pages: 567-571
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- Citations: 4
Borges RC, Costa MH, Naylor PA, et al., 2014, Impact of the vent size in the feedback-path and occlusion-effect in hearing aids, IEEE Biomedical Circuits and Systems Conference (BioCAS), Publisher: IEEE, Pages: 25-28, ISSN: 2163-4025
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- Citations: 5
Zahedi A, Ostergaard J, Jensen SH, et al., 2014, Distributed Remote Vector Gaussian Source Coding with Covariance Distortion Constraints, IEEE International Symposium on Information Theory (ISIT), Publisher: IEEE, Pages: 586-590
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- Citations: 3
Moore AH, Brookes M, Naylor PA, 2013, Roomprints for forensic audio applications, Proc. IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), Publisher: IEEE
A roomprint is a quantifiable description of an acoustic environment which can be measured under controlled conditions and estimated from a monophonic recording made in that space. We here identify the properties required of a roomprint in forensic audio applications and review the observable characteristics of a room that, when extracted from recordings, could form the basis of a roomprint. Frequency-dependent reverberation time is investigated as a promising characteristic and used in a room identification experiment giving correct identification in 96% of trials.
Gaubitch N, Brookes M, Naylor P, 2013, Blind Channel Magnitude Response Estimation in Speech using Spectrum Classification, IEEE Transactions on Audio, Speech, and Language Processing, Vol: 21, Pages: 2162-2171, ISSN: 1558-7916
Moore AH, Brookes M, Naylor PA, 2013, Room geometry estimation from a single channel acoustic impulse response, Proc. European Signal Processing Conference (EUSIPCO)
Eaton D, Brookes DM, Naylor PA, 2013, A Comparison of Non-Intrusive SNR Estimation Algorithms and the Use of Mapping Functions, EUSIPCO, Publisher: EURASIP, Pages: 1-5
We present a comparative evaluation of six methods for non-intrusive Signal-to-Noise Ratio (SNR) estimation for narrowband speech in noise. We demonstrate that the performance of all methods can be improved by applying a non-linear mapping function to their estimates of SNR. We have employed phrases built from the TIMIT speech corpus and noises from a broad range of sources including ITU-T P.501, NOISEX-92, and Soundjay. We compare the accuracy of the methods in estimating the SNR of both stationary and non-stationary noise and we conclude that with the mapping function, the best current methods can estimate the SNR to within approximately 3.5 dB for SNRs from -5 dB to 35 dB.
Kowalczyk K, Habets EAP, Kellermann W, et al., 2013, Blind System Identification Using Sparse Learning for TDOA Estimation of Room Reflections, IEEE SIGNAL PROCESSING LETTERS, Vol: 20, Pages: 653-656, ISSN: 1070-9908
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- Citations: 35
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