329 results found
Gaubitch ND, Habets E, Naylor PA, 2008, Multimicrophone speech dereverberation using spatiotemporal and spectral processing, Pages: 3222-3225-3222-3225
Speech signals acquired in a reverberant room with microphones positioned at a distance from the talker are degraded in quality due to reverberation and measurement noise. Therefore, enhancement of reverberant speech is important in hands-free teleco.....
Zhang W, Khong AWH, Naylor PA, 2008, ADAPTIVE INVERSE FILTERING OF ROOM ACOUSTICS, 42nd Asilomar Conference on Signals, Systems and Computers, Publisher: IEEE, Pages: 788-+, ISSN: 1058-6393
Naylor PA, Lin XS, Khong AWH, 2008, Near-common zeros in blind identification of SIMO acoustic systems, Workshop on Hands-Free Speech Communication and Microphone Arrays, Publisher: IEEE, Pages: 22-25
Khong AWH, Lin X, Naylor PA, 2008, Algorithms for identifying clusters of near-common zeros in multichannel blind system identification and equalization, Pages: 389-392-389-392
Blind system identification (BSI) and equalization algorithms have been applied to multichannel systems with high order such as found in acoustic impulse responses. Studies on the performance of such algorithms in the presence of near-common zeros ha.....
Wen Y-CJ, Habets E, Naylor PA, 2008, Blind estimation of reverberation time based on the distribution of signal decay rates, Pages: 329-332-329-332
The reverberation time is one of the most prominent acoustic characteristics of an enclosure. Its value can be used to predict speech intelligibility, and is used by speech enhancement techniques to suppress reverberation. The reverberation time is u.....
Khong AWH, Lin X, Doroslovacki M, et al., 2008, Frequency domain selective tap adaptive algorithms for sparse system identification, Pages: 229-232-229-232
We propose a new low complexity and fast converging frequency-domain adaptive algorithm for sparse system identification. This is achieved by exploiting the MMax and SP tap-selection criteria for complexity reduction and fast convergence respectively.....
Khong AWH, Gan W-S, Naylor PA, et al., 2008, A low complexity fast converging partial update adaptive algorithm employing variable step-size for acoustic echo cancellation, Pages: 237-240-237-240
Partial update adaptive algorithms have been proposed as a means of reducing complexity for adaptive filtering. The MMax tap-selection is one of the most popular tap-selection algorithms. It is well known that the performance of such partial update a.....
Habets E, Gaubitch ND, Naylor PA, 2008, Temporal selective dereverberation of noisy speech using one microphone, Pages: 4577-4580-4577-4580
Reverberant speech can be described as sounding distant with noticeable coloration and echo. These detrimental perceptual effects are caused by early and late reflections, respectively, and reduces the fidelity and intelligibility of speech. It is we.....
Zhang W, Gaubitch ND, Naylor PA, 2008, Computationally efficient equalization of room impulse responses robust to system estimation errors, Pages: 4025-4028-4025-4028
Equalization techniques for room impulse responses (RIRs) are important in acoustic signal processing applications such as speech dereverberation. In practice, only approximate estimates of the RIRs are available and the inverse filters designed from.....
Loganathan P, Khong WHA, Naylor PA, 2008, A Sparseness Controlled Proportionate Algorithm for Acoustic Echo Cancellation
X Lin MD, Naylor PA, 2008, Frequency Domain Adaptive Algorithm for Network Echo Cancellation in VoIP, EURASIP Journal on Audio, Speech, and Music Processing
Gaubitch ND, Lin X, Naylor PA, 2008, Scale Factor Ambiguity Correction for Subband Blind Multichannel Identification
Zhang W, Naylor PA, 2008, An Algorithm to Generate Representations of System Identification Errors, Research Letters in Signal Processing
Thomas MRP, Naylor PA, 2008, The SIGMA Algorithm for Estimation of Reference-Quality Glottal Closure Instants from Electroglottograph Signals
Thomas MRP, Gudnason J, Naylor PA, 2008, Application of the DYPSA Algorithm to Segmented Time Scale Modification of Speech
Sehr A, Wen Y-CJ, Kellermann W, et al., 2008, A Combined Approach for Estimating a Feature-Domain Reverberation Model in Non-diffuse Environments
Maqsood H, Naylor P, 2007, Improved DYPSA Algorithm for Noise and Unvoiced Speech, Pages: 243-248-243-248
The DYPSA algorithm detects glottal closure instants (GCI) in speech signals. We present an improvement in the algorithm in which a voiced/unvoiced/silence discrimination measure is applied in order to reduce the spurious GCIs detected incorrectly fo.....
Wen Y-CJ, Naylor PA, 2007, Objective Measurement of Colouration in Reverberation
Khong AWH, Naylor PA, 2007, Selective-Tap Adaptive Filtering With Performance Analysis for Identification of Time-Varying Systems, IEEE Trans Audio Speech and Language, Vol: 15, Pages: 1681-1695
Lin X, Gaubitch ND, Naylor PA, 2007, Blind Speech Dereverberation in the Presence of Common Acoustic Zeros
Khong WHA, Naylor PA, 2007, Selective-tap Adaptive Filtering with Performance Analysis for Non-stationary System Identification, IEEE Transactions on Audio, Speech and Language Processing, Vol: 15, Pages: 1681-1695
Gaubitch N D, Naylor P A, 2007, Spatiotemporal Averaging Method for Enhancement of Reverberant Speech
Ahmad R, Khong, W H A, et al., 2007, A Practical Adaptive Blind Multichannel Estimation Algorithm with Application to Acoustic Impulse Responses
Naylor P A, Khong, W H A, et al., 2007, Misalignment Performance Of Selective Tap Adaptive Algorithms For System Identification Of Time-Varying Unknown Systems
Ariful Haque M, Shafi Bashar M, Naylor P A, et al., 2007, Energy Constrained Frequency-Domain Normalized LMS Algorithm for Blind Channel Identification, Signal Image and Video Processing, Vol: 1, Pages: 203-213
Khong, W H A, Naylor P A, et al., 2007, A Low Delay and Fast Converging Improved Proportionate Algorithm for Sparse System Identification, EURASIP Journal of Audio, Speech and Music Processing
Dogancay K, Naylor PA, 2007, Adaptive Partial-Update and Sparse System Identification, EURASIP Journal on Audio, Speech, and Music Processing, Vol: 2007
Thomas MRP, Gaubitch ND, Gudnason J, et al., 2007, A practical multichannel dereverberation algorithm using Multichannel DYPSA and spatiotemporal averaging, IEEE Workshop on Applications of Signal Processing to Audio and Acoutics, Publisher: IEEE, Pages: 41-44
Wen JYC, Naylor PA, 2007, Semantic colouration space investigation: Controlled colouration in the bark-sone domain, IEEE Workshop on Applications of Signal Processing to Audio and Acoutics, Publisher: IEEE, Pages: 89-92
Gaubitch ND, Thomas MRP, Naylor PA, 2007, Subband method for multichannel least squares equalization of room transfer functions, IEEE Workshop on Applications of Signal Processing to Audio and Acoutics, Publisher: IEEE, Pages: 33-36
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