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Conference paperGaubitch ND, Brookes M, Naylor PA, 2009,
Blind Channel Identification in Speech using the Long-term Average Speech Spectrum
Conference paperGudnason J, Thomas MRP, Naylor PA, et al., 2009,
Voice Source Waveform Analysis and Synthesis using Principal Component Analysis and Gaussian Mixture Modelling, 10th INTERSPEECH 2009 Conference, Publisher: ISCA-INT SPEECH COMMUNICATION ASSOC, Pages: 120-+
Journal articleThomas MRP, Naylor PA, 2009,
Conference paperLin X, Khong AWH, Naylor PA, 2009,
Conference paperTsakiris MC, Naylor PA, 2009,
FAST EXACT AFFINE PROJECTION ALGORITHM USING DISPLACEMENT STRUCTURE THEORY, 16th International Conference on Digital Signal Processing, Publisher: IEEE, Pages: 69-74
Journal articleManmontri U, Naylor PA, 2008,
A Class of Frobenius Norm-Based Algorithms Using Penalty Term and Natural Gradient for Blind Signal Separation, IEEE Trans. Audio, Speech, and Language Processing, Vol: 16, Pages: 1181-1193-1181-1193
We consider the blind signal separation (BSS) problem of instantaneous mixtures using penalty term and natural gradient. A class of Frobenius norm-based algorithms consisting of the offline/block processing (BP), online processing (OP) algorithms, an.....
Conference paperGaubitch ND, Habets E, Naylor PA, 2008,
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.....
Conference paperKhong 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.....
Conference paperWen Y-CJ, Habets E, Naylor PA, 2008,
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.....
Conference paperKhong AWH, Lin X, Doroslovacki M, et al., 2008,
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.....
Conference paperKhong 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.....
Conference paperHabets E, Gaubitch ND, Naylor PA, 2008,
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.....
Conference paperZhang W, Gaubitch ND, Naylor PA, 2008,
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.....
Conference paperThomas MRP, Naylor PA, 2008,
The SIGMA Algorithm for Estimation of Reference-Quality Glottal Closure Instants from Electroglottograph Signals
Journal articleZhang W, Naylor PA, 2008,
Journal articleX Lin MD, Naylor PA, 2008,
Conference paperNaylor 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
Conference paperGaubitch ND, Lin X, Naylor PA, 2008,
Scale Factor Ambiguity Correction for Subband Blind Multichannel Identification
Conference paperThomas MRP, Gudnason J, Naylor PA, 2008,
Application of the DYPSA Algorithm to Segmented Time Scale Modification of Speech
Conference paperSehr A, Wen Y-CJ, Kellermann W, et al., 2008,
A Combined Approach for Estimating a Feature-Domain Reverberation Model in Non-diffuse Environments
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