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Conference paperThomas MRP, Gudnason J, Naylor PA, 2009,
Data-driven voice soruce waveform modelling
, Pages: 3965-3968-3965-3968 -
Conference paperGaubitch ND, Brookes M, Naylor PA, 2009,
Blind Channel Identification in Speech using the Long-term Average Speech Spectrum
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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,
The SIGMA Algorithm: A Glottal Activity Detector for Electroglottographic Signals
, IEEE Trans. Audio Speech and Language Processing, Vol: 17, Pages: 1557-1566-1557-1566 -
Conference paperLin X, Khong AWH, Naylor PA, 2009,
Blind system identification for speech dereverberation with Forced Spectral Diversity
, Pages: 3737-3740-3737-3740 -
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- Author Web Link
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- Citations: 4
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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: 1181-1193-1181-1193We 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.....
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Conference paperGaubitch ND, Habets E, Naylor PA, 2008,
Multimicrophone speech dereverberation using spatiotemporal and spectral processing
, Pages: 3222-3225-3222-3225Speech 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.....
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Conference paperKhong AWH, Lin X, Naylor PA, 2008,
Algorithms for identifying clusters of near-common zeros in multichannel blind system identification and equalization
, Pages: 389-392-389-392Blind 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.....
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Conference paperWen Y-CJ, Habets E, Naylor PA, 2008,
Blind estimation of reverberation time based on the distribution of signal decay rates
, Pages: 329-332-329-332The 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.....
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Conference paperKhong AWH, Lin X, Doroslovacki M, et al., 2008,
Frequency domain selective tap adaptive algorithms for sparse system identification
, Pages: 229-232-229-232We 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.....
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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: 237-240-237-240Partial 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.....
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Conference paperHabets E, Gaubitch ND, Naylor PA, 2008,
Temporal selective dereverberation of noisy speech using one microphone
, Pages: 4577-4580-4577-4580Reverberant 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.....
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Conference paperZhang W, Gaubitch ND, Naylor PA, 2008,
Computationally efficient equalization of room impulse responses robust to system estimation errors
, Pages: 4025-4028-4025-4028Equalization 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.....
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Conference paperThomas MRP, Naylor PA, 2008,
The SIGMA Algorithm for Estimation of Reference-Quality Glottal Closure Instants from Electroglottograph Signals
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Journal articleZhang W, Naylor PA, 2008,
An Algorithm to Generate Representations of System Identification Errors
, Research Letters in Signal Processing -
Journal articleX Lin MD, Naylor PA, 2008,
Frequency Domain Adaptive Algorithm for Network Echo Cancellation in VoIP
, EURASIP Journal on Audio, Speech, and Music Processing -
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- Author Web Link
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- Citations: 4
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Conference paperGaubitch ND, Lin X, Naylor PA, 2008,
Scale Factor Ambiguity Correction for Subband Blind Multichannel Identification
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Conference paperThomas MRP, Gudnason J, Naylor PA, 2008,
Application of the DYPSA Algorithm to Segmented Time Scale Modification of Speech
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