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Conference paperThomas MRP, Gudnason J, Naylor PA, 2009, , Pages: 3965-3968-3965-3968
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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, , IEEE Trans. Audio Speech and Language Processing, Vol: 17, Pages: 1557-1566-1557-1566
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Conference paperLin X, Khong AWH, Naylor PA, 2009, , Pages: 3737-3740-3737-3740
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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- Cite
- Citations: 4
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Journal articleManmontri U, Naylor PA, 2008, , 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.....
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Conference paperGaubitch ND, Habets E, Naylor PA, 2008, , 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.....
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Conference paperKhong AWH, Lin X, Naylor PA, 2008, , 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.....
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Conference paperWen Y-CJ, Habets E, Naylor PA, 2008, , 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.....
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Conference paperKhong AWH, Lin X, Doroslovacki M, et al., 2008, , 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.....
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Conference paperKhong AWH, Gan W-S, Naylor PA, et al., 2008, , 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.....
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Conference paperHabets E, Gaubitch ND, Naylor PA, 2008, , 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.....
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Conference paperZhang W, Gaubitch ND, Naylor PA, 2008, , 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.....
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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, , Research Letters in Signal Processing
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Journal articleX Lin MD, Naylor PA, 2008, , EURASIP Journal on Audio, Speech, and Music Processing
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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- Cite
- 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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