A kepstrum approach to real-time speech enhancement : thesis for the degree of Doctor of Philosophy, Information Engineering, Institute of Technology and Engineering, Massey University at Albany

Content removed due to copyright: Conference proceedings (I) J. Jeong, and T.J. Moir, "Kepstrum approach to real-time speech enhancement methods using two microphones", Proceedings of the International Conference on Sensing Technology (ICST), pp 691-695, November 21-23, 2005, Palmerston North, New Zealand Conference proceedings (II) J. Jeong and T. J. Moir, "Two-microphone kepstrum approach to real-time speech enhancement methods" Proceedings of the IEEE International Conference on Engineering of Intelligent Systems (ICEIS), pp 392-397, April 22-23, 2006, Islamabad, Pakistan Conference proceedings (III) T. J. Moir and J. Jeong, "Identification of non-minimum phase transfer function components" Proceedings of the IEEE International Symposium on Signal Processing and Information Technology (ISSPIT), pp 380-384, August 27-30, 2006, Vancouver, Canada

[1]  John Homer,et al.  Detection guided NLMS estimation of sparsely parametrized channels , 2000 .

[2]  Philipos C. Loizou,et al.  A multi-band spectral subtraction method for enhancing speech corrupted by colored noise , 2002, 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[3]  Behrouz Farhang-Boroujeny,et al.  Robust microphone arrays using subband adaptive filters , 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).

[4]  Yi Hu,et al.  A subspace approach for enhancing speech corrupted by colored noise , 2002, IEEE Signal Processing Letters.

[5]  Dirk Van Compernolle,et al.  Switching adaptive filters for enhancing noisy and reverberant speech from microphone array recordings , 1990, ICASSP.

[6]  M. S. Babtlett Smoothing Periodograms from Time-Series with Continuous Spectra , 1948, Nature.

[7]  Paul W. Shields,et al.  Speech enhancement using sub-band adaptive Griffiths-Jim signal processing , 2003, Speech Commun..

[8]  Charles M. Rader,et al.  Digital processing of signals , 1983 .

[9]  John W. Fay Confidence bounds for signal-to-noise ratios from magnitude-squared coherence estimates , 1980, ICASSP.

[10]  Israel Cohen,et al.  An Integrated Real-Time Beamforming and Postfiltering System for Nonstationary Noise Environments , 2003, EURASIP J. Adv. Signal Process..

[11]  Petre Stoica,et al.  Maximum likelihood methods for direction-of-arrival estimation , 1990, IEEE Trans. Acoust. Speech Signal Process..

[12]  D. J. Campbell,et al.  Sub-band adaptive filtering applied to speech enhancement , 1996, Proceeding of Fourth International Conference on Spoken Language Processing. ICSLP '96.

[13]  Boaz Rafaely,et al.  Frequency-domain adaptation of causal digital filters , 2000, IEEE Trans. Signal Process..

[14]  M Kompis,et al.  Noise reduction for hearing aids: combining directional microphones with an adaptive beamformer. , 1994, The Journal of the Acoustical Society of America.

[15]  Lucas C. Parra,et al.  Convolutive blind separation of non-stationary sources , 2000, IEEE Trans. Speech Audio Process..

[16]  J M Kates,et al.  A comparison of hearing-aid array processing techniques. , 1996, The Journal of the Acoustical Society of America.

[17]  J. W. Tukey,et al.  The Measurement of Power Spectra from the Point of View of Communications Engineering , 1958 .

[18]  Masato Miyoshi,et al.  Inverse filtering of room acoustics , 1988, IEEE Trans. Acoust. Speech Signal Process..

[19]  Teng Joon Lim,et al.  Adaptive allpass filtering for nonminimum-phase system identification , 1994 .

[20]  Jont B. Allen,et al.  Invertibility of a room impulse response , 1979 .

[21]  Tom J. Moir,et al.  The kepstrum method for spectral analysis , 1986 .

[22]  S. Boll,et al.  Suppression of acoustic noise in speech using two microphone adaptive noise cancellation , 1980 .

[23]  L. Auger The Journal of the Acoustical Society of America , 1949 .

[24]  Tom J. Moir,et al.  A unified approach to multivariable discrete-time filtering based on the Wiener theory , 1987, Kybernetika.

[25]  Fred C. Schweppe,et al.  Sensor-array data processing for multiple-signal sources , 1968, IEEE Trans. Inf. Theory.

[26]  R. Zelinski Noise reduction based on microphone array with LMS adaptive post-filtering , 1990 .

[27]  Gérard Faucon,et al.  Proposal of a voice activity detector for noise reduction , 1994 .

[28]  Dirk Van Compernolle,et al.  Signal separation by symmetric adaptive decorrelation: stability, convergence, and uniqueness , 1995, IEEE Trans. Signal Process..

[29]  Hanseok Ko,et al.  Speech enhancement for robust speech recognition in car environments using Griffiths-Jim ANC based on two-paired microphones , 2004, IEEE International Symposium on Consumer Electronics, 2004.

[30]  Sailes K. Sengijpta Fundamentals of Statistical Signal Processing: Estimation Theory , 1995 .

[31]  L. C. Wood,et al.  Seismic signal processing , 1975, Proceedings of the IEEE.

[32]  J. Tukey,et al.  An algorithm for the machine calculation of complex Fourier series , 1965 .

[33]  G. Carter,et al.  Estimation of the magnitude-squared coherence function via overlapped fast Fourier transform processing , 1973 .

[34]  Lucas C. Parra,et al.  The generalized sidelobe decorrelator , 2001, Proceedings of the 2001 IEEE Workshop on the Applications of Signal Processing to Audio and Acoustics (Cat. No.01TH8575).

[35]  Robert H. Baran,et al.  Dual channel based speech enhancement using novelty filter for robust speech recognition in automobile environment , 2006, IEEE Transactions on Consumer Electronics.

[36]  Sven Fischer,et al.  Beamforming microphone arrays for speech acquisition in noisy environments , 1996, Speech Commun..

[37]  Jörg Meyer,et al.  Multi-channel speech enhancement in a car environment using Wiener filtering and spectral subtraction , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.