Andreas Spanias

A. Spanias is a Professor of electrical engineering at Fulton School of Engineering, Arizona State University. His research interests are in adaptive signal processing and speech processing. He received the 2003 Teaching Award from the IEEE Phoenix Section for the development of J-DSP. He is a member of the IEEE-CAS Society DSP Technical Committee and has served as a Member in the Technical Committee on Statistical Signal and Array Processing of the IEEE Signal Processing Society (SPS). He has served as an Associate Editor of the IEEE Transactions on Signal Processing, General Cochair of the 1999 International Conference on Acoustics Speech and Signal Processing (Phoenix), IEEE Signal Processing Vice President for Conferences, and Chair of the Conference Board. He served as a Member in the IEEE Signal Processing Executive Committee and as an Associate Editor of IEEE Signal Processing Letters. He is currently serving as a Member in the IEEE SPS Publications Board, and Memberat- Large of the IEEE SPS Conference Board. He has been Chair of the Phoenix IEEE Communications and Signal Processing Chapter, and is a Member in Eta Kappa Nu and Sigma Xi. Andreas Spanias is corecipient of the 2002 IEEE Donald G. FinkPaper Award, and was recently elected as a Fellow of the IEEE. He is appointed as 2004 Distinguished Lecturer of the IEEE SPS.

Biography Updated on 15 September 2003

Articles in Scholarly Journals [Incomplete List]

  1. Wideband Speech Recovery Using Psychoacoustic Criteria
    EURASIP Journal on Audio, Speech, and Music Processing, vol. 2007, Article ID 16816, 18 pages, 2007
  2. Performance of Precoded OFDM With Channel Estimation Error
    IEEE Transactions on Signal Processing, vol. 54, no. 3, pp. 1165–1171, 2006
  3. A multi-dimensional scheme for controlling unstable periodic orbits in chaotic systems
    Physics Letters A, vol. 349, no. 1-4, pp. 116–127, 2006
  4. Interactive Online Undergraduate Laboratories Using J-DSP
    IEEE Transactions on Education, vol. 48, no. 4, pp. 735–749, 2005
  5. Perceptual Segmentation and Component Selection for Sinusoidal Representations of Audio
    IEEE Transactions on Speech and Audio Processing, vol. 13, no. 2, pp. 149–162, 2005
  6. Autoregressive Modeling and Feature Analysis of DNA Sequences
    EURASIP Journal on Applied Signal Processing, vol. 2004, no. 1, pp. 13–28, 2004
  7. Measuring the Direction and the Strength of Coupling in Nonlinear Systems—A Modeling Approach in the State Space
    IEEE Signal Processing Letters, vol. 11, no. 7, pp. 617–620, 2004
  8. On-Line Signal Processing Using J-DSP
    IEEE Signal Processing Letters, vol. 11, no. 10, pp. 821–825, 2004
  9. Fast adaptive algorithms using eigenspace projections
    Signal Processing, vol. 83, no. 9, pp. 1929–1935, 2003
  10. Adaptive eigen-projection beamforming algorithms for 1-D and 2-D antenna arrays
    Antennas and Wireless Propagation Letters, vol. 2, no. 2, pp. 62–65, 2003
  11. Sinusoidal Analysis-Synthesis of Audio Using Perceptual Criteria
    EURASIP Journal on Applied Signal Processing, vol. 2003, no. 1, pp. 15–20, 2003
  12. Smart antenna system analysis, integration and performance for mobile ad-hoc networks (MANETs)
    IEEE Transactions on Antennas and Propagation, vol. 50, no. 5, pp. 571–581, 2002
  13. Smart-antenna systems for mobile communication networks. Part 1. Overview and antenna design
    IEEE Antennas and Propagation Magazine, vol. 44, no. 3, pp. 145–154, 2002
  14. Smart-antenna system for mobile communication networks .2. Beamforming and network throughput
    IEEE Antennas and Propagation Magazine, vol. 44, no. 4, pp. 106–114, 2002
  15. Adaptive modified covariance algorithms for spectral analysis
    Signal Processing, vol. 82, no. 5, pp. 715–720, 2002
  16. Low bit-rate speech coding based on an improved sinusoidal model
    Speech Communication, vol. 34, no. 4, pp. 369–390, 2001
  17. An improved approach to robust speech recognition using minimum error classification
    Speech Communication, vol. 30, no. 1, pp. 27–36, 2000
  18. Perceptual coding of digital audio
    Proceedings of the IEEE, vol. 88, no. 4, pp. 451–515, 2000
  19. Cepstrum-based pitch detection using a new statistical V/UV classification algorithm
    IEEE Transactions on Speech and Audio Processing, vol. 7, no. 3, pp. 333–338, 1999
  20. Improved speech recognition using a subspace projection approach
    IEEE Transactions on Speech and Audio Processing, vol. 7, no. 3, pp. 343–345, 1999
  21. A new phase model for sinusoidal transform coding of speech
    IEEE Transactions on Speech and Audio Processing, vol. 6, no. 5, pp. 495–501, 1998
  22. Improving discrimination of confusable words using the divergence measure
    The Journal of the Acoustical Society of America, vol. 101, no. 2, p. 1106, 1997
  23. Speech enhancement using state-based estimation and sinusoidal modeling
    The Journal of the Acoustical Society of America, vol. 102, no. 2, p. 1141, 1997
  24. High-performance alphabet recognition
    IEEE Transactions on Speech and Audio Processing, vol. 4, no. 6, pp. 430–445, 1996
  25. A software tool for introducing speech coding fundamentals in a DSP course
    IEEE Transactions on Education, vol. 39, no. 2, pp. 143–152, 1996
  26. System identification based on bounded error constraints
    IEEE Transactions on Signal Processing, vol. 43, no. 12, pp. 3071–3075, 1995
  27. Automatic recognition of syllable-final nasals preceded by /e/
    The Journal of the Acoustical Society of America, vol. 97, no. 3, p. 1925, 1995
  28. Practical considerations in the implementation of a frequency-domain adaptive noise canceller
    IEEE Transactions on Circuits and Systems II: Analog and Digital Signal Processing, vol. 41, no. 2, pp. 164–168, 1994
  29. Speech coding: a tutorial review
    Proceedings of the IEEE, vol. 82, no. 10, pp. 1541–1582, 1994
  30. Block time and frequency domain modified covariance algorithms for spectral analysis
    IEEE Transactions on Signal Processing, vol. 41, no. 11, pp. 3138–3152, 1993
  31. A frequency selective adaptive algorithm
    Computers & Electrical Engineering, vol. 18, no. 3-4, pp. 301–313, 1992
  32. A hybrid transform method for analysis/synthesis of speech
    Signal Processing, vol. 24, no. 2, pp. 217–229, 1991
  33. Transform methods for seismic data compression
    IEEE Transactions on Geoscience and Remote Sensing, vol. 29, no. 3, pp. 407–416, 1991
  34. Accurate representation of time-varying signals using mixed transforms with applications to speech
    IEEE Transactions on Circuits and Systems, vol. 36, no. 2, pp. 329–331, 1989
  35. Efficient modeling of dominant transform components representing time-varying signals
    IEEE Transactions on Circuits and Systems, vol. 36, no. 2, pp. 331–334, 1989
  36. A fast frequency-domain adaptive algorithm
    Proceedings of the IEEE, vol. 76, no. 1, pp. 80–82, 1988
  37. Comparison of several frequency-domain LMS algorithms
    IEEE Transactions on Circuits and Systems, vol. 34, no. 5, pp. 586–588, 1987
  38. A two-stage pole-zero predictor
    IEEE Transactions on Circuits and Systems, vol. 33, no. 3, pp. 352–354, 1986