Thippur V. Sreenivas

T. V. Sreenivas graduated from Bangalore University in 1973, obtained M.E. from Indian Institute of Science(IISc), Bangalore, in 1975 and Ph. D. degree from Indian Institute of Technology, Bombay, India, in 1981, working as Research Scholar at Tata Institute of Fundamental Research, Bombay. During 19821985, he worked with LRDE, Bangalore, in the area of low bitrate speech coding. During 19861987, he worked with Norwegian Institute of Technology, Trondheim, Norway, developing new techniques for speech coding and speech recognition. During 19881989, he was Visiting Assistant Professor at Marquette University, Milwaukee, USA, teaching and researching in speech enhancement and spectral estimation. Since 1990, he has joined the faculty of IISc, Bangalore, where he is currently Associate Professor. At IISc, he leads the activity of Speech and Audio Group. His research is focussed on auditory spectral estimation, speech/audio modeling and novel algorithms for speech/audio compression, recognition and enhancement. He is also a Faculty Entrepreneur and has jointly founded “Esqube Communication Solutions Pvt. Ltd.,” a startup company in Bangalore. He has been a Visiting Faculty at Fraunhofer Institute for Integrated Circuits, Erlangen, Germany and Griffith University, Australia. He is a Senior Member of IEEE and currently Chairman of IEEE Signal Processing Society, Bangalore Chapter.

Biography Updated on 2 March 2004

Articles in Scholarly Journals [Incomplete List]

  1. Conditional PDF-Based Split Vector Quantization of Wideband LSF Parameters
    IEEE Signal Processing Letters, vol. 14, no. 9, pp. 641–644, 2007
  2. Analysis of Conditional PDF-Based Split VQ
    IEEE Signal Processing Letters, vol. 14, no. 11, pp. 781–784, 2007
  3. Signal-to-noise ratio estimation using higher-order moments
    Signal Processing, vol. 86, no. 4, pp. 716–732, 2006
  4. Time-varying filter interpretation of Fourier transform and its variants
    Signal Processing, vol. 86, no. 11, pp. 3258–3263, 2006
  5. Increased watermark-to-host correlation of uniform random phase watermarks in audio signals
    Signal Processing, 2006
  6. Auditory Motivated Level-Crossing Approach to Instantaneous Frequency Estimation
    IEEE Transactions on Signal Processing, vol. 53, no. 4, pp. 1450–1462, 2005
  7. Adaptive Window Zero-Crossing-Based Instantaneous Frequency Estimation
    EURASIP Journal on Applied Signal Processing, vol. 2004, no. 12, pp. 1791–1806, 2004
  8. Effect of interpolation on PWVD computation and instantaneous frequency estimation
    Signal Processing, vol. 84, no. 1, pp. 107–116, 2004
  9. Adaptive spectrogram vs. adaptive pseudo-Wigner–Ville distribution for instantaneous frequency estimation
    Signal Processing, vol. 83, no. 7, pp. 1529–1543, 2003
  10. IF estimation using higher order TFRs
    Signal Processing, vol. 82, no. 2, pp. 127–132, 2002
  11. Cone-kernel representation versus instantaneous power spectrum
    IEEE Transactions on Signal Processing, vol. 47, no. 1, pp. 250–254, 1999
  12. Incorporating phonetic properties in hidden Markov models for speech recognition
    The Journal of the Acoustical Society of America, vol. 102, no. 2, p. 1149, 1997
  13. Codebook constrained Wiener filtering for speech enhancement
    IEEE Transactions on Speech and Audio Processing, vol. 4, no. 5, pp. 383–389, 1996
  14. Zero-crossing based spectral analysis and SVD spectral analysis for formant frequency estimation in noise
    IEEE Transactions on Signal Processing, vol. 40, no. 2, pp. 282–293, 1992
  15. On designing a microprogram translator
    Signal Processing, vol. 13, no. 1, pp. 91–100, 1987
  16. Simulation of a programmable signal processor
    Signal Processing, vol. 6, no. 2, pp. 135–142, 1984
  17. Functional demarcation of pitch
    Signal Processing, vol. 3, no. 3, pp. 277–284, 1981
  18. High-resolution narrow-band spectra by FFT pruning
    IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. 28, no. 2, pp. 254–257, 1980
  19. Pitch extraction from corrupted harmonics of the power spectrum
    The Journal of the Acoustical Society of America, vol. 65, no. 1, p. 223, 1979
  20. FFT algorithm for both input and output pruning
    IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. 27, no. 3, pp. 291–292, 1979