Expert profile: Edmilson Morais [ Da Silva ]

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Description: Expert in Speech Recognition, Speech Synthesis and Speech
   Coding

  Edmilson S. Morais
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   The portuguese version of this page is been prepared.
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   (Additional information can be found on my homepage.)

  Academic Background

   State University of Campinas - UNICAMP - SP - Brazil
     + PHD in Eletrical Engenieering (Not completed, yet) August 1997 -
       August 1998 (Not completed)
     + Master Degree in Eletrical Engenieering January 1996 - December
       1997

   University of Brasília - UnB - DF - Brazil
     + Undergraduate in Eletrical Engenieering January 1991 - December
       1995

  Professional Experience

   CSTR (Centre for Speech Technology Research) / University of Edinburgh
   Edinburgh. Scotland, UK January 2000 - July 2000
     + Visiting Researcher : Prosodic Speech Modification Using Prototype
       Waveform Interpolation (A Time / Frequence Approach)

   IBM - International Bussines Machine
   Seville, Spain (European Speech Recognition Group) September 1998 -
   December 1999
     + Project Supervisor : Acoustic and Language modeling for the
       Brazilian Portuguese IBM ViaVoice System. These project was
       developed by team composed by : Eletrical Engineers, Computer
       Engineers, Telecommunication Engineers and Linguistics.

   Z-Tecnology in Communications July 1994 - January 1995
   Brasília - DF - Brazil
     + Intern : Development of application for MODEM based on C++ for
       windows.

   Z-Tecnology in Communications May 1992 - August 1993
   Brasília - DF - Brazil
     + Intern : Development of software in C, Assembly for PC and
       Assembly for TMS320C51 from Texas Instruments.



  Research Activities

   GPDF - Group of Speech Digital Signal Processing of UNICAMP January
   1996 - September 1998
   Campinas - SP - Brazil
     + Master Degree : Application of adaptative algorithm LMS ("Least
       Mean Square") and RLS ("Recursive Least Square") for speech signal
       analysis. Applications of the FIR-MLP ("Finite Impulse Response -
       Multilayer Perceptron") for phones analysis and phones
       recognition. Analysis and Development of all modules of a Hybrid
       Continuous Speech Recognition System based on ANN (Artificial
       Neural Network) and HMM (Hidden Markov Model). Study and
       development of algorithm for TTS-Sythesis
       (Text-To-Speech-Synthesis) based on PSOLA, TFI (Time Frequence
       Interpolation) and Harmonic Decomposition. Obs : During all my
       work in the GPDF-UNICAMP I mantained a strong relationship with
       the LAFAPE-UNICAMP (Laboratory of Acoustic Phonetic and
       Experimental Psychoacoustc Group of UNICAMP).

   GPDS - Group of Digital Signal Processing of UnB August 1993 -
   December 1995
   Campinas - SP - Brazil
     + Reserch study : Study of fundamental concepts of digital signal
       processing. Research and development of techniques for speech
       signal processing , such as : (1) Prosodic speech modification
       using a technique based on CELP/TFI (Coding excited linear
       prediction / Time frequency interpolation), (2) Low bit rate
       speech coding based on a CELP/TFI system, (3) Development of a TTS
       system based on the CELP/TFI technique.

   Program Languages
     + Matlab
     + C and C++
     + Shell scripts for Unix

   Teaching Experience
     + Diferencial and Integral Calculus (Teacher assistent) March 1992 -
       July 1992
     + Digital Signal Processing (Teacher assistent) March 1995 - July
       1995

   Language Skills
     + Portuguese Native Language.
     + English Proficiency Level .
     + Spanish Intermediate Level : Spoken and Written.

   Scientific Production
     + Morais, E., Taylor, P., Prototype Waverform Interpolation Applied
       for Concatenative Speech Synthesis ( A Time Frequency Approach) ,
       Submitted for ICSLP - International Congress of Speech and
       Language Processing , Beijin, China, October 2000.
     + Morais, E., Violaro F., Prosodic Speech Modification Using TFI
       (Time Frequence Interpolation), International Telecommunication
       Symposium, São Paulo, SP, Brazil, June 1998.
     + Ynoguti, C., Morais, E, S. Violaro F., A comparative Study Between
       a Hybrid System and a Traditional HMM System for Continuous Speech
       Recognition, International Telecommuication Symposium, São Paulo,
       SP, Brazil, June 1998.
     + Morais, E., Violaro F., Automatic Continuous Speech Recognition
       Based on ANN-HMM Hibryd System, Master Degree Thesis, UNICAMP,
       Campinas - SP, Brazil, 1997.
     + Morais, E., Violaro F., Netto M. L. A., Hidden Markov Model
       Trained by a ANN and Its Application for Speech Recognition, III
       CBRN - Brazilian Congress of Artificial Neural Networks,
       Florianópolis - SC - Brazil, 1997.
     + Morais, E., Violaro F., Netto M. L. A., Hybrid Hidden Markov Model
       / Neural Network System Following MAP and ML Aproaches and their
       Application to Time Series Recognition , III SBAI - Brazilian
       Simposium of Inteligent Control , Espirito Santo - ES - Brazil,
       1999
     + Morais, E., Violaro F., Hybrid System ANN-HMM for Continuous
       Speech Recognition, XV SBT - Brazilian Telecommunication
       Symposium, Recife - PE - Brazil, 1997.
     + Morais, E., Silva L. M., A Comparative Study of Prototype
       Transmission Rate for TFI Speech Coding, TELEMO 96 - XIV Brazilian
       Telecommunication Symposium and VII Brazilian Optic and Microwave
       Symposium, Curitiba - PR - Brazil, 1996.
     + Morais, E., Silva, L, M., Cobra, D, T, Q., System CELP/TFI for
       Speech Coding and Its Application for Text To Speech Synthesis.,
       Final Project of Undergraduation in Eletrical Engenieering,
       University of Brasília, Brasília - DF - Brazil, 1995.
     + Morais, E., Cobra, D, T, Q., Prosodic Speech Modifications (Pitch
       and Time Scale), Anais of the First Congress of Science and
       Culture of the University of Brasília - DF - Brazil, 1995.