Speech analysis/synthesis/conversion by using sequential processing

نویسندگان

  • Panuthat Boonpramuk
  • Tetsuo Funada
  • Noboru Kanedera
چکیده

This paper presents a method for speech analysis/synthesis/ conversion by using sequential processing. The aims of this method are to improve the quality of synthesized speech and to convert the original speech into another speech of different characteristics. We apply the Kalman Filter for estimating the auto-regressive coefficients of ‘time varying AR model with unknown input (ARUI model)’, which we have proposed to improve the conventional AR model, and we use a band-pass filter for making ‘a guide signal’ to extract the pitch period from the residual signal. These signals are utilized to make the driving source signal in speech synthesis. We also use the guide signal for speech conversion, such as in pitch and utterance length. Moreover, we show experimentally that this method can analyze/synthesize/convert speech without causing instability by using the smoothed auto-regressive coefficients.

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تاریخ انتشار 1999