Estimation of the instantaneous harmonic parameters of speech

This paper describes a method of accurate estimation of the instantaneous speech signal harmonic parameters. The method is based on adaptive filtering of the speech signal along its harmonic components. A simple way of filter synthesis based on the Fourier transform is also proposed. The synthesized filters have a closed form impulse response which can be modulated in frequency domain to achieve better performance for components with high frequency alteration. This method is also applicable to give an accurate estimate of the fundamental frequency of speech.