A mathematical model for the determination of speech signal parameters with the aid of instant memory
The system enables the analysis of sounds in real time, as well as the automatic segmentation of sound as a function of time. The accuracy of the identification of speech does not depend on the frequency of the larynx tone or on the speech rate, etc. With the system it is possible to use bandpass filters with comparatively broad transmission band widths. A formal description of the system by means of characteristic functions of the features allows direct design and construct of the system by means of generally available integrated circuits.
References
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Z. M. WÓJCIK, Conception of instant memory in the identification of sounds, Works by IBIB-PAN, Warszawa 1975 [in Polish].