U.S. Philips Corporation

N.v. Philips' Gloeilampenfabrieken

Technische Hogeschool Eindhoven

Koninklijke Philips Corporation N.v.

Leonardus F Willems

Leonardus L Vogten

Chang X Ma

Robert J Sluyter

Hendrikus Duifhuis

In the natural production of human speech, the instant of closure of the vocal cords occurs usually at well defined instants. These instants are used for speech processing, such as glottal synchronous processing or speech synthesis with observed natural vocal cord excitation signals. To detect the instants of glottal closure from an observed speech signal, the observed speech signal is high pass filtered, and a temporally localized aggregate of the number and amplitudes of peaks in the high pass filtered signal is determined for possible instants of glottal closure. The instants of glottal closure are determined as instants where the aggregate takes maximal values.

Human speech processing apparatus for detecting instants of glottal closure

A method of and a device for deriving formant frequencies from a part of a speech signal. For determining the formant frequencies from a part of a speech signal located within a given time interval, the Split Levinson algorithm is used. In the Split Levinson algorithm a higher order singular predictor polynomial (P.sub.k (z)) is each time determined in successive recursion steps. After the last recursion step the formant frequencies (f.sub.1, f.sub.2, . . . ) are determined.

Method of and device for deriving formant frequencies using a Split

The pitch of human speech segments is analyzed using at least two different pitch detection algorithms, a respective plurality of most likely values of pitch is selected by each of those algorithms, and these values and their respective quality figures are analyzed statistically to determine the most likely pitch. One algorithm operates in the frequency domain, by analyzing an amplitude spectrum, and the other algorithm operates in the time domain using an autocorrelation function. Significant peak positions of the amplitude spectrum or autocorrelation function are evaluated in respective harmonic sieves, to provide respective quality figures indicating the degree to which peak frequency or period periods of the spectrum or autocorrelation function output match the apertures of the harmonic sieve. A predetermined number of values of pitch, and of period, are selected having the highest quality figures. After conversion of the values for period into values of pitch, these values with their associated quality figures are analyzed statistically to form an estimation of the most likely pitch.

System for analyzing human speech

Method of and arrangement for the determination of the pitch of speech signals in a system of speech analysis, wherein sequences of significant peak positions of the amplitude spectrum of a speech signal are derived (13) from time segments of the speech signal by means of a discrete Fourier transform (12). In order to reduce the influence of noise signals and noise components, respectively, in the amplitude spectrum the significant peak positions are compared with different masks (15), which have apertures at harmonic distances of the associated fundamental tone. The mask which matches the sequence of significant peak positions best is selected (20). A probable value for the pitch is now computed with the harmonic numbers now known of the significant peak positions which are located in apertures of the selected mask. The mean square error between these significant peak positions and the corresponding harmonics of the finished tone can be used as a criterion (21).

Method of and system for determining the pitch in human speech

LPC-synthesizing device, in which a modulation of the synthesized signal with a window signal is used to reduce the buzz which is characteristic for such devices. This window signal has an amplitude which initially increases gradually from substantially zero value to a constant value, and then decreases gradually from the constant value to substantially zero value. As a result of this modulation the signal in the transition between two segments of voiced speech is forced to zero thereby eliminating any transition discontinuities, the existence of which causes the buzz.

Arrangement and method for generating a speech signal

In a formant speech analysis synthesis system, formant extraction to control a recursive digital all-pole filter encounters the problem that pole-pairs are not orderly arranged and that real poles may occur which are not representative of formants. The problem is solved by transforming the coefficients of the second-order sections of the filter to coefficients which can be easily ordered and by means of which it is simple to assign formants to the real poles.

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Leonardus F Willems