Abstract: The invention provides a speech synthesis apparatus which can produce synthetic speech of a high quality with reduced distortion. To this end, upon production of synthetic speech based on prosodic information and phonological unit information, the prosodic information is modified using the phonological unit information, and duration length information and pitch pattern information of phonological units of the prosodic information and the phonological unit information are modified with each other.

Abstract: Input data are analyzed by FFT, etc. in an analyzing section and power every frequency band is calculated and sent to a control section. In a gain control section, changing characteristics of a gain used in the control section are calculated on the basis of hearing ability characteristics of a user obtained from a memory section and a gain setting memory section, a sound pressure for starting a reduction in gain, and a sound pressure for setting the gain to be equal to or greater than 0 dB. The calculated changing characteristics are sent to the control section. In the control section, the gain every frequency band required in a hearing sense compensating section is determined on the basis of analyzed results obtained from the analyzing section, the hearing ability characteristics of the user obtained from the memory section, and the changing characteristics of the gain obtained from the gain control section. The control section sends data of the gain to the hearing sense compensating section.

Abstract: A digital hearing aid having a variable hearing compensating characteristics, comprises a hearing compensating circuit having a transposed transversal filter, an analyzer for frequency-analyzing an input signal, a memory storing a hearing characteristics of a person to be fitted with the hearing aid, and a controller receiving a frequency analysis result of the input signal and the hearing characteristics, for deriving coefficients for the transposed transversal filter to supply the derived coefficients to the transposed transversal filter. Since the transposed transversal filter is used, the S/N ration is improved, and the control of the characteristics of the filter becomes easy.

Abstract: A digital hearing aid has input means for converting an input sound into a digital data for generating an input data, analyzing means for analyzing the input data converted by the input means by a digital conversion and calculating an acoustic pressure at each frequency band, control means for inputting a result of calculation by the analyzing means, acoustic sense characteristics storage means for preliminarily storing acoustic sense characteristics of a deafness and a person having healthy acoustic sense from a fitting means, gain calculation data storage means for preliminarily storing an acoustic pressure range the easiest to hear for the deafness from the fitting means, and acoustic sense compensating means for performing acoustic sense compensation process by amplifying the input data with a given gain.

Abstract: An external device to be used along with a hearing aid device comprises an input means through which voice data are inputted, a hearing aid processor coupled to the input means for receiving the voice data from the input means to make an acoustic sense compensation of the voice data, and a transmitter coupled to the hearing aid processor for receiving compensated voice data from the hearing aid processor and transmitting the compensated voice data to the hearing aid device.

Abstract: Respective samples of input speech data are transformed by a discrete Fourier transformer to obtain a spectrum of the speech data. Simultaneously, values of times of the respective samples are multiplied with the input speech data by a multiplier and a differential spectrum is obtained by transforming a result of the multiplication by a discrete Fourier transformer. A real part of a value obtained by dividing the differential spectrum by the spectrum by a quotient real part calculator and the real part is inverse-transformed by a discrete inverse Fourier transformer. The result of the inverse-transformation is divided by the values of the times of the respective samples to obtain a time function corresponding to phase. On the other hand, a time function corresponding to a logarithmic amplitude spectrum is obtained from an output of the inverse-Fourier transformer by means of a logarithmic amplitude spectrum calculator.

Abstract: A text-to-speech synthesizer comprises an analyzer that decomposes a sequence of input characters into phoneme components and classifies them as a first group of phoneme components or a second group if they are to be synthesized by a speech parameter or by a formant rule, respectively. Speech parameters derived from natural human speech are stored in first memory locations corresponding to the phoneme components of the first group and the stored speech parameters are recalled from the first memory in response to each of the phoneme components of the first group. Formant rules capable of generating formant transition patterns are stored in second memory locations corresponding to the phoneme components of the second group, the formant rules being recalled from the second memory in response to each of the phoneme components of the second group.

Abstract: In order to simplify a speech synthesizer arrangement concurrently with improvement of operation flexibility thereof, a digital memory is arranged to store at least one voiced sound source and at least one unvoiced sound source. One of the sound sources is selected in accordance with the content of a first register, while the data within the selected source is specified by the content of a shift register sequence generator. Each of the bit patterns obtained at the shift register sequence generator is compared with the content of a second register. In the event that the contents of the sequence generator and the second register coincide, the shift register sequence generator is reset and/or the operating conditions(s) of the synthesizer is changed.

Abstract: A logarithmic frequency spectrum related to an input signal is converted by the use of an inverse Fourier transform into a cepstrum. The cepstrum has a first and a second frequency component which have a first peak and a second peak spaced apart from the first peak by a preselected period on an axis of frequency, respectively. The second frequency component is processed into a peak controlled frequency component having a controlled peak coincident with the first peak. The peak controlled frequency component and the first frequency component are summed up to produce an ultimate frequency component which corresponds to the value of the envelope parameter.

Abstract: For use in deciding optimum pole and zero parameters for an input signal with reference to an input cepstrum of the signal, candidate pole and zero parameters are stored in advance in a table. Supplied with an impulse and controlled by a candidate set pair of a pole parameter set and a zero parameter set selected from the candidate pole and zero parameters of the table, first and second filters produce first and second outputs defined by terms of up to a certain order. Responsive to the first and second outputs and to factors of multiplication which are given by inverse numbers of time intervals related to the respective terms, an analysis filter produces a converted signal which is equivalent to a model output cepstrum of a model output signal produced by a pole-zero model defined by the candidate set pair. A cepstrum subtracter calculates a cepstrum difference between the input cepstrum and the converted signal.

Abstract: A pole-zero analyzer includes an autocorrelation impulse response calculator, first and second buffers, a pole parameter value table, a first coefficient calculator, a first inverse filter, a second coefficient calculator, a second inverse filter, a pole-zero calculator for calculating a pole parameter and a zero parameter in response to outputs from the first and second inverse filters and calculating an error value for determining the pole parameter, a buffer memory for generating an output from the pole parameter value table obtained when the error value satisfies predetermined conditions and predetermined portions of the zero parameter and the pole parameter calculated by the pole-zero calculator as outputs of the pole-zero analyzer, and a control circuit for controlling operation timings of the respective parts.