Pioneer Corporation

Techexperts Incorporation

Pioneer Electronic Corporation

Mitsuya Komamura

Soichi Toyama

Hiroshi Seo

Keiichi Yamauchi

Ikuo Fujita

A band extending apparatus () is provided with: first generating device () for generating a baseband signal (XB(n)) by up-sampling an input signal (X(n)) and then transmitting it through a low-pass filter; a second generating device () for generating a high-frequency signal (X(n)), by extracting a signal component on a higher-frequency side of a signal which is obtained by squaring a band limited signal (X(n)) which is a signal component with a predetermined band of the baseband signal; and a third generating device () for generating an output signal (X(n)) by adding the high-frequency signal to the baseband signal.

Band extending apparatus and method

An operator recognition device is provided that eliminates the registration of data such as HMM data having a characteristic amount for which error in recognition occurs easily when recognizing an operator, and thus reduces the possibility of errors in recognition, and has stable recognition performance. When registering HMM data that is used when performing recognition processing, a speaker recognition deviceeliminates the registration of HMM data of a password having a characteristic amount of the spoken voice component that is similar to a characteristic amount that is indicated by HMM data that is already registered, and does not allow the registration of HMM data for which it is estimated that error in recognition will occur easily during the recognition process.

Operator recognition device, operator recognition method and operator recognition program

A noise suppression apparatus calculates a sound spectrum and a noise spectrum from an input sound, further calculates gain based on the sound spectrum and noise spectrum, and suppresses noise in the input sound. The noise suppression apparatus includes a first frame-dividing unit that divides the input sound into frames having a predetermined frame length, a second frame-dividing unit that divides the input sound into frames having a longer frame length than the frame length of the first frame-dividing unit, a second converting unit that converts, into a spectrum, the input sound divided into frames by the second frame-dividing unit, a smoothing unit that smoothes the converted spectrum in a frequency direction, and a gain calculating unit that calculates gain based on the smoothed spectrum and the noise spectrum.

Apparatus and methods for noise suppression in sound signals

A pulse width modulator for producing a PWM signal having reduced nonlinear distortion through interpolation processing with fewer computation steps is provided. The computational processing, W={1+&#x3b1;(X&#x2212;X)}{0.25 (X+X)}+0.5, is performed in accordance with successive sample values (X, X) in a PCM data train to determine a weighting factor (W). The computational processing, Xq=0.5&#xb7;X(W&#x2212;W)+X(1&#x2212;W)+0.5&#xb7;X(W+W), is performed using the sample values (X, X), a sample value (X) previous to the sample value (X), and the weighting factor (W) to thereby determine an interpolated sample value (Xq) having an amplitude close to that of an original analog signal (X(t)) generating the PCM data train. A point in time (tq) at which a reference signal (R(t)) takes on the interpolated sample value (Xq) is then determined to produce a PWM signal (Spwm) which is logically inverted at the point in time (tq).

Pulse width modulator and pulse width modulation method

An initial combination HMMis generated from a voice HMMhaving multiplicative distortions and an initial noise HMM of additive noise, and at the same time, a Jacobian matrix J is calculated by a Jacobian matrix calculating sectionNoise variation Namh (cep), in which an estimated value Ha^(cep) of the multiplicative distortions that are obtained from voice that is actually uttered, additive noise Na(cep) that is obtained in a non-utterance period, and additive noise Nm(cep) of the initial noise HMMare combined, is multiplied by a Jacobian matrix, wherein the result of the multiplication and initial combination HMMare combined, and an adaptive HMMis generated. Thereby, an adaptive HMMthat is matched to the observation value series RNah(cep) generated from actual utterance voice can be generated in advance. When performing voice recognition by collating the observation value series RNah(cep) with adaptive HMMinfluences due to the multiplicative distortions and additive distortions are counterbalanced, wherein an effect that is equivalent to a case where voice recognition is carried out with clean voice can be obtained, and a robust voice recognition system can be achieved.

Voice recognition system

A multiplicative distortion Hm(cep) is subtracted from a voice HMM, a multiplicative distortion Ha(cep) of the uttered voice is subtracted from a noise HMMformed by HMM, and the subtraction results Sm(cep) and {Nm(cep)&#x2212;Ha (cep)} are combined with each other to thereby form a combined HMMin the cepstrum domain. A cepstrum R^a(cep) obtained by subtracting the multiplicative distortion Ha (cep) from the cepstrum Ra (cep) of the uttered voice is compared with the distribution R^m(cep) of the combined HMMin the cepstrum domain, and the combined HMM with the maximum likelihood is output as the voice recognition result.

The digital signal conversion apparatus comprises: an over-sampling circuit that samples the input digital signal at high frequency; a polarity-inversion circuit that inverts the polarity of the sampled digital signal; interpolation circuit (A) and interpolation circuit (B) that perform interpolation of each respective digital signal; noise-shaping circuit (A) and noise-shaping circuit (B) that perform noise shaping on the interpolated signals; PWM conversion circuit (A) and PWM conversion circuit (B) that perform PWM conversion on the noise-shaped signals; and a switching circuit for driving a load based on the PWM signals from PWM conversion circuit (A) and PWM conversion circuit (B).

Digital signal conversion apparatus, digital signal conversion method, and computer-readable recording medium in which digital signal conversion program is recorded

A dither generating apparatus which generates a sufficiently random auto dither even for a small-level signal and even if the buffer length is short. The LSB of quantized data is extracted and is stored in a buffer memory which serves as an M-bit shift register. An index buffer storing old 2.sup.M indexes is referred to with the M bits. A look-up table, which outputs a random value, is referred to in accordance with the value in the index buffer and outputs a dither. After the reference to the look-up table, current M-bit data is input to the index buffer so that the content of the index buffer is shifted piece by piece. As the values of the outputs of the index buffer and the buffer memory are both always variable, different values are always supplied to the look-up table, which therefore generates sufficiently random dithers.

Growing community of inventors

Tsurugashima, Japan

Mitsuya Komamura