Abstract: In a masking sound generating apparatus, a band divider divides a target sound signal into a plurality of frequency bands to generate a plurality of band signals. An envelope signal generating part generates a plurality of envelope signals representing respective envelopes of the plurality of the band signals. A signal converter segments each of the plurality of the envelope signals into a plurality of frames, then specifies frames of segmented envelope signals which have an amplitude greater than a first threshold and less than a second threshold, and changes an order of the specified frames in an arrangement of the plurality of the frames. A multiplier multiplies each of the plurality of the envelope signals by a noise signal, each envelope signal having the order of the frames changed by the signal converter, and outputs the plurality of the envelope signals multiplied by the noise signal as individual band masking signals.

Abstract: In a method for suppressing a noise by the spectrum subtraction method, it is possible to improve the noise suppression capability by simultaneously obtaining a frequency resolution required for the noise estimation spectrum and a temporal resolution required for the noise suppression spectrum. The signal length of an observation signal cut out for analyzing the spectrum of the observation signal used for estimation calculation of the noise spectrum is set longer than the signal length of an observation signal cut out for analyzing the spectrum of the observation signal as a value to be subtracted for performing subtraction with the noise spectrum.

Abstract: An impulse response synthesis method is carried out by a dispersion calculation process for calculating a dispersion of phase characteristics in association with a given room based on a volume of the room, a degree of sound absorption of the room, and a distance between a sound source and a receiving point arranged in the room, a noise creation process for creating a noise having the dispersion of the phase characteristics calculated in the dispersion calculation process, a trend addition process for adding a phase trend to the noise created by the noise creation process in accordance with the distance between the sound source and the receiving point and obtaining a phase characteristic of a minimum-phase component from the noise added with the phase trend, and a synthesis process for synthesizing an impulse response based on the phase characteristic of the minimum-phase component, the impulse response being used to create reverberation for the room.

Abstract: A similarity degree estimation method is performed by two processes. In a first process, an inter-band correlation matrix is created from spectral data of an input voice such that the spectral data are divided into a plurality of discrete bands which are separated from each other with spaces therebetween along a frequency axis, a plurality of envelope components of the spectral data are obtained from the plurality of the discrete bands, and elements of the inter-band correlation matrix are correlation values between the respective envelope components of the input voice. In a second process, a degree of similarity is calculated between a pair of input voices to be compared with each other by using respective inter-band correlation matrices obtained for the pair of the input voices through the inter-band correlation matrix creation process.

Abstract: Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

Abstract: Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

Abstract: A hearing aid device for processing an input sound signal via a microphone so as to output a sound signal from an earphone comprises a signal processing element, a couple of acoustic nerve excitation pattern calculation elements, a couple of acoustic filter shape memory elements, a comparison element, a correction processing element, etc. The couple of acoustic nerve excitation pattern calculation elements calculate acoustic nerve excitation patterns of a normal hearing person and a hearing impaired person based on an output signal of the signal processing element and acoustic filter shapes of the normal hearing person and the hearing impaired person which are stored in the couple of acoustic filter shape memory elements. Each of the acoustic nerve excitation patterns is compared via the comparison element.

Abstract: In a masking sound generating apparatus, a band divider divides a target sound signal into a plurality of frequency bands to generate a plurality of band signals. An envelope signal generating part generates a plurality of envelope signals representing respective envelopes of the plurality of the band signals. A signal converter segments each of the plurality of the envelope signals into a plurality of frames, then specifies frames of segmented envelope signals which have an amplitude greater than a first threshold and less than a second threshold, and changes an order of the specified frames in an arrangement of the plurality of the frames. A multiplier multiplies each of the plurality of the envelope signals by a noise signal, each envelope signal having the order of the frames changed by the signal converter, and outputs the plurality of the envelope signals multiplied by the noise signal as individual band masking signals.

Abstract: A method is provided for recurrently estimating a spectrum of noise at each signal observation interval from a sound signal which contains the noise and which is observed at each signal observation interval. In the method, there are acquired an envelope of a previous spectrum of the noise which has been previously estimated from the sound signal observed at a previous signal observation interval, and an envelope of a current spectrum of the sound signal which is observed at a current signal observation interval subsequent to the previous signal observation interval. Then, a value of correlation is computed between the envelop of the previous spectrum of the noise and the envelope of the current spectrum of the sound signal. A current spectrum of the noise contained in the sound signal observed at the current signal observation interval is estimated in accordance with the computed value of the correlation and based on the previous spectrum of the noise and the current spectrum of the sound signal.

Abstract: A method is designed for emphasizing a howling frequency component of a sound signal observed in an acoustic feedback system during an observation period having a predetermined length.

Abstract: Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

Abstract: Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

Abstract: Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

Abstract: A similarity degree estimation method is performed by two processes. In a first process, an inter-band correlation matrix is created from spectral data of an input voice such that the spectral data are divided into a plurality of discrete bands which are separated from each other with spaces therebetween along a frequency axis, a plurality of envelope components of the spectral data are obtained from the plurality of the discrete bands, and elements of the inter-band correlation matrix are correlation values between the respective envelope components of the input voice. In a second process, a degree of similarity is calculated between a pair of input voices to be compared with each other by using respective inter-band correlation matrices obtained for the pair of the input voices through the inter-band correlation matrix creation process.

Abstract: An impulse response synthesis method is carried out by a dispersion calculation process for calculating a dispersion of phase characteristics in association with a given room based on a volume of the room, a degree of sound absorption of the room, and a distance between a sound source and a receiving point arranged in the room, a noise creation process for creating a noise having the dispersion of the phase characteristics calculated in the dispersion calculation process, a trend addition process for adding a phase trend to the noise created by the noise creation process in accordance with the distance between the sound source and the receiving point and obtaining a phase characteristic of a minimum-phase component from the noise added with the phase trend, and a synthesis process for synthesizing an impulse response based on the phase characteristic of the minimum-phase component, the impulse response being used to create reverberation for the room.

Abstract: Sound picked up by a microphone of a first sound field is reproduced by a speaker of a second sound field, and a sound picked up by a microphone of the second sound field is reproduced by the speaker of the first sound field. Sound pressure detection section detects a sound pressure of a sound present in the second (or first) sound field picked up by the microphone of the second (or first) sound field, other than a sound reproduced by the speaker of the first (or second) sound field. Other sound pressure detection section detects a sound pressure with which the sound picked up by the microphone of the second (or first) sound field and reproduced by the speaker of the first (or second) sound field is picked up by the microphone of the first (or second) sound field. Sound-pressure-difference detection section adjusts a gain of an automatic gain adjustment section in such a manner that the two detected sound pressures assume a predetermined relationship.

Abstract: A hearing aid device for processing an input sound signal via a microphone so as to output a sound signal from an earphone comprises a signal processing element, a couple of acoustic nerve excitation pattern calculation elements, a couple of acoustic filter shape memory elements, a comparison element, a correction processing element, etc. The couple of acoustic nerve excitation pattern calculation elements calculate acoustic nerve excitation patterns of a normal hearing person and a hearing impaired person based on an output signal of the signal processing element and acoustic filter shapes of the normal hearing person and the hearing impaired person which are stored in the couple of acoustic filter shape memory elements. Each of the acoustic nerve excitation patterns is compared via the comparison element.

Abstract: A method is provided for recurrently estimating a spectrum of noise at each signal observation interval from a sound signal which contains the noise and which is observed at each signal observation interval. In the method, there are acquired an envelope of a previous spectrum of the noise which has been previously estimated from the sound signal observed at a previous signal observation interval, and an envelope of a current spectrum of the sound signal which is observed at a current signal observation interval subsequent to the previous signal observation interval. Then, a value of correlation is computed between the envelop of the previous spectrum of the noise and the envelope of the current spectrum of the sound signal. A current spectrum of the noise contained in the sound signal observed at the current signal observation interval is estimated in accordance with the computed value of the correlation and based on the previous spectrum of the noise and the current spectrum of the sound signal.

Abstract: A method is designed for emphasizing a howling frequency component of a sound signal observed in an acoustic feedback system during an observation period having a predetermined length.

Abstract: Two signals, which are mutually correlated, are subjected to a principal component analysis and converted into two signals being put in an orthogonal relation, thereby generating two signals being non-correlated. Those two signals are reproduced by speakers, and the voice generated from the speakers are collected by microphones. The cross spectra of a signal as the result of subtracting an echo canceling signal from a voice collected by each microphone, and a voice before it is generated from the speaker, are obtained. Those cross spectra are ensemble-averaged for a predetermined period of time, and inverse Fourier transformed, thereby producing impulse response estimation errors of each filter. Impulse responses of those filters are updated so as to cancel those impulse response estimation errors.