Abstract: Disclosed is a hearing aid system capable of increasing the clearness of sound spoken by a speaker while reproducing the incoming direction of the sound spoken by the speaker without using an inverse mapping rule.

Abstract: An echo cancellation apparatus cancels an echo signal included in a second speech signal including the echo signal of a first speech signal, and includes: a first echo replica generating unit generating a first echo replica by processing the first speech signal through a filter whose characteristic is equalized with a acoustic transfer function of amplified sound; a first echo cancelling unit generating and providing a first echo-cancelled signal by subtracting from the second speech signal the first echo replica generated by the first echo replica generating unit; and a filter updating unit generating a nonlinear-converted echo-cancelled signal by performing nonlinear conversion on the first echo-cancelled signal generated by the first echo cancelling unit, and updating the filter characteristic thereby equalizing the filter characteristic with the acoustic transfer function, using the nonlinear-converted echo-cancelled signal, the first speech signal, and a norm of the first speech signal.

Abstract: The present invention provides a hearing aid capable of detecting contact vibration noise from a collected sound signal. A hearing aid (100) is provided with two microphones (110-1, 110-2), a vibration component extracting unit (120) which extracts from collected sound signals respectively obtained by the two microphones (110-1, 110-2) an uncorrelated component between two collected sound signals as a vibration component for each frequency band, a vibration noise identifying unit (130) which determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting unit (120), an acoustic signal processing unit (140) which, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver (150) which converts the acoustic signal to sound.

Abstract: A hearing aid device includes a plurality of microphones which converts picked sound to audio signals, an inter-signal phase difference calculation unit which calculates a phase difference between a first audio signal and a second audio signal, a hand placement determination unit, a directivity sound pickup unit which generates an output signal, a nonlinear amplification unit which controls the signal level of the output signal acquired from the directivity sound pickup unit, and a hearing aid process control unit which, when the hand placement determination unit determines that a state where a phase difference is equal to or smaller than a first threshold value continues for a first time, controls the directivity sound pickup unit such that at least one of the sensitivity-frequency characteristic and the frequency characteristics of the audio signals becomes nondirective, and controls the nonlinear amplification unit to amplify the signal levels of the acquired audio signals.

Abstract: The directional microphone device according to the present invention solves a problem of increase in thermal noise (problem of decrease in sensitivity) that occurs at the time of directivity synthesis. The directional microphone device includes: a plurality of microphones which have directional and non-directional characteristics; a control unit which generates an output signal using signals outputted from each of the plurality of microphones; and an output unit which outputs the output signal generated by the control unit. The control unit generates the output signal such that a nearly non-directional directivity and a high sensitivity are obtained in small amplitude range of the output signal, and a directivity and a low sensitivity are obtained in large amplitude range of the output signal.

Abstract: A noise extraction device of the present invention includes: first and second microphone units (11 and 12) each picking up a sound; a directivity synthesis unit which performs a directivity synthesis on output signals respectively received from the first and second microphone units (11 and 12) and generates two directionally synthesized signals which have: different sensitivities to noise; the same directional pattern with respect to sound pressure; and the same effective acoustic center position; and an acoustic cancellation unit which cancels an acoustic component of one of the two directionally synthesized signals by subtracting the one of the two directionally synthesized signals from the other of the two directionally synthesized signal, so as to extract a noise component.

Abstract: A beamforming method according to the present invention is a method of processing echo signals of a target region which are obtained from a probe including a plurality of receiving elements arrayed on a predetermined line. The beamforming method includes the following steps (S1 to S3). At S1, seed beams are formed from echo signals received by at least two receiving elements from among the plurality of receiving elements. At S2, a main beam and sub beams are formed by synthesizing at least one of the seed beams. At S3, a narrow beam for the target region is formed by multiplying the sub beams by respective predetermined coefficients and subtracting the multiplied sub beams from the main beam.

Abstract: The present invention provides a sound collecting device capable of, even when using a freely-movable microphone, generating multichannel signals realizing sound image localization without processing using information about the current position of the microphone.

Abstract: A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(?), based on a power spectrum P1(?) and on a first calculated value obtained by at least multiplying a power spectrum P2(?) by a weight coefficient A2(?). A coefficient update unit (300) updates the weight coefficient A2(?) and a weight coefficient A1(?) so that a second calculated value approximates to the power spectrum P1(?). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(?) and the estimated target sound power spectrum Ps(?) by the weight coefficient A2(?) and the weight coefficient A1(?), respectively.

Abstract: A directional microphone apparatus and directivity control method that corrects a level difference and a phase difference generated in a low band in a plurality of non-directional microphone units, improve the directivity, and reduce the size are provided. Level difference calculation section (105) calculates the level difference between first signal x1(t) obtained by first non-directional microphone unit (101) and second signal x2(t) obtained by second non-directional microphone unit (102), and correction parameter calculation section (106) calculates coefficients of a linear IIR filter configuring correction process section (103) based on the level difference. Correction process section (103) simultaneously corrects the level difference and a phase difference in the low band between two non-directional microphone units by using the calculated coefficients.

Abstract: A sound collection apparatus includes a target sound collection unit that collects a sound including a target sound and outputs a collected-sound signal, a non-target sound collection unit, provided at positions different from each other, forms dead zones of sensitivity in a direction of the target sound source so as to collect a sound outside the dead zones and outputs a collected-sound signal. A sensitivity suppression unit generates a sensitivity suppression signal for suppressing a sound collection sensitivity in an overlap region in which dead zones overlap, as compared to a region surrounding the overlap region, by subjecting, to a predetermined signal processing, the collected-sound signal outputted by the non-target sound collection unit. An extraction unit removes, from the collected-sound signal, the sensitivity suppression signal generated, so as to extract a signal of a sound generated in the overlap region in which the dead zones overlap.

Abstract: A multichannel echo canceller according to the present invention includes an echo cancellation section which receives loudspeaker input signals sp1, sp2 including a second acoustic signal to be respectively inputted to a plurality of loudspeakers 10, 20 provided in a first location and detection signals m1, m2 detected by a plurality of microphones 11, 21 provided in the first location; separates a first acoustic signal and a second acoustic signal, which are included in the detection signals m1, m2, by performing signal processing based on an independent component analysis; cancels as an echo the second acoustic signal, which is included in the detection signals m1, m2, by outputting only the separated first acoustic signal to a plurality of loudspeakers 30, 40 provided in a second location.

Abstract: A wearable terminal is constantly worn by a user and continually picks up images and sounds from the surroundings. Even when using a directional microphone to sensitively pick up targeted audio, the wearable terminal can reduce noise occurring due to motion of the device, for example when the user is walking, and reduce the influence of a shift of the sound pickup direction. For this purpose, the wearable terminal includes a sensor for detecting motion, and performs microphone directivity control to use the directional microphone when the amount of motion is small, and to use an omnidirectional microphone that is not likely to be influenced by noise when the amount of motion is large.

Abstract: Disclosed are a hearing aid device and a hearing aid method capable of controlling a directivity process or an amplification process in a hearing aid with a user's hand placement so as to match a users intuition.

Abstract: Disclosed is a hearing aid system capable of increasing the clearness of sound spoken by a speaker while reproducing the incoming direction of the sound spoken by the speaker without using an inverse mapping rule.

Abstract: A sound processing apparatus (100), which can improve precision of analyses on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors (first sound collector 110-1 and second sound collector 110-2), and the sound processing apparatus (100) is provided with a level signal conversion section (first level signal conversion section 130-1, second level signal conversion section 130-2) that converts the collected sound signal into a level signal, from which phase information is removed, a level signal synthesizing section (140) that generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors (first sound collector 110-1 and second sound collector 110-2) are synthesized, and a detecting and identifying section (160) that carries out analysis on the ambient sounds based upon the synthesized level signal.

Abstract: The directional microphone device according to the present invention solves a problem of increase in thermal noise (problem of decrease in sensitivity) that occurs at the time of directivity synthesis. The directional microphone device includes: a plurality of microphones which have directional and non-directional characteristics; a control unit which generates an output signal using signals outputted from each of the plurality of microphones; and an output unit which outputs the output signal generated by the control to unit. The control unit generates the output signal such that a nearly non-directional directivity and a high sensitivity are obtained in small amplitude range of the output signal, and a directivity and a low sensitivity are obtained in large amplitude range of the output signal.

Abstract: A howling detection device detects a dominance ratio, which indicates a risk of howling to occur when a mixed signal obtained by mixing a plurality of sound signals collected by a plurality of microphones is outputted by a speaker. The howling detection device detects levels of the plurality of sound signals, compares, in a same time domain, the mixed signal with a signal regarding a sound to be outputted by the speaker as a noise reference signal, detects a time period, as a word ending section, during which the mixed signal is inputted after the noise reference signal falls, and calculates a dominance ratio by extracting only a level of the plurality of sound signals corresponding to the word ending section and determining a ratio of each of the extracted levels of each of the sound signals to a sum of the extracted levels of the plurality of sound signals.

Abstract: The present invention provides a sound collecting device capable of, even when using a freely-movable microphone, generating multichannel signals realizing sound image localization without processing using information about the current position of the microphone.

Abstract: Howling, which occurs when amplifying a target sound collected by a first microphone through an amplification section and outputting the amplified sound as an intensified sound from a loudspeaker, is suppressed using a first and second acoustic signal. A first power spectrum is produced according to the first acoustic signal output from the first microphone collecting a sound. A second power spectrum is produced according to the second acoustic signal of a sound including at least the intensified sound and not including the target sound. Then, the first acoustic signal is filtered based on the first power spectrum, and the second power spectrum to output only an acoustic signal of the target sound to the amplification section.