Abstract: A first correlation matrix, which corresponds to observed signals, is calculated, a non-scan range is specified, a second correlation matrix, which corresponds to an acoustic signal from a sound source within the non-scan range, is estimated, a third correlation matrix, which corresponds to a target sound source within a scan range, is calculated by removing the second correlation matrix from the first correlation matrix, and a first spatial spectrum, which is a localization result, is calculated from the third correlation matrix. In the estimation, the second correlation matrix is estimated from direction vectors calculated from a direction range of the non-scan range and a second spatial spectrum, which is a localization result calculated immediately before the first spatial spectrum is calculated.

Abstract: A first correlation matrix, which corresponds to observed signals, is calculated, a non-scan range is specified, a second correlation matrix, which corresponds to an acoustic signal from a sound source within the non-scan range, is estimated, a third correlation matrix, which corresponds to a target sound source within a scan range, is calculated by removing the second correlation matrix from the first correlation matrix, and a first spatial spectrum, which is a localization result, is calculated from the third correlation matrix. In the estimation, the second correlation matrix is estimated from direction vectors calculated from a direction range of the non-scan range and a second spatial spectrum, which is a localization result calculated immediately before the first spatial spectrum is calculated.

Abstract: Provided is a vehicle-mounted sound processing device for improving the accuracy of processing of sounds collected in a vehicle interior room. The vehicle-mounted sound processing device comprises: a microphone module disposed in the vehicle interior room; and a head-unit disposed separately from the microphone module in the vehicle interior room. The microphone module comprises: a plurality of sound conversion elements that convert input voice signals to electric signals; and a sensitivity correction unit that corrects variations of sensitivity between the plurality of sound conversion elements for the signals input from the plurality of sound conversion elements. The signals as processed by the sensitivity correction unit are output to the head-unit.

Abstract: An apparatus includes a first microphone disposed near a first seat in a car cabin, the first seat configured to seat a first occupant, a second microphone disposed near a second seat in the car cabin, the second seat configured to seat a second occupant, a processor, and a memory storing a computer program. The computer program, when executed, causes the processor to execute operations including: estimating a first mixed speech using a second signal acquired by the second microphone, the first mixed speech including a first speech that has been uttered by the second occupant and has entered the first microphone, and removing the first mixed speech from a first signal acquired by the first microphone.

Abstract: A sound source localization apparatus is provided which can more surely detect a sound source located in a detection target region. The sound source localization apparatus includes a plurality of microphone and a buffle. The buffle has a first surface and a second surface. The second surface is a surface opposite to the first surface. The plurality of microphones are two-dimensionally arrayed and fixed in the first surface. The buffle allows the plurality of microphones to pick up direct sound arriving at the first surface and prevents the plurality of microphones from picking up direct sound arriving at the second surface.

Abstract: A beamforming method includes: generating a main beam signal by performing, with a focus on a first region, a delay and sum operation on receiving echo signals; generating a sub beam signal which has a low sensitivity to ultrasound signals reflected off the first region; and generating a narrow beam signal by (i) calculating a coefficient for narrowing an angle of the main beam signal, and (ii) multiplying the main beam signal by the coefficient, wherein in the generating of a sub beam signal, the sub beam signal is generated using a differential signal that is a difference between two beam signals each of which is generated by performing a delay and sum operation on the receiving echo signals, with a focus on a corresponding one of two regions of a subject which are different from the first region and are different from each other.

Abstract: A hearing aid capable of detecting contact vibration noise from a collected sound signal. The hearing aid is provided with two microphones, a vibration component extracting section that extracts from collected sound signals respectively obtained by the two microphones an uncorrelated component between two collected sound signals as a vibration component for each frequency band. Additionally, a vibration noise identifying section determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting section, an acoustic signal processing section, 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 converts the acoustic signal to sound.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

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: A sound processing apparatus which can improve precision of analyzes on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors. The sound processing apparatus is provided with a level signal converter that converts the collected sound signal into a level signal, which indicates an absolute value of the collected sound signal from which phase information is removed. A level signal synthesizer generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors are synthesized, and a detector/identifier carries out analysis on the ambient sounds, based upon the synthesized level signal.

Abstract: A directional microphone includes: a microphone which generates a first acoustic signal that has sensitivity in a target direction; a microphone which generates a second acoustic signal that has a blind spot in sensitivity in the target direction; a correction unit which multiplies, in a frequency domain, the second acoustic signal by the first acoustic signal N times, where N is greater than zero, to generate a third acoustic signal which includes the second acoustic signal that has a narrowed angular range of the blind spot in sensitivity in the target direction; and a suppression unit which performs noise suppression using the first acoustic signal as a main signal and the third acoustic signal generated as a reference signal, to generate an output acoustic signal which is the first acoustic signal that has narrowed directivity in the target direction.

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. Here, an signal intensity for the target region regarding the main beam is higher than a signal intensity for the target region regarding each of the sub beams.

Abstract: A noise cancelling device includes an extracting unit configured to extract a first noise from a signal, the signal being based on an input audio signal, a storing unit configured to store noise characteristic information on a second noise, the second noise remaining after subtracting the extracted first noise from the signal based on the audio signal. And the device further includes a cancelling unit configured to perform cancelling processing for cancelling a noise on the input audio signal based on the first noise and the noise characteristic information on the second noise.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

Abstract: A directional microphone includes: a microphone which generates a first acoustic signal that has sensitivity in a target direction; a microphone which generates a second acoustic signal that has a blind spot in sensitivity in the target direction; a correction unit which multiplies, in a frequency domain, the second acoustic signal by the first acoustic signal N times, where N is greater than zero, to generate a third acoustic signal which includes the second acoustic signal that has a narrowed angular range of the blind spot in sensitivity in the target direction; and a suppression unit which performs noise suppression using the first acoustic signal as a main signal and the third acoustic signal generated as a reference signal, to generate an output acoustic signal which is the first acoustic signal that has narrowed directivity in the target direction.

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 processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

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 beamforming method includes: generating a main beam signal by performing, with a focus on a first region, a delay and sum operation on receiving echo signals; generating a sub beam signal which has a low sensitivity to ultrasound signals reflected off the first region; and generating a narrow beam signal by (i) calculating a coefficient for narrowing an angle of the main beam signal, and (ii) multiplying the main beam signal by the coefficient, wherein in the generating of a sub beam signal, the sub beam signal is generated using a differential signal that is a difference between two beam signals each of which is generated by performing a delay and sum operation on the receiving echo signals, with a focus on a corresponding one of two regions of a subject which are different from the first region and are different from each other.