摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(ω), based on a power spectrum P1(ω) and on a first calculated value obtained by at least multiplying a power spectrum P2(ω) by a weight coefficient A2(ω). A coefficient update unit (300) updates the weight coefficient A2(ω) and a weight coefficient A1(ω) so that a second calculated value approximates to the power spectrum P1(ω). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(ω) and the estimated target sound power spectrum Ps(ω) by the weight coefficient A2(ω) and the weight coefficient A1(ω), respectively.
摘要:
A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(ω), based on a power spectrum P1(ω) and on a first calculated value obtained by at least multiplying a power spectrum P2(ω) by a weight coefficient A2(ω). A coefficient update unit (300) updates the weight coefficient A2(ω) and a weight coefficient A1(ω) so that a second calculated value approximates to the power spectrum P1(ω). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(ω) and the estimated target sound power spectrum Ps(ω) by the weight coefficient A2(ω) and the weight coefficient A1(ω), respectively.
摘要:
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.
摘要:
A low-cost speech recognition apparatus for AV equipment capable of speech recognition with high accuracy while 2-channel sound is being produced from loudspeakers is achieved. A monaural conversion part converts 2-channel signals to be inputted to the loudspeakers into a monaural signal. A single echo canceller is provided with an output from a microphone and an output from the monaural conversion part (monaural signal). The echo canceller estimates an echo of multichannel sound based on the monaural signal, and then eliminates the echo sound from the microphone output. Thus, with only a single echo canceller, speech recognition can be carried out while 2-channel sound is being produced from the loudspeakers. Moreover, unlike the case where two echo cancellers are provided, the present invention can prevent the occurrence of mutual interference between the echo cancellers that leads to deterioration in speech recognition performance.
摘要:
An array microphone is a directional microphone having an improved directional characteric in which both the sensitivity and the sound pressure frequency response are uniform within the recording area and more particularly, the quality and level of sound remain unchanged. The array microphone includes a microphone array including a plurality of microphone units, and a two-dimensional filter for filtering an output of the microphone array in the dimensions of both time and space. When the two-dimensional filter is a digital filter and varied in its two-dimensional filter coefficient and sampling frequency, the array microphone serves as a variable directional microphone whose directional characteristic can be varied with the sound quality and level remaining unchanged throughout the recording area.
摘要:
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.