Abstract:
The aim of the invention is to receive and treat audiosignals with a good user signal to fault signal ratio in noise conditions and with a good ratio between the direct and the reflected echo in surroundings which are especially not free from reverberation. Electrical signals are produced by converting recorded audiosignals. Said electrical signals are treated by a given microphone assembly in such a way that electrical signals having different strengths (different sensitivities of the microphones) and being produced by the microphones are compensated automatically, i.e. without manual and individual compensation procedures which have to be carried out separately, when the sound pressure levels of the microphones pertaining to the microphone assembly are equal. According to the invention, the properties of an array of microphones are combined to the properties of a method for compensating the sensitivity of microphones.
Abstract:
A method of processing sound comprises the steps of: detecting sounds at at least two spaced detecting locations; analysing the detected sounds to identify the angular relation between respective sound sources and the detecting locations; permitting selection of an angular relation associated with a particular sound source; and processing the detected sounds in response to the selection to highlight a stream of sound associated with the particular sound source. The method may be utilised in a hearing aid, to allow a user to stream sounds by interactively selecting a particular source, and thus minimise background "noise".
Abstract:
The invention relates to a method for extracting statistically independent signals from a mass of mixed input signals, based on the information maximization principle. Numerical stability of a method based on the information maximization principle is guaranteed when the determinants of the transformation matrix have a constant value (step 504) in the rule determining maximum likelihood expectation. The transformation matrix and likelihood expectation models of intermediate signals are adapted in a training phase (step 502). The statistically independent intermediate signals are determined (step 505) in relation to the input signals in an application phase using the transformation matrix arising after adaptation.
Abstract:
This invention relates to a method and an apparatus for reducing ambient noise for use with a headset or a boom headset attached to a boom microphone device or the like. The apparatus can include a sensor microphone to detect a background noise signal, a desired input audio transmission, and signal processing means for canceling the noise signals to create an inverted antinoise signal within an acoustical waveguide located adjacent to the earphone of headset. The method for reducing noise according to this invention is provided by an open loop circuit allowing the input audio signal from an operator or caller to be transmitted to the user's ear without the disturbance of unwanted ambient noise. The method provides ajustments to the gain/or and phase of a noise signal for canceling the noise component detected, within an acoustical waveguide to produce a quiet zone for the desired audio speech to be transmitted. The apparatus can also include a noise cancellation microphone transmitter system having a first and second microphone arranged such that the first microphone receives a desired speech input and the background noise present in the vicinity of the speech, and the second microphone receives substantially only the background noise. The background noise from the second microphone is converted into a corresponding electrical signal and subtracted from a signal corresponding to the speech and background noise obtained from the first microphone so as to produce a signal representing substantial the speech. The active noise cancellation and noise reduction system is enhanced with the following features: an automatic audio microphone transmission by sensing speech (a "VOX" circuit), transmitting portion of microphone signal to earcup speaker by an increased gain sidetone channel ("sidetone"), and converting an active noise cancellation microphone to a standard omni-directional microphone by removing the voice microphone from the circuit design and increasing the gain of the noise microphone amplifier. The first and second microphones may be utilized as a directional microphone according to this invention when a far field response is desired. The method of the invention also relates to the use of a two terminal microphone configuration.
Abstract:
A system insensitive to nonspeech sounds utilizes a pair or spatially separated microphones (101, 102) to obtain the direction of origin of speech signals from a common sound source. The speech signal from each microphone is transformed (330, 340, 350, 355, 360, 365, 370, 375) into a pulse representative signal having a rapid increase responsive to pitch peaks of energy from the sound source. The cross correlation of these pulses accurately reflects the phase relationship between the speech signals arriving at the microphones. The cross correlation is implemented (450, 460) as time interval histograms which are periodically read to identify the direction of the common sound source.
Abstract:
The present invention relates to an acoustic camera using a MEMS microphone array comprising an acoustic sensing device using the MEMS microphone array, in which a plurality of MEMS microphones (10) is mounted on a printed circuit board (20), and a display unit (60) displaying a calculated noise level by color. The MEMS microphone (10) has 11-30 wing units expanded in a radial longitudinal direction, and the 2-30 MEMS microphones (10) are arranged on the one wing unit.
Abstract:
본 발명은 어레이 스피커 시스템 및 그 구현 방법에 관한 것으로, 본 발명에 따른 어레이 스피커 시스템은 미리 설정된 조절 인자에 따라 어레이에 포함된 복수 개의 스피커들의 조합들을 각각 하나의 그룹으로 설정하는 그룹 설정부, 조절 인자에 따라 입력 신호를 하나 이상의 음원 신호로 분리하여 설정된 그룹에 선택적으로 할당하는 신호 할당부, 그룹의 음원 특성에 따라 그룹에 해당하는 스피커들을 통해 출력할 음원 신호를 조절하는 음원 조절부 및 그룹에 해당하는 스피커들을 통해 조절된 음원 신호를 출력하는 신호 출력부를 포함함으로써, 광대역 주파수의 출력시 주파수 대역별로 최적화된 음 조절과 출력이 가능하고, 출력 음원 신호가 왜곡되는 근접장 효과의 발생을 방지하고, 스테레오 및 멀티 채널 효과를 얻을 수 있으며, 어레이 스피커의 지향성을 향상시킬 수 있다.