Abstract:
A conferencing unit, comprising an array of microphones embedded in a diffracting object configured to provide a desired high frequency directivity response at predetermined microphone positions, and a low frequency beamformer operable to achieve a desired low frequency directivity response, wherein the beamformer is linearly constrained to provide a smooth transition between low and high frequency directivity responses.
Abstract:
A new approach to tracking head motion for headphone-based sound is described. Called MTB2.0 for “Motion-Tracked Binaural with 2 Channels”, the method may be used for any 2-channel binaural signals without any increase in the bandwidth requirement. MTB2.0 provides a simple and effective means to improve the quality of headphone-based sound reproduction by sensing the orientation of the listener's head and using the sensed orientation to appropriately modify the signals sent to the two ears. MTB2.0 method increases the realism and removes some of the shortcomings of binaural sound capture and recording, as well as improves the quality of binaural rendering of stereo.
Abstract:
A line array loudspeaker, including a first plurality of acoustic drivers each acoustic driver comprising an axis, the first plurality of acoustic drivers arranged so that the axes of first plurality of acoustic drivers are coplanar in a first plane and so that a straight line intersects each axis at a same position on each of the first plurality of acoustic drivers, and a second plurality of acoustic drivers each acoustic driver comprising an axis, the second plurality of acoustic drivers arranged so that the axes of second plurality of acoustic drivers are coplanar in a second plane and so that the straight line intersects each axis at a same position on each of the second plurality of acoustic drivers, in which the first plurality and the second plurality arranged so that the first plane intersects with the second plane along a straight intersection line.
Abstract:
A system such as a speakerphone may include a processor, memory, a speaker and a microphone. The processor may be configured (via program instructions stored in the memory) to calibrate the speaker by: outputting a stimulus signal; receiving an input signal corresponding to the stimulus signal; computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; performing an iterative search for current parameters of a speaker model using the input signal spectrum, the stimulus signal spectrum and the speaker-related sensitivity; and updating averages of the speaker model parameters using the current parameter values. The stimulus signal may be transmitted during periods of silence in the external environment. The parameter averages may be used to perform echo cancellation.
Abstract:
The loudspeaker has a first pair of drivers arranged in a line, a center point along the line, wherein the pair of drivers are substantially centered about the center point with a center to center distance, d0, between the drivers in the first pair of drivers, whereby the maximum frequency with out high amplitude side lobes is equal to c/2d0, and at least a subsequent pair of drivers arranged in the line array with the first pair of drivers and substantially centered about the center point, wherein the subsequent pair of drivers are spaced such that the center to center distance between each driver in the subsequent pair, dn, is equal to 4nd0, where n=0 at the innermost pair of drivers and n increases by 1 with each pair of drivers sequentially added. Each pair of drivers for n>0 has a first order low pass filter with a frequency equal to 2c/dn.
Abstract:
Method of noise reduction in a hearing aid or a listening device to be used by a hearing impaired person in which the noise reduction is provided primarily in the frequency range wherein the hearing impaired has the smallest hearing loss or the best hearing.
Abstract:
A system and method to provide maximum power over large angular sectors using an array of transducers is disclosed. Array segments of transducers and phase shifters form a beam from each of the array segments, wherein the set of beams overlap to form a large sector coverage beam. The phase of each of the signals fed to the radiating elements is shifted, such that the difference between beam point directions of the beams of two adjacent array segments is substantially equal to one half of the sum of the beamwidths of the beams of the two adjacent array segments. The phase of each of the signals fed to the radiating elements may also be shifted in proportion to the square of the distance between one end of the array of transducers and the position of each of the plurality of radiating elements.
Abstract:
A microphone array, beam forming method and apparatus using the microphone array, and a method and apparatus for estimating an acoustic source direction using the microphone array are provided. The apparatus for forming constant directivity beams comprising: a microphone array, which is comprised of first through n-th microphone sub-arrays, wherein each of the microphone sub-arrays comprises: a first microphone placed at a predetermined location on a flat plate, which commonly belongs to each of the microphone sub-arrays; and second and third microphones placed at locations perpendicularly spaced by a predetermined segment from a straight line connecting the first microphone and the center of the flat plate, the predetermined segment being determined depending on a target frequency allotted to reach of the microphone sub-arrays, a beam formation unit receiving voice signals output from the first through n-th microphone sub-arrays and generating a beam for each of the first through n-th microphone sub-arrays; a filtering unit filtering the beams output from the beam formation unit; and an adding unit adding the filtered signals output from the filtering unit.
Abstract:
Loudspeaker system having various loudspeakers (SP.sub.i, i=0, 1, 2, . . . , m) which are arranged in accordance with a predetermined pattern and have associated filters (F.sub.i, i=0, 1, 2, . . . , m), which filters all receive an audio signal (AS) and are equipped to transmit output signals to the respective loudspeakers (SP.sub.i) such that they, during operation, generate a sound pattern of a predetermined form, wherein the loudspeakers (SP.sub.i) have a mutual spacing (l.sub.i), which, insofar as physically possible, substantially corresponds to a logarithmic distribution, wherein the minimum spacing is determined by the physical dimensions of the loudspeakers used.
Abstract:
An acoustic signal processing method and system using a pair of spatially separated microphones to obtain the direction or location of speech or other acoustic signals from a common sound source is disclosed. The invention includes a method and apparatus for processing the acoustic signals by determining whether signals acquired during a particular time frame represent the onset or beginning of a sequence of acoustic signals from the sound source, identifying acoustic received signals representative of the sequence of signals, and determining the direction of the source based upon the acoustic received signals. The invention has applications to videoconferencing where it may be desirable to automatically adjust a video camera, such as by aiming the camera in the direction of a person who has begun to speak.