Abstract:
A loudspeaker system for the optimization of sound production so as to achieve limbic and cortical arousal, comprising a resistance-controlled (or partially mass-controlled) woofer system, a mass-controlled (or partially resistance-controlled) midrange system, and a resistance-controlled tweeter system. This system may further comprise crossover networks of a particular configuration. By use of unsymmetrical networks of low order, it is possible to obtain a complete system which exhibits flat delay response.
Abstract:
A plurality of speakers are linked in the present invention. The linked position of each speaker can be detected. Audio signal is input to any one of the master speakers. The master speaker synchronizes the other linked speakers, and supplies audio signals to other speakers. It also controls the delay quantity of the speaker unit of each speaker. For a single speaker, the apparent width of this array speaker system becomes twice the width, and the speaker unit spacing becomes one third the spacing. Consequently, the frequency band at which direction is controllable becomes enhanced.Additionally, a plurality of microphone devices are linked at the top, bottom, left and right sides in the present invention. The linked position of each microphone device can be detected. Audio data is output from each microphone device to the master microphone device. The master microphone device synchronizes with other linked microphone devices, treats them as array microphones in the entire linked array microphone system, and controls the delay quantity of the microphone unit of the microphone device. For a single microphone device, the apparent width of this array microphone system becomes twice the width, and the microphone unit spacing becomes one-third the spacing. Consequently, the frequency band at which direction is controllable becomes enhanced.
Abstract:
An acoustic apparatus without increasing noise etc. even when plural directional microphones collect sounds from a place of the same distances is provided. Sound signals output from the microphone arrays are subjected to phase shift by phase shift circuits 211A to 211H, and the sound signals are combined by an adder 212, The phase shift circuits 211A to 211H performs phase shifts according to installation positions of the microphone arrays. The phase shift circuit 211A makes the shift 0 degree, the phase shift circuit 211B makes the shift 45 degrees, the phase shift circuit 211C makes the shift 90 degrees, and sequentially to the phase shift circuit 211H, the shifts are made according to rotational angles.
Abstract:
A speaker device according to the present invention is a speaker device including a plurality of speaker units arranged in a line when seen from the front side of the speaker device. At least one of intervals between effective vibration regions of adjacent speaker units is set to a predetermined length. The predetermined length is a length that is set such that a difference between a distance from an end of one of the effective vibration regions, which form the at least one of intervals therebetween, to a listening position, and a distance from an end of the other of the effective vibration regions to the listening position can be less than half the shortest wavelength of a reproduced sound of each of the speaker units.
Abstract:
An apparatus and method for generating directional sound are provided to output sound towards a specific sound zone. The apparatus includes an internal loudspeaker array having at least one sound source and edge loudspeakers, each with a sound source having directivity, located at respective ends of the internal loudspeaker array.
Abstract:
In various embodiments, the invention pertains to systems for acoustic beamforming that include one or more speaker membranes, such as, for example, a continuous ribbon membrane, and several independently addressable drivers. Moreover, certain embodiments relate to methods for beamforming with improved directionality.
Abstract:
A voice sound input apparatus, adapted to be inputted a sound and configured to output sound data, includes: a display unit; a first microphone, related to a first sound hole; a second microphone, related to a second sound hole; a signal processing unit; and a microphone holding unit, formed with the first sound hole, and adapted to extend toward a sound source predicted position; wherein a distance between the first sound hole and the second sound hole is a distance that a phase component of a sound strength ratio is lower than or equal to 0 dB, the sound strength ratio being a ratio between a strength of a sound component contained in differential sound pressure of sounds entered to the first sound hole and the second sound hole and a strength of sound pressure of the sound entered to the first sound hole.
Abstract:
An audio apparatus inputs audio signals oriented toward a plurality of positions to speakers which output sound toward the respective positions with narrow directivity, including a table storage section which stores a table for registering the plurality of positions and volume information showing set sound levels of sounds directed toward the positions in correspondence with each other; a signal processing section for adjusting output levels of respective audio signals in accordance with set level control values; and a signal processing control section which reads the volume information for the plurality of positions by reference to the table and which sets, in the signal processing section, the level control values for the audio signals directed to the respective positions in accordance with the read information.
Abstract:
A microphone array system including an input unit to receive sound signals using a plurality of microphones; a frequency splitter splitting each sound signal received into a plurality of narrowband signals; an average spatial covariance matrix estimator using spatial smoothing to obtain a spatial covariance matrix for each frequency component of the sound signal, by which spatial covariance matrices for a plurality of virtual sub-arrays, which are configured in the plurality of microphones, are obtained with respect to each frequency component of the sound signal and an average spatial covariance matrix is calculated; a signal source location detector to detect an incidence angle of the sound signal according to the average spatial covariance matrix calculated; a signal distortion compensator to calculates a weight for each frequency component of the sound signal based on the incidence angle of the sound signal and multiply the calculated weight by each frequency component.
Abstract:
A transducer senses sounds produced by a talker or other source and measures acceleration of air. Enhancement of acceleration is accompanied by reduction of the portion of the sound energy that escapes from the regions around the transducer. The result is a high sensitivity transducer, with increased privacy for use in communication systems, especially cell phones and in a multi-person environment. A pressure sensor array with a weighted output is sensitive to sound from a source talker only, and not to acoustic background noise, and not to a local loudspeaker. The weighted signal is a source sum pressure signal. The array produces a signal (using a different weighting) that corresponds to an estimate of a derivative of pressure. The derivative signal is proportional to the volume velocity fluctuations produced by the source. This signal is enhanced, rather than reduced. A local loudspeaker is driven to make the source sum pressure signal as small as desired. The loudspeaker is driven to produce volume velocity fluctuations approximately equal and opposite to those produced by the source. No compression of air arises due to the talker, and no sound is radiated into the far field. All happens because the system is driven to reduce the source pressure sum signal to below a desired threshold. It is not necessary to directly measure the volume velocity fluctuations of the talker source.