10829results about "Frequency/directions obtaining arrangements" patented technology

The invention enables the capture of sound fields with coding and efficient distribution of electrical signals representing the sound fields for subsequent reproduction in a listening environment utilising pairs of non-equidistant apparatus in various defined configurations such that the acoustic distance as well as direction of sound sources is consistently presented with regard to in front of, behind, beside or below from a listening point anywhere in the listening environment and for any listener orientation, thus making the whole listening area a sweet spot and enabling true shared audio experiences, without the need for worn apparatus. New apparatus and methods for capture, distribution and reproduction or render are also disclosed that enable enhanced perception of captured and reproduced sound for shared experiences with both common and individual capture and reproduction apparatus and new communications and control methods and apparatus for multiple users including heightened perception capability and also improved security aspects are also disclosed.

An apparatus for tracking and identifying objects includes an audio likelihood module which determines corresponding audio likelihoods for each of a plurality of sounds received from corresponding different directions, each audio likelihood indicating a likelihood a sound is an object to be tracked; a video likelihood module which receives a video and determines video likelihoods for each of a plurality of images disposed in corresponding different directions in the video, each video likelihood indicating a likelihood that the image is an object to be tracked; and an identification and tracking module which determines correspondences between the audio likelihoods and the video likelihoods, if a correspondence is determined to exist between one of the audio likelihoods and one of the video likelihoods, identifies and tracks a corresponding one of the objects using each determined pair of audio and video likelihoods.

A sound produced at the location of a listener is captured by a microphone in each of a plurality of speaker devices. A sever apparatus receives an audio signal of the captured sound from all speaker devices, and calculates a distance difference between the distance of the location of the listener to the speaker device closest to the listener and the distance of the listener to each of the plurality of speaker devices. When one of the speaker devices emits a sound, the server apparatus receives an audio signal of the sound captured by and transmitted from each of the other speaker devices. The server apparatus calculates a speaker-to-speaker distance between the speaker device that has emitted the sound and each of the other speaker devices. The server apparatus calculates a layout configuration of the plurality of speaker devices based on the distance difference and the speaker-to-speaker distance.

The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

A sound produced at the location of a listener is captured by a microphone in each of a plurality of speaker devices. A sever apparatus receives an audio signal of the captured sound from all speaker devices, and calculates a distance difference between the distance of the location of the listener to the speaker device closest to the listener and the distance of the listener to each of the plurality of speaker devices. When one of the speaker devices emits a sound, the server apparatus receives an audio signal of the sound captured by and transmitted from each of the other speaker devices. The server apparatus calculates a speaker-to-speaker distance between the speaker device that has emitted the sound and each of the other speaker devices. The server apparatus calculates a layout configuration of the plurality of speaker devices based on the distance difference and the speaker-to-speaker distance.

An encoder includes a sampler that samples an audio signal and that generates from the samples a plurality of short blocks of sampled audio. Each of the short blocks has a duration less than a minimum audibly perceivable signal delay. A processor combines the plurality of short blocks into a long block. The long block is transformed into a frequency domain signal having a plurality of independently modulatable frequency indices. The frequency difference between adjacent indices is determined by the minimum duration and the sampling rate of the sampler. A neighborhood of frequency indices is selected so that the frequency difference between a lowest index and a highest index within the neighborhood is less than a predetermined value. Two or more of the indices are modulated in the neighborhood so as to make a selected one of the indices an extremum while keeping the total energy of the neighborhood constant. A plurality of frequency bands are so coded. A decoder decides that a bit or bits have been received if, in a majority of the frequency bands, the decoder detects a modulated index.

The present invention comprises an ultrasonic resonant rod waveguide-radiator with at least three cylindrical sections, one of which is an entrance section having a planar entrance surface and another of which is an exit section having a planar exit surface, and at least two sections having a variable cross-section. The cylindrical sections and sections of variable cross-section are arranged in alternating fashion and connected to each other acoustically rigidly. The dimensions of the cylindrical sections and the sections of variable cross-section are selected so that the gain of the waveguide-radiator is significantly greater than unity and the strain created by passage of ultrasonic waves through the waveguide-radiator is minimized, increasing the operational life of the waveguide-radiator and maximizing the amount of useful energy transmitted by the waveguide-radiator.

An apparatus comprises microphone receivers (101) which receive microphone signals from a plurality of microphones (103). A comparator (105) determines a speech similarity indication indicative of a similarity between the microphone signal and non-reverberant speech for each microphone signal. The determination is in response to a comparison of a property derived from the microphone signal to a reference property for non-reverberant speech. In some embodiments, the comparator (105) determines the similarity indication by comparing to reference properties for speech samples of a set of non-reverberant speech samples. A generator (107) generates a speech signal by combining the microphone signals in response to the similarity indications. In many embodiments, the apparatus may be distributed over a plurality of devices each containing a microphone, and the approach may determine the most suited microphone for generating the speech signal.

According to the first aspect of the invention, a hearing instrument with at least one microphone and signal processing comprises at least two receivers having a different frequency response. At least a first one of the receivers is placed outside the ear canal, for example in a behind-the-ear component. According to the second aspect of the invention, a hearing instrument comprising a behind-the-ear component and an external component for being placed in the user's ear canal or in the user's ear as well as a connection link between the behind-the-ear component and the external component is provided, where the connection link is reversibly pluggable to the behind-the-ear component and / or the in-the-ear-canal component and has a length that is reversibly adjustable. The hearing instrument also comprises fixation means for reversibly fixing the adjusted length of the connection link. A hearing instrument according to the third aspect comprises a fixation means separate from the in-the-ear-canal component. The fixation means may be positioned in the ear canal and rest therein. The in-the-ear-canal component may be connected to the fixation means and detached therefrom when the same is already in the ear canal. A hearing instrument according to the fourth aspect of the invention comprises an in-the-ear-canal component to be placed in a user's ear canal and fixation means for fixing it in the ear canal. The fixation means comprise an outer shell which is shaped to fit (i.e. custom shaped to fit the specific user's ear geometry) in the user's ear canal and a mounting structure for holding the in-the-ear-canal component. The outer shell is preferably resilient, i.e. has an elasticity allowing temporal deformation.

A method of communicating with an electronic device, comprising: providing a computer having an audible sound receiving and generating sub-system including a microphone; transmitting from a source at least one ultrasonic acoustic signal, encoded with information to the computer; and receiving said at least one signal by said microphone, to be detected by said computer. Preferably, the ultrasonic frequency used is below 50 kHz.

A method and apparatus are disclosed to permit real time, distributed acoustic performance by multiple musicians at remote locations. The latency of the communication channel is reflected in the audio monitor used by the performer. This allows a natural accommodation to be made by the musician. Simultaneous remote acoustic performances are played together at each location, though not necessarily simultaneously at all locations. This allows locations having low latency connections to retain some of their advantage. The amount of induced latency can be overridden by each musician. The method preferably employs a CODEC able to aesthetically synthesize packets missing from the audio stream in real time.

A neural network is provided for recognition and enhancement of multi-channel sound signals received by multiple microphones, which need not be aligned in a linear array in a given environment. Directions and distances of sound sources may also be detected by the neural network without the need for a beamformer connected to the microphones. The neural network may be trained by knowledge gained from free-field array impulse responses obtained in an anechoic chamber, array impulse responses that model simulated environments of different reverberation times, and array impulse responses obtained in actual environments.

A port for a loudspeaker is disclosed which is sufficiently weather resistant for long term outdoor use and which prevents debris, insects and other vermin from entering the loudspeaker and which is compact and efficient.

A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.