10872results 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.

Acoustic device including a member extending transversely of its thickness and capable of sustaining bending waves at least over an intendedly consequentially acoustically active area of the transverse extent of said member, the member having, by reason of orderly design methodology disclosed and claimed, a distribution of resonant modes of its natural bending wave vibration at least over said area that is dependent on values of particular parameters of said members, including geometrical configuration and directional bending stiffness(es), which values have been selected to predetermine said distribution of natural resonant modes being consonant with required achievable acoustic action of said member for operation of said device over a desired operative acoustic frequency range.

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.

A method of communicating with an electronic device. The method includes providing an electronic device having an audible sound receiving and generating sub-system including a microphone, transmitting from a source at least one acoustic signal encoded with information, receiving said at least one acoustic signal by said microphone and determining a spatial position, distance or movement of the microphone relative to the source, responsive to the received at least one signal.

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 includes multiple microphones to generate audio signals based on sound of a far-field acoustic environment. The apparatus also includes a signal processing system to process the audio signals to generate at least one processed audio signal. The signal processing system is configured to update one or more processing parameters while operating in a first operational mode and is configured to use a static version of the one or more processing parameters while operating in the second operational mode. The apparatus further includes a keyword detection system to perform keyword detection based on the at least one processed audio signal to determine whether the sound includes an utterance corresponding to a keyword and, based on a result of the keyword detection, to send a control signal to the signal processing system to change an operational mode of the signal processing system.

A hearing compensation system for the hearing impaired comprises an input transducer for converting acoustical information at an input to electrical signals at an output, an output transducer for converting electrical signals at an input to acoustical information at an output, a plurality of bandpass filters, each bandpass filter having an input connected to the output of said input transducer, a plurality of AGC circuits, each individual AGC circuit associated with a different one of the bandpass filters and having an input connected to the output of its associated bandpass filter and an output connected to the input of the output transducer. The bandpass filters and AGC circuits may be divided into two processing channels, one for low frequencies and one for high frequencies and may drive separate audio transducers, one configured for maximum efficiency at low frequencies and one configured for maximum efficiency at high frequencies.

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 microphone has a substrate including an acoustically transparent substrate region, a lid with an acoustically transparent lid region, and a membrane which is held by a membrane carrier between the lid and the substrate. The acoustically transparent substrate region or the acoustically transparent lid region is provided with at least one impedance hole sized so that an acoustic impedance of the impedance hole is larger than an acoustic impedance of the acoustically transparent region of the respective other region of substrate region and lid region.

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 telepresence video conference endpoint device includes spaced-apart microphone arrays each configured to transduce sound into corresponding sound signals. A processor receives the sound signals from the arrays and determines a direction-of-arrival (DOA) of sound at each array based on the set of sound signals from that array, determines if each array is blocked or unblocked based on the DOA determined for that array, selects an array among the arrays based on whether each array is determined to be blocked or unblocked, and perform subsequent sound processing based on one or more of the sound signals from the selected array.

An adaptive signal processing system eliminates noise from input signals while retaining desired signal content, such as speech. The resulting low noise output signal delivers improved clarity and intelligibility. The low noise output signal also improves the performance of subsequent signal processing systems, including speech recognition systems. An adaptive beamformer in the signal processing system consistently updates beamforming signal weights in response to changing microphone signal conditions. The adaptive weights emphasize the contribution of high energy microphone signals to the beamformed output signal. In addition, adaptive noise cancellation logic removes residual noise from the beamformed output signal based on a noise estimate derived from the microphone input signals.

A multi-channel audio system is constituted using at least one line array speaker unit, in which plural speakers are arrayed in line, wherein the same audio signal is supplied with a prescribed delay time to each of the speakers, thus forming plural sound beams. The plural sound beams are reflected on a wall surface and a ceiling of a room so as to form plural virtual sound sources surrounding a listening position, and emission directions and intensities of the sound beams are controlled so as to localize a phantom at a prescribed position based on the plural virtual sound sources. By appropriately arranging plural line array speaker units horizontally, vertically, and slantingly in such a way that each line array speaker unit forms sound beams distributed and spread in a sectorial form, it is possible to realize a surround audio system having a high degree of freedom with regard to setup positions for forming virtual sound sources.

Multi-function transducers (130, 700) that are suitable for use in handheld devices such as cellular telephones (100, 900) and are capable of emitting audio, and generating accelerations of sufficient magnitude, and at frequencies that can be sensed by tactile sensation and which exhibit resonant modes that are characterized by center frequencies within a frequency range that can be sensed by tactile sensation, and specifically located between two musical notes on a musical scale are provided. Devices (100, 900) including such transducer are provided, and methods of operating such transducers are provided.

A loudspeaker system is described including an array of electro-acoustic transducers capable of generating steerable beams of sound and additional transducers adapted to reproduce low frequency sound being placed at the perimeter of said array.

Mobile terminals include a housing, an electronic circuit and a speaker assembly positioned in the housing. The speaker assembly includes a first speaker positioned adjacent a first side of the electronic circuit and a second speaker positioned adjacent the first speaker on the first side of the electronic circuit. A grommet may be positioned in the housing that is configured to receive the first speaker and the second speaker and the first speaker and the second speaker are positioned in the grommet. The first speaker can be a receiver speaker and the second speaker can be a micro-speaker.

A microphone comprises a housing defining an inner volume and including a first exterior surface with an aperture leading to the inner volume. The microphone includes a transducing assembly within the housing for converting sound into an electrical signal. A sound inlet plate defines, typically in combination with the first exterior surface, a passageway for transmitting sound to the aperture The passageway receives the sound from an opening in the sound inlet plate. The opening is offset from the location at which the aperture is positioned on the exterior surface. The sound inlet plate is made very thin so that it does not extend substantially away from the housing.