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Небесная энциклопедия

Космические корабли и станции, автоматические КА и методы их проектирования, бортовые комплексы управления, системы и средства жизнеобеспечения, особенности технологии производства ракетно-космических систем

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Мониторинг СМИ

Мониторинг СМИ и социальных сетей. Сканирование интернета, новостных сайтов, специализированных контентных площадок на базе мессенджеров. Гибкие настройки фильтров и первоначальных источников.

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Форма поиска

Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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Применить Всего найдено 11810. Отображено 100.
09-02-2012 дата публикации

Method of processing signal, encoding apparatus thereof, decoding apparatus thereof, and signal processing system

Номер: US20120035939A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A method processing a signal, an encoding apparatus, and a decoding apparatus are provided. The method of processing a signal includes restoring a down-mixed original signal using a re-quantized prediction parameter to generate a restored signal in an encoding apparatus; generating mute information indicating whether the down-mixed original signal has been muted, according to a value of the restored signal; and transmitting the mute information and the down-mixed original signal from the encoding apparatus to a decoding apparatus.

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08-03-2012 дата публикации

Apparatus for determining a spatial output multi-channel audio signal

Номер: US20120057710A1
Принадлежит: Individual

An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

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08-03-2012 дата публикации

Method and apparatus of adjusting distribution of spatial sound energy

Номер: US20120057732A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Provided is a method of adjusting a distribution of spatial sound energy, including storing information associated with a sound transfer function from each of speakers of a speaker array to a position of at least one listener, and information associated with the sound transfer function from each of the speakers of the speaker array to a far-field position, and generating at least two sound beams maximizing a far-field sound pressure attenuation with respect to a source signal, based on information associated with the sound transfer function, in order to form a personal sound zone in the position of the at least one listener.

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22-03-2012 дата публикации

Terminal device, mobile terminal, and navigation program

Номер: US20120069711A1
Принадлежит: Fujitsu Ltd

A terminal device includes an orientation calculating unit that calculates the orientation of a device with respect to the target. Furthermore, the terminal device also includes a degree-of-processing determining unit that determines the degree of processing related to an attribute of a sound that indicates the target in accordance with the orientation calculated by the orientation calculating unit. Furthermore, the terminal device also includes an output control unit that controls an output of a sound in accordance with the degree of processing determined by the degree-of-processing determining unit.

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22-03-2012 дата публикации

Sound output apparatus and method of controlling the same

Номер: US20120070008A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A method of controlling a sound output apparatus is provided. The method includes determining frequency bands respectively corresponding to an audible band and an inaudible band in relation to a sound that is received via the sound output apparatus by referring to hearing characteristics of a user using the sound output apparatus, compensating for a perception degree in relation to the inaudible band by emphasizing information regarding the audible band of the sound based on a result of the determining, and outputting a sound obtained by reflecting the compensating.

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29-03-2012 дата публикации

Apparatus, method, and program product for presenting moving image with sound

Номер: US20120076304A1
Автор: Kaoru Suzuki
Принадлежит: Toshiba Corp

According to one embodiment, an apparatus for presenting a moving image with sound includes an input unit, a setting unit, a main beam former unit, and an output control unit. The input unit inputs data on a moving image with sound including a moving image and a plurality of channels of sounds. The setting unit sets an arrival time difference according to a user operation, the arrival time difference being a difference in time between a plurality of channels of sounds coming from a desired direction. The main beam former unit generates a directional sound in which a sound in a direction having the arrival time difference set by the setting unit is enhanced, from the plurality of channels of sounds included in the data on the moving image with sound. The output control unit outputs the directional sound along with the moving image.

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14-06-2012 дата публикации

Telephone or other device with speaker-based or location-based sound field processing

Номер: US20120150542A1
Автор: Wei Ma
Принадлежит: National Semiconductor Corp

A method includes obtaining audio data representing audio content from at least one speaker. The method also includes spatially processing the audio data to create at least one sound field, where each sound field has a spatial characteristic that is unique to a specific speaker. The method further includes generating the at least one sound field using the processed audio data. The audio data could represent audio content from multiple speakers, and generating the at least one sound field could include generating multiple sound fields around a listener. The spatially processing could include performing beam forming to create multiple directional beams, and generating the multiple sound fields around the listener could include generating the directional beams with different apparent origins around the listener. The method could further include separating the audio data based on speaker, where each sound field is associated with the audio data from one of the speakers.

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05-07-2012 дата публикации

Immersive audio rendering system

Номер: US20120170757A1
Принадлежит: SRS Labs Inc

A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

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16-08-2012 дата публикации

Audio signal of an fm stereo radio receiver by using parametric stereo

Номер: US20120207307A1
Принадлежит: DOLBY INTERNATIONAL AB

The invention relates to an apparatus for improving a stereo audio signal of an FM stereo radio receiver. The apparatus comprises a parametric stereo (PS) parameter estimation stage. The parameter estimation stage is configured to determine one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the apparatus comprises an upmix stage. The upmix stage is configured to generate the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation in a downmix stage. The PS parameter estimation stage may be part of a PS encoder. The upmix stage may be part of a PS decoder.

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06-09-2012 дата публикации

Directional Electroacoustical Transducing

Номер: US20120224729A1
Принадлежит: Bose Corp

A multichannel audio system for radiating sound to a listening area that includes a plurality of listening spaces. The audio system includes directional audio devices, positioned in a first of the listening spaces, close to a head of a listener, for radiating first sound waves corresponding to components of one of the channels and nondirectional audio devices, positioned inside the listening area and outside the listening space, distant from the listening space, for radiating sound waves corresponding to components of a second of the channels.

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13-12-2012 дата публикации

Apparatus and method for extracting a direct/ambience signal from a downmix signal and spatial parametric information

Номер: US20120314876A1

An apparatus for extracting a direct and/or ambience signal from a downmix signal and spatial parametric information, the downmix signal and the spatial parametric information representing a multi-channel audio signal having more channels than the downmix signal, wherein the spatial parametric information has inter-channel relations of the multi-channel audio signal, is described. The apparatus has a direct/ambience estimator and a direct/ambience extractor. The direct/ambience estimator is configured for estimating a level information of a direct portion and/or an ambient portion of the multi-channel audio signal based on the spatial parametric information. The direct/ambience extractor is configured for extracting a direct signal portion and/or an ambient signal portion from the downmix signal based on the estimated level information of the direct portion or the ambient portion.

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13-12-2012 дата публикации

Parametric joint-coding of audio sources

Номер: US20120314879A1
Автор: Christof Faller

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

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10-01-2013 дата публикации

Method and apparatus for reproducing three-dimensional sound

Номер: US20130010969A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Stereophonic sound is reproduced by acquiring image depth information indicating a distance between at least one object in an image signal and a reference location, acquiring sound depth information indicating a distance between at least one sound object in a sound signal and a reference location based on the image depth information, and providing sound perspective to the at least one sound object based on the sound depth information.

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24-01-2013 дата публикации

Binaural decoder to output spatial stereo sound and a decoding method thereof

Номер: US20130022205A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.

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14-02-2013 дата публикации

Sound localization for user in motion

Номер: US20130041648A1
Автор: Steven Osman
Принадлежит: Sony Computer Entertainment Inc

Methods, apparatus, and computer programs for simulating the source of sound are provided. One method includes operations for determining a location in space of the head of a user utilizing face recognition of images of the user. Further, the method includes an operation for determining a sound for two speakers, and an operation for determining an emanating location in space for the sound, each speaker being associated with one ear of the user. The acoustic signals for each speaker are established based on the location in space of the head, the sound, the emanating location in space, and the auditory characteristics of the user. In addition, the acoustic signals are transmitted to the two speakers. When the acoustic signals are played by the two speakers, the acoustic signals simulate that the sound originated at the emanating location in space.

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14-03-2013 дата публикации

Camera Driven Audio Spatialization

Номер: US20130064376A1
Принадлежит: Intel Corp

A camera input can be used by the computer to support audio spatialization or to improve audio spatialization of an application that already supports it. A computer system may to support audio spatialization, for example, by modifying the relative latency or relative amplitude of the rendered audio packets. If a sound is intended, for example, to be located on the left side of the user, then the audio channel that is rendered on the headset speaker located on the user's left ear may have a somewhat decreased latency and increased amplitude compared to the other audio channel.

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09-05-2013 дата публикации

Method and apparatus for estimating interchannel delay of sound signal

Номер: US20130114817A1
Принадлежит: Huawei Technologies Co Ltd

A method and an apparatus for estimating an interchannel delay of a sound signal are disclosed, related to the communication field and capable of realizing a stable sound field in a crosstalk. The method includes: calculating an error between an actual interchannel phase difference and a predicted interchannel phase difference of a sound signal, where the predicted interchannel phase difference is predicted according to a predetermined interchannel delay of the sound signal; determining whether the sound signal is a sound signal in a crosstalk according to the error; and if the sound signal is a sound signal in the crosstalk, setting an interchannel delay corresponding to the sound signal to a fixed value

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04-07-2013 дата публикации

Front wave field synthesis (wfs) system and method for providing surround sound using 7.1 channel codec

Номер: US20130170652A1

Provided is a system and method for front wave field synthesis (WFS) to provide a surround sound with a reduced number of loudspeakers. An apparatus for encoding a front WFS signal in the front WFS system may include a content receiving unit to receive WFS content including the first channels corresponding to a front array speaker and the second channels corresponding to a rear surround sound speaker, and an encoding unit to encode the first channels and the second channels using a multichannel encoder corresponding to the number of the channels included in the WFS content.

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04-07-2013 дата публикации

Hearing aid with improved localization

Номер: US20130170680A1
Принадлежит: GN Resound AS

A BTE hearing aid includes a BTE hearing aid housing, at least one BTE sound input transducer, a processor configured to generate a hearing loss compensated output signal, a sound signal transmission member for transmission of a signal from a sound output of the BTE hearing aid housing to an ear canal of a user at a second end of the sound signal transmission member, an earpiece configured to be inserted in the ear canal, an output transducer, and an ITE microphone housing accommodating at least one ITE microphone, wherein the ITE microphone housing is configured to be positioned in an outer ear, wherein the processor is further configured for processing an audio signal from the at least one ITE microphone and an audio signal from the at least one BTE sound input transducer in such a way that the hearing loss compensated output signal substantially preserves spatial cues.

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11-07-2013 дата публикации

Apparatus

Номер: US20130177168A1
Автор: Kai Inha, Peter Sladen
Принадлежит: Nokia Oyj

An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform determining a change in position and/or orientation of an apparatus, and processing at least two audio signals dependent on the change in position and/or orientation to generate at least one output signal wherein the processing of the two audio signals dependent on the change in position and/or orientation produces the output signal comprising a representation of acoustic energy from a first direction.

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11-07-2013 дата публикации

Modular audio systems and related assemblies and methods

Номер: US20130177195A1
Принадлежит: Skullcandy Inc

Modular audio systems comprise two speaker assemblies and a wiring system. Each speaker assembly may comprise a speaker and an audio jack integral to each speaker assembly. The wiring system comprises a first wiring assembly comprising two audio jacks configured to connect to the audio jacks of the two speaker assemblies and two wires connected to the two audio jacks at first ends of the two wires. Headphone assemblies may comprise two speaker assemblies and a headband configured for removable attachment to the speaker assemblies. Each speaker assembly may comprise an attachment structure configured for attachment to another device or structure, wherein the attachment structure of each speaker assembly of the two speaker assemblies comprises a frustoconical surface and two first attachment features comprising elongated features on the frustoconical surface.

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01-08-2013 дата публикации

Multi-Channel Audio Processing

Номер: US20130195276A1
Автор: Pasi Ojala
Принадлежит: Nokia Oyj

A method including: receiving at least a first input audio channel and a second input audio channel; and using an inter-channel prediction model to form at least an inter-channel direction of reception parameter.

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08-08-2013 дата публикации

Controllable Playback System Offering Hierarchical Playback Options

Номер: US20130202114A1
Принадлежит: Nokia Oyj

A first apparatus performs the following: determining, using at least two microphone signals corresponding to left and right microphone signals and using at least one further microphone signal, directional information of the left and right microphone signals; outputting a first signal corresponding to the left microphone signal; outputting a second signal corresponding to the right microphone signal; and outputting a third signal corresponding to the determined directional information. Another apparatus performs the following: performing at least one of the following: outputting first and second signals as stereo output signals; or converting the first and second signals to mid and side signals, and converting, using directional information for the first and second signals, the mid and side signals to at least one of binaural signals or multi-channel signals, and outputting the corresponding binaural signals or multi-channel signals. Additional apparatus, program products, and methods are disclosed.

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22-08-2013 дата публикации

Menu navigation method for user of audio headphones

Номер: US20130216045A1

A method of operating an audio system having first and second sources includes outputting a first audio signal from the first source on first and second speakers. In response to a listener input, the first audio signal and a second audio signal from the second source are output simultaneously on the first and second speakers. The first signal is output from each of the first and second speakers such that the first signal is perceived by a listener to originate from a first direction toward the listener. The second signal is output from each of the first and second speakers such that the second signal is perceived by the listener to originate from a second direction toward the listener. The second direction is offset at least ninety degrees from the first direction.

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05-09-2013 дата публикации

Method of adjusting a hearing apparatus with the aid of the sensory memory

Номер: US20130230182A1
Принадлежит: Siemens Medical Instruments Pte Ltd

Individual adjustment of a hearing apparatus, and in particular a hearing aid, is improved by testing the auditory sensory memory of a user of the hearing apparatus. At least one test result is obtained and the hearing apparatus is adjusted in dependence on the at least one test result. The test person must detect, perceive and be able to reproduce changes in a signal.

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12-09-2013 дата публикации

Augmented reality (ar) audio with position and action triggered virtual sound effects

Номер: US20130236040A1
Принадлежит: Disney Enterprises Inc

An augmented reality (AR) audio system for augmenting environment or ambient sound with sounds from a virtual speaker or sound source positioned at a location in the space surrounding an AR participant. The sound from the virtual speaker may be triggered by an action of the listener and/or by the location or relative orientation of the listener. The AR audio system includes stereo earphones receiving an augmented audio track from a control unit, and binaural microphones are provided to capture ambient sounds. The control unit operates to process trigger signals and retrieve one or more augmentation sounds. The control unit uses an AR audio mixer to combine the ambient sound from the microphones with the augmentation sounds to generate left and right ear augmented audio or binaural audio, which may be modified for acoustic effects of the environment including virtual objects in the environment or virtual characteristics of real objects.

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19-09-2013 дата публикации

Sound processing apparatus

Номер: US20130243211A1
Принадлежит: Yamaha Corp

In a sound processing apparatus, a likelihood calculation unit calculates an in-region coefficient and an out-of-region coefficient indicating likelihood of generation of each frequency component of a sound signal inside and outside a target localization range, respectively, according to localization of each frequency component. A reverberation analysis unit calculates a reverberation index value according to the ratio of a reverberation component for each frequency component. A coefficient setting unit generates a process coefficient for suppressing or emphasizing a reverberation component generated inside or outside the target localization range, for each frequency component of the sound signal, on the basis of the in-region coefficient, the out-of-region coefficient and the reverberation index value. A signal processing unit applies the process coefficient of each frequency component to each frequency component of the sound signal.

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17-10-2013 дата публикации

Audio system and method of operation therefor

Номер: US20130272096A1
Принадлежит: Koninklijke Philips NV

An audio system comprises an ultrasound sensor array ( 105 ) which has a plurality of ultrasound sensor elements, and an audio band array ( 101 ) comprising a plurality of audio band elements. The same array of wideband audio transducers may be used for both the ultrasound sensor array ( 105 ) and the audio band array ( 101 ). An estimator ( 107 ) generates a presence characteristic of a user in response to ultrasound signals received from the ultrasound sensor array. The presence characteristic may specifically comprise a position estimate for the user. An audio array circuit ( 103 ) generates a directional response for the audio band array ( 101 ) by applying weights to individual audio band signals for the audio band elements. A weight circuit ( 109 ) determines the weights in response to the presence characteristic. The system may provide improved adaptation of the directivity of the audio band array ( 101 ) and specifically does not require the sound source in the audio band to be active for adaptation.

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09-01-2014 дата публикации

Method for generating and consuming 3-d audio scene with extended spatiality of sound source

Номер: US20140010372A1

A method of generating and consuming 3D audio scene with extended spatiality of sound source describes the shape and size attributes of the sound source. The method includes the steps of: generating audio object; and generating 3D audio scene description information including attributes of the sound source of the audio object.

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06-01-2022 дата публикации

ACTIVE NOISE CONTROL SYSTEM

Номер: US20220007109A1
Принадлежит:

Adaptive operations of a first noise control system and a second noise control system may include a speaker that outputs noise cancellation sound, a microphone that detects an error signal, an auxiliary filter that generates, from a noise signal, a correction signal that corrects the error signal so that a difference in a position between the microphone and a noise cancellation position is compensated, and an adaptive filter that performs an adaptive operation using the corrected error signal to generate the noise cancellation sound from the noise signal are alternately performed. A transfer function learned in a state in which the second noise control system is stopped is set in the auxiliary filter of the first noise control system, and a transfer function learned in a state in which the adaptive operation of the first noise control system is stopped is set in the auxiliary filter of the second noise control system. 1. An active noise control system that reduces noise , comprising:a plurality of noise control systems; anda controller,wherein each noise control system of the plurality of noise control systems includes one or more noise control lines, a speaker that is configured to output noise cancellation sound,', 'a microphone that is configured to detect an error signal,', 'an auxiliary filter corresponding to a noise cancellation position, and', 'an adaptive filter,, 'wherein each noise control line includeswherein the auxiliary filter of each noise control line is configured to generate, from a noise signal representing the noise, a correction signal that corrects the error signal detected by the microphone so that a difference between the noise cancellation position corresponding to the auxiliary filter and a position of the microphone is compensated,wherein the adaptive filter of each noise control line is configured to perform adaptive operations using each error signal corrected by the correction signals generated by the auxiliary filter of each noise ...

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06-01-2022 дата публикации

METHODS AND APPARATUS TO GENERATE SPATIAL AUDIO BASED ON COMPUTER VISION

Номер: US20220007127A1
Автор: Thomas Tigi
Принадлежит:

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source. 1. An apparatus comprising:at least one memory;instructions in the apparatus; and determine a position of an audio source based on an image generated via a camera; and', 'apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source., 'processor circuitry to execute the instructions to2. The apparatus of claim 1 , wherein the processor circuitry is to determine the position of the audio source as an azimuth angle.3. The apparatus of claim 1 , wherein the audio source is a person speaking.4. The apparatus of claim 3 , wherein the processor circuitry is to:detect a face of the person speaking in the image; andgenerate a bounding box of the face of the person speaking in the image.5. The apparatus of claim 4 , wherein the processor circuitry is to:determine a lateral location within the image of the person speaking based on a location of the bounding box; anddetermine a proximity of the person speaking with respect to the camera based on a size of the bounding box.6. The apparatus of claim 5 , wherein the processor circuitry is to determine the position of the person speaking based on the lateral location and the proximity.7. The apparatus of claim 5 , wherein the processor circuitry is to determine the proximity of the person speaking to be on a near plane claim 5 , a medium plane claim 5 , or a far plane.8. The apparatus of claim 1 , wherein the microphone is located at the audio source.9. The apparatus of claim 1 , wherein the audio ...

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06-01-2022 дата публикации

METHOD, SYSTEM AND COMPUTER PROGRAM PRODUCT FOR RECORDING AND INTERPOLATION OF AMBISONIC SOUND FIELDS

Номер: US20220007128A1
Принадлежит:

A method of recording ambisonic sound fields with a spatially distributed plurality of ambisonic microphones comprising a step of recording sound signals from plurality of ambisonic microphones a step of converting recorded sound signals to ambisonic sound fields and a step of interpolation of the ambisonic sound fields according to the invention comprises a step of generating synchronizing signals for particular ambisonic microphones for synchronized recording of sound signals from plurality of ambisonic microphones and during the step of interpolation of the ambisonic sound fields it includes filtering sound signals from particular microphones with individual filters having a distance-dependent impulse response having a cut-off frequency f(d) depending on distance dbetween point of interpolation and m-th microphone applying gradual distance dependent attenuation applying re-balancing with amplification of 0ordered ambisonic component and attenuating remaining ambisonic components. Invention further concerns recording system and computer program product. 1. A method of recording and interpolation of ambisonic sound field with a spatially distributed plurality of ambisonic microphones comprising a step of recording sound signals from a plurality of ambisonic microphones , a step of converting recorded sound signals to an ambisonic sound fields , and a step of interpolation of the ambisonic sound fields , the step of recording further comprises a step of generating synchronizing signals for particular ambisonic microphones for synchronized recording of sound signals from a plurality of ambisonic microphones and the step of interpolation of the ambisonic sound fields includes:{'sub': c', 'm', 'm, 'filtering ambisonic fields from particular microphones with individual filter having a distance-dependent impulse response having a cut-off frequency f(d) depending on distance dbetween point of interpolation and m-th microphone,'}applying gradual distance dependent ...

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05-01-2017 дата публикации

Method of constructing street guidance information database, and street guidance apparatus and method using street guidance information database

Номер: US20170003132A1

A walking guidance apparatus using a walking guidance information database includes a feature point extracting unit configured to extract a feature point from an acquired image, a corresponding point search unit configured to search for a corresponding point based on a correspondence relationship between feature points extracted from consecutive images, a current position and walking direction calculation unit configured to calculate a current position and a walking direction of a pedestrian by calculating a 3D position and pose of a camera between the images by using camera internal parameters and a relationship between corresponding points, a guidance information generating unit configured to generate guidance information according to the current position and the walking direction of the pedestrian, and a guidance sound source reproducing unit configured to reproduce a guidance sound source corresponding to the guidance information in 3D based on the current position and the walking direction of the pedestrian.

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02-01-2020 дата публикации

USER AUTHENTICATION METHOD AND USER AUTHENTICATION DEVICE

Номер: US20200004944A1
Автор: Yamashita Kosei
Принадлежит:

Provided is a user authentication method including reproducing sound data of which a sound source in a first position of a space around a user is virtually localized using a Head-Related Transfer Function (HRTF) of the user toward the user, acquiring a second position of the space around the user, the second position being estimated by the user who has listened to the reproduced sound data as a position of the sound source; and authenticating the user according to a coincidence between the first position and the second position. 1. A user authentication method , comprising: the sound source is in a first position of a space around a user,', 'the sound source is virtually localized, based on a Head-Related Transfer Function (HRTF) of the user convoluted with the sound data, and', 'the sound source is virtually localized by use of at least one of an earphone, a headphone, or a transaural system;, 'reproducing sound data of a sound source, wherein'}acquiring, as a user input, a second position of the space around the user, wherein the user input is based on the reproduced sound data; andauthenticating the user based on at least one coincidence between the first position and the second position.2. The user authentication method according to claim 1 , further comprising acquiring the user input from a terminal device of the user.3. The user authentication method according to claim 1 , wherein the sound data is reproduced binaurally.4. The user authentication method according to claim 1 , wherein the first position is from a position group in the space around the user.5. The user authentication method according to claim 4 , whereinthe position group includes a plurality of positions, andeach position of the plurality of positions has a different height.6. The user authentication method according to claim 4 , wherein the first position is in front of the user.7. The user authentication method according to claim 4 , wherein the position group further includes the second ...

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07-01-2016 дата публикации

Methods for Audio Signal Transient Detection and Decorrelation Control

Номер: US20160005405A1
Принадлежит: Dolby Laboratories Licensing Corp

Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

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07-01-2016 дата публикации

Methods for Controlling the Inter-Channel Coherence of Upmixed Audio Signals

Номер: US20160005406A1

Audio characteristics of audio data corresponding to a plurality of audio channels may be determined. The audio characteristics may include spatial parameter data. Decorrelation filtering processes for the audio data may be based, at least in part, on the audio characteristics. The decorrelation filtering processes may cause a specific inter-decorrelation signal coherence (“IDC”) between channel-specific decorrelation signals for at least one pair of channels. The channel-specific decorrelation signals may be received and/or determined. Inter-channel coherence (“ICC”) between a plurality of audio channel pairs may be controlled. Controlling ICC may involve at receiving an ICC value and/or determining an ICC value based, at least partially, on the spatial parameter data. A set of IDC values may be based, at least partially, on the set of ICC values. A set of channel-specific decorrelation signals, corresponding with the set of IDC values, may be synthesized by performing operations on the filtered audio data. 1100-. (canceled)101. A method , comprising:receiving audio data corresponding to a plurality of audio channels;determining audio characteristics of the audio data, the audio characteristics including spatial parameter data and at least one of tonality information or transient information;determining at least two channel-specific decorrelation filtering processes for the audio data based, at least in part, on the tonality information or the transient information, the channel-specific decorrelation filtering processes causing a specific inter-decorrelation signal coherence (“IDC”), which is a measure of correlation between decorrelation signals, between channel-specific decorrelation signals for at least one pair of channels, each of the channel-specific decorrelation filtering processes comprising applying a decorrelation filter to at least a portion of a corresponding audio channel of the audio data to produce filtered audio data, the channel-specific ...

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07-01-2016 дата публикации

Methods for Parametric Multi-Channel Encoding

Номер: US20160005407A1
Принадлежит: DOLBY INTERNATIONAL AB

The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system () configured to generate a bitstream () indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system () comprises a downmix processing unit () configured to generate the downmix signal from a multi-channel input signal (); wherein the downmix signal comprises m channels and wherein the multi-channel input signal () comprises n channels; n, m being integers with m Подробнее

04-01-2018 дата публикации

Method for decoding a higher order ambisonics (hoa) representation of a sound or soundfield

Номер: US20180005641A1
Принадлежит: Dolby Laboratories Licensing Corp

When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (β e ) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to β e =┌log 2 (┌log 2 (√{square root over ( K MAX )}·O)┐+ 1 )┐.

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02-01-2020 дата публикации

USER APPARATUS AND METHOD OF OPERATING SAME

Номер: US20200005608A1
Автор: KIM Yong Joo
Принадлежит:

Disclosed are a user apparatus and a method of operating the same. The user apparatus includes a warning element management device that obtains location information of a warning element generated based on game data, a sensor that senses a rotation of the user apparatus to generate rotation angle information, a corrector that corrects the location information of the warning element by using the rotation angle information, and a sound source processor that binaurally renders a sound source by using the location information of the warning element or the corrected location information. 1. A user apparatus comprising:a warning element management device configured to obtain location information of a warning element generated based on game data;a sensor configured to sense a rotation of the user apparatus to generate rotation angle information;a corrector configured to correct the location information of the warning element by using the rotation angle information; anda sound source processor configured to binaurally render a sound source by using the location information of the warning element or the corrected location information.2. The user apparatus of claim 1 , further comprising:an output device configured to output the binaurally rendered sound source.3. The user apparatus of claim 2 , further comprising:a vibration generating device configured to generate a vibration to the user apparatus.4. The user apparatus of claim 3 , wherein the warning element management device is configured to compare the location information of the warning element and the rotation angle information and control the vibration generating device based on a comparison result after the binaurally rendered sound source is output.5. The user apparatus of claim 4 , wherein the warning element management device is configured to control the vibration generating device to generate a vibration when a difference between a location of the warning element corresponding to the location information of the ...

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13-01-2022 дата публикации

Apparatus And Method To Provide Situational Awareness Using Positional Sensors And Virtual Acoustic Modeling

Номер: US20220014865A1
Принадлежит:

A situation awareness system is provided and includes sensors configured to sense an object and generate signals of the sensed object. An identification and classification module is in communication with the sensors. The identification and classification module is configured to identify the sensed object and determine the position, speed and direction of each sensed object. A sound synthesis module is in communication with the identification and classification module. The sound synthesis module is configured to create sounds for the sensed object and further configured to create a binaural virtual acoustic model for the sensed object. The binaural virtual acoustic model is used to position the sensed object in virtual space at a location corresponding to its location in real space. One or more beam forming arrays is in communication with the sound synthesis module and configured to delivery sound. The delivered sound increases the situational awareness of a vehicle operator. 1. A situation awareness system configured to create a binaural virtual acoustic model of a sensed object , the situation awareness system comprising:one or more sensors configured to sense an object and generate signals of the sensed object;an identification and classification module in communication with the one or more sensors, the identification and classification module configured to identify the sensed object and further configured to determine the position, speed and direction of each sensed object;a sound synthesis module in communication with the identification and classification module, the sound synthesis module configured to create sounds for the sensed object and further configured to create a binaural virtual acoustic model for the sensed object, the binaural virtual acoustic model being used to position the sensed object in virtual space at a location corresponding to its location in real space; andone or more beam forming arrays in communication with the sound synthesis module ...

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02-01-2020 дата публикации

INTEGRATED RECONSTRUCTION AND RENDERING OF AUDIO SIGNALS

Номер: US20200005801A1
Принадлежит: DOLBY INTERNATIONAL AB

A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output. 1. A method for rendering an audio output based on an audio data stream , comprising: M audio signals which are combinations of N audio objects, wherein N>1 and M≤N,', {'sub': 'i', 'side information including a series of reconstruction instances cof a reconstruction matrix C and first timing data defining transitions between said instances, said side information allowing reconstruction of the N audio objects from the M audio signals, and'}, {'sub': 'i', 'time-variable object metadata including a series of metadata instances mdefining spatial relationships between the N audio objects and second timing data defining transitions between said metadata instances;'}], 'receiving a data stream including{'sub': sync', 'sync', 'i', 'i, 'generating a synchronized rendering matrix Rbased on the object metadata, the first timing data, and information relating to a current playback system configuration, said synchronized rendering matrix Rhaving a rendering instance rcorresponding in time with each reconstruction instance c;'}{'sub': i', 'i, 'multiplying each reconstruction instance ...

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05-01-2017 дата публикации

Method and apparatus for higher order ambisonics encoding and decoding using singular value decomposition

Номер: US20170006401A1
Принадлежит: DOLBY INTERNATIONAL AB

The encoding and decoding of HOA signals using Singular Value Decomposition includes forming ( 11 ) based on sound source direction values and an Ambisonics order corresponding ket vectors (|(Ω5))) of spherical harmonics and an encoder mode matrix (Ξ 0χs ). From the audio input signal (|χ(Ω s ))) a singular threshold value (σ ε ) determined. On the encoder mode matrix a Singular Value Decomposition ( 13 ) is carried out in order to get related singular values which are compared with the threshold value, leading to a final encoder mode matrix rank ( r fin e ). Based on direction values (Ω ι ) of loudspeakers and a decoder Ambisonics order (N ι ), corresponding ket vectors (IY(Ω ι ) ) and a decoder mode matrix (Ψ 0χL ) are formed ( 18 ). On the decoder mode matrix a Singular Value Decomposition ( 19 ) is carried out, providing a final decoder mode matrix rank ( r fin d ). From the final encoder and decoder mode matrix ranks a final mode matrix rank is determined, and from this final mode matrix rank and the encoder side Singular Value Decomposition an adjoint pseudo inverse (Ξ + ) † of the encoder mode matrix (Ξ 0χs ) and an Ambisonics ket vector (Ia′ s ) are calculated. The number of components of the Ambisonics ket vector is reduced ( 16 ) according to the final mode matrix rank so as to provide an adapted Ambisonics ket vector (|a′ ι ). From the adapted Ambisonics ket vector, the output values of the decoder side Singular Value Decomposition and the final mode matrix rank an adjoint decoder mode matrix (Ψ) † is calculated ( 15 ), resulting in a ket vector (|y(Ω ι )

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05-01-2017 дата публикации

Apparatus and Method for Estimating an Overall Mixing Time Based on at Least a First Pair of Room Impulse Responses, as well as Corresponding Computer Program

Номер: US20170006403A1
Принадлежит:

An apparatus for estimating an overall mixing time, where the apparatus comprises a processing element configured to determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses and a second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses, set a sample time of the plurality of sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value, and determine the overall mixing time based on the mixing time for the first pair of room impulse responses. 1. An apparatus for estimating an overall mixing time based on at least a first pair of room impulse responses , comprising a processing element configured to:determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses and a second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses;set a sample time of the plurality of different sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value; anddetermine the overall mixing time based on the mixing time for the first pair of room impulse responses.2. The apparatus according to claim 1 , wherein the apparatus is configured to estimate the overall mixing time based on a plurality of pairs of room impulse responses claim 1 , wherein the first pair of room impulse responses being comprised in the plurality of pairs of room ...

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07-01-2016 дата публикации

Converting Multi-Microphone Captured Signals To Shifted Signals Useful For Binaural Signal Processing And Use Thereof

Номер: US20160007131A1
Принадлежит:

A method includes, estimating directional information based on multiple input channel signals representing at least one arriving sound from a sound source captured by respective multiple microphones that have respective known locations relative to each other, wherein said estimating comprises finding a time delay that removes a time difference between said first and second input channel signals; deriving a mid-signal and a side signal on a basis of a first input channel signal, a second input channel signal and said estimated directional information; and generating an output signal comprising a plurality of output channels using said mid-signal, said side signal and said estimated directional information such that the output signal retains a spatial representation of the captured at least one arriving sound. Apparatus and program products are also disclosed. 1. A method comprising:estimating directional information based on multiple input channel signals representing at least one arriving sound from a sound source captured by respective multiple microphones that have respective known locations relative to each other, wherein said estimating comprises finding a time delay that removes a time difference between said first and second input channel signals;deriving a mid-signal and a side signal on a basis of a first input channel signal, a second input channel signal and said estimated directional information; andgenerating an output signal comprising a plurality of output channels using said mid-signal, said side signal and said estimated directional information such that the output signal retains a spatial representation of the captured at least one arriving sound.2. The method as claimed in claim 1 , wherein said deriving comprises; deriving the mid-signal as a sum of one of said first and second input channel signals shifted by said time delay and the other one of said first and second input channel signals; and deriving the side signal as a difference between the ...

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07-01-2016 дата публикации

REDUCING CORRELATION BETWEEN HIGHER ORDER AMBISONIC (HOA) BACKGROUND CHANNELS

Номер: US20160007132A1
Принадлежит:

In general, techniques are described for compression and decoding of audio data are generally disclosed. An example device for compressing audio data includes one or more processors configured to apply a decorrelation transform to ambient ambisonic coefficients to obtain a decorrelated representation of the ambient ambisonic coefficients, the ambient HOA coefficients having been extracted from a plurality of higher order ambisonic coefficients and representative of a background component of a soundfield described by the plurality of higher order ambisonic coefficients, wherein at least one of the plurality of higher order ambisonic coefficients is associated with a spherical basis function having an order greater than one. 1. A method comprising:obtaining a decorrelated representation of ambient ambisonic coefficients having at least a left signal and a right signal, the ambient ambisonic coefficients having been extracted from a plurality of higher order ambisonic coefficients and representative of a background component of a soundfield described by the plurality of higher order ambisonic coefficients, wherein at least one of the plurality of higher order ambisonic coefficients is associated with a spherical basis function having an order greater than one; andgenerating a speaker feed based on the decorrelated representation of the ambient ambisonic coefficients.2. The method of claim 1 , further comprising applying a recorrelation transform to the decorrelated representation of the ambient ambisonic coefficients to obtain a plurality of correlated ambient ambisonic coefficients.3. The method of claim 2 , wherein applying the recorrelation transform comprises applying an inverse phase-based transform to the ambient ambisonic coefficients.4. The method of claim 3 , wherein the inverse phase-based transform has been normalized according to one of N3D (full three-D) normalization.5. The method of claim 3 , wherein the inverse phase-based transform has been normalized ...

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07-01-2021 дата публикации

ADAPTING AUDIO STREAMS FOR RENDERING

Номер: US20210006918A1
Принадлежит:

In general, techniques are described for adapting audio streams for rendering. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a plurality of audio streams that include one or more sub-streams. The one or more processors may determine, based on the plurality of audio streams, a total number of the one or more sub-streams for all of the plurality of audio streams, and adapt, when the total number of the sub-streams is greater than a render threshold, the plurality of audio streams to decrease the number of the one or more sub-streams and obtain an adapted plurality of audio streams. The one or more processors may also apply the renderer to the adapted plurality of audio streams to obtain the one or more speaker feeds, and output the one or more speaker feeds to one or more speakers. 1. A device configured to play one or more of a plurality of audio streams , the device comprising:a memory configured to store a plurality of audio streams, each of the plurality of audio streams representative of a soundfield and include one or more sub-streams; andone or more processors coupled to the memory, and configured to:determine, based on the plurality of audio streams, a total number of the one or more sub-streams for all of the plurality of audio streams;adapt, when the total number of the one or more sub-streams is greater than a render threshold indicative of a total number of sub-streams a renderer supports when rendering the plurality of audio streams to one or more speaker feeds, the plurality of audio streams to decrease the number of the one or more sub-streams and obtain an adapted plurality of audio streams including a reduced total number of the one or more sub-streams that is equal to or less than the render threshold;apply the renderer to the adapted plurality of audio streams to obtain the one or more speaker feeds; andoutput the one or more speaker feeds to one or more speakers.2. The ...

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07-01-2021 дата публикации

COMPUTING SYSTEM FOR BINAURAL AMBISONICS DECODING

Номер: US20210006923A1
Принадлежит:

A computing system that facilitates decoding a spherical harmonics (SH) representation of a three-dimensional sound signal to a binaural sound signal is described herein. The computing system generates a binaural sound signal based upon the SH representation, a tapering window function that is selected based on an SH encoding order of the SH representation, and a coloration compensation filter that incorporates the tapering window function. The computing system causes the binaural sound signal to be played over at least two speakers. 1. A computing system , comprising:a processor; and accessing a spherical harmonics (SH) representation of a three-dimensional sound signal, wherein the SH representation is of an SH encoding order;', a tapering window function that is determined by the SH encoding order of the SH representation; and', 'a coloration compensation filter, wherein the coloration compensation filter incorporates the tapering window function,, 'generating a binaural sound signal from the SH representation based upon], 'memory storing instructions that, when executed by the processor, cause the processor to perform acts comprising 'causing the first signal of the binaural sound signal to be played over a first speaker and the second signal of the binaural sound signal to be played over a second speaker.', 'wherein the binaural sound signal comprises a first signal and a second signal; and'}2. The computing system of claim 1 , wherein the tapering window function is one of:a Hann tapering window function;a Hamming tapering window function;a Blackman tapering window function;a Nuttal tapering window function;a Blackman-Nuttal tapering window function; ora Blackman-Harris tapering window function.3. The computing system of claim 1 , wherein generating the binaural sound signal comprises:applying the tapering window function to the SH representation to generate an intermediate SH representation of the three-dimensional sound signal;applying head-related transfer ...

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07-01-2021 дата публикации

ELECTRONIC DEVICE, METHOD AND COMPUTER PROGRAM

Номер: US20210006926A1
Принадлежит: SONY CORPORATION

An electronic device for a vehicle comprising circuitry configured to obtain information about a planned maneuver of the vehicle; and circuitry configured to determine the position and/or orientation of a sound field based on the information about the planned maneuver of the vehicle. 1. An electronic device for a vehicle comprisingcircuitry configured to obtain information about a planned maneuver of the vehicle; andcircuitry configured to determine the position and/or orientation of a sound field based on the information about the planned maneuver of the vehicle.2. The electronic device of claim 1 , wherein the circuitry is configured to determine the position and/or orientation of the sound field by moving one or more virtual sound sources based on information about the planned maneuver of a vehicle.3. The electronic device of claim 1 , wherein the circuitry is configured to anticipate the planned maneuver by moving the sound field in accordance with the planned maneuver.4. The electronic device of claim 1 , wherein the information about the planned maneuver of the vehicle comprises a braking force claim 1 , a steering angle claim 1 , a velocity and/or accelerating parameter.5. The electronic device of claim 1 , wherein the circuitry is configured to anticipate the planned maneuver by tilting the sound field based on an inclination angle in accordance with the planned maneuver.6. The electronic device of claim 1 , wherein the circuitry is configured to anticipate the planned maneuver by moving the sound field horizontally in accordance with the planned maneuver.7. The electronic device of claim 1 , wherein the circuitry is configured to obtain the information about the planned maneuver from an autonomous driving section.8. The electronic device of claim 2 , wherein the circuitry is configured to render the sound field according to the determined positions the virtual sound sources.9. The electronic device of claim 1 , wherein the circuitry is configured to produce ...

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07-01-2021 дата публикации

SOUND OUTPUT DEVICE, SOUND GENERATION METHOD, AND PROGRAM

Номер: US20210006927A1
Принадлежит: SONY CORPORATION

According to the present disclosure, a sound output device includes: a sound acquisition part configured to acquire sound to be output to the other end of a sound guide part, one end of which is arranged near an entrance of an ear canal of a listener, the sound guide part having a hollow structure; and a head-related transfer function adjustment part configured to adjust a head-related transfer function of sound captured by the sound guide part. Since the head-related transfer function adjustment part adjusts the head-related transfer function of sound captured by the sound guide part, it is possible to listen to both ambient sound and sound provided from a sound output device such that the listener does not feel strangeness even in the state in which the listener is wearing the sound output device. 120-. (canceled)21. A sound output device comprising:a support configured to fit the sound output device to and support the sound device from an intertragic notch of an ear of a listener without hanging from a top of the ear; and acquire sound to be output to a first end of a sound guide; and', 'adjust a head-related transfer function of sound captured by the sound guide,', 'wherein the sound guide comprises a second end that is closer to the intertragic notch than the first end of the sound guide is,', 'wherein the support is configured to suspend the first end of the sound guide behind a lobe of the ear,', 'wherein the sound guide has a hollow structure, and', 'wherein the hollow structure of the sound guide curves around an axis parallel to an ear canal of the listener., 'at least one processor configured to22. The sound output device according to claim 21 , wherein the at least one processor is further configured to:adjust a sound environment of sound captured by the sound guide.23. The sound output device according to claim 21 ,wherein the head-related transfer function is adjusted such that a location of sound source of the sound is localized at a place different ...

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07-01-2021 дата публикации

Audio Processing

Номер: US20210006929A1
Автор: Laaksonen Lasse
Принадлежит:

A method for rendering a spatial audio signal that represents a sound field in a selectable viewpoint audio environment that includes one or more audio objects associated with respective audio content and a respective position in the audio environment. The method includes receiving an indication of a selected listening position and orientation in the audio environment; detecting an interaction concerning a first audio object on basis of one or more predefined interaction criteria; modifying the first audio object and one or more further audio objects linked thereto; and deriving the spatial audio signal that includes at least audio content associated with the modified first audio object in a first spatial position of the sound field that corresponds to its position in the audio environment in relation to said selected listening position and orientation, and audio content associated with the modified one or more further audio objects. 1. A method for rendering a spatial audio signal that represents a sound field in a selectable viewpoint audio environment that includes one or more audio objects , wherein each audio object is associated with respective audio content and a respective position in the audio environment , the method comprising:receiving an indication of a selected listening position and orientation in the audio environment;detecting an interaction concerning a first audio object on basis of one or more predefined interaction criteria;modifying, in response to said detected interaction, the first audio object and one or more further audio objects linked thereto; andderiving the spatial audio signal that includes at leastaudio content associated with the modified first audio object in a first spatial position of the sound field that corresponds to its position in the audio environment in relation to said selected listening position and orientation, andaudio content associated with the modified one or more further audio objects in respective further spatial ...

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04-01-2018 дата публикации

In-Ear Headphone For Gaming, High Fidelity Music and 3D Effect

Номер: US20180007461A1
Автор: Wun Tack Ching
Принадлежит:

A multi-purpose earphone having a vocal enhancement at 150 Hz and 2.4 kHz and having an inline dongle mechanism for toggling sound effects for playing games, watching movies, and listening to high fidelity musics. 1. An in-ear headphone for multi-purpose uses , having a left speaker for the left ear of a human and a right speaker for the right ear of a human , comprising:a circuitry for boosting output audio signals both at 150 Hz frequency and at 2.4 KHz frequency; andan inline dongle mechanism configured on an input cable for sending input signals into a specific processing circuitry.2. The in-ear headphone of claim 1 , wherein said circuitry for boosting output audio signals comprises two boosting filters.3. The in-ear headphone of claim 2 , wherein said circuitry for boosting output audio signals further comprises two TL-082 or its equivalent amplifiers.4. The in-ear headphone of claim 1 , further comprising a circuitry for decoding 24 bit/192 kHz digital signals claim 1 , a dongle in said inline dongle mechanism being configured to send high-fidelity audio digital signals into said circuitry for decoding 24 bit/192 kHz digital signals for processing and generating input signals for said circuitry for boosting.5. The in-ear headphone of claim 4 , wherein said circuitry for decoding 24 bit/192 kHz digital signals comprises an OTK5262 chip or an equivalent type of chip in combination with a PCM1793 DAC chip or an equivalent type of chip.6. The in-ear headphone of further comprising a circuitry for cross-feeding between a left input channel and a right input channel and for adjusting phase shift with HRTF claim 1 , producing simulation of 3D surround sound signals; and a dongle in said inline dongle mechanism being configured to send audio signals into said circuitry for cross-feeding for 3D simulation output.7. The in-ear headphone of claim 6 , wherein said circuitry for cross-feeding comprises a PT2387 chip or its equivalent.8. The in-ear headphone of claim 1 , ...

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04-01-2018 дата публикации

AUDIO PROVIDING APPARATUS AND AUDIO PROVIDING METHOD

Номер: US20180007483A1
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

An audio providing apparatus and method are provided. The audio providing apparatus includes: an object renderer configured to render an object audio signal based on geometric information regarding the object audio signal; a channel renderer configured to render an audio signal having a first channel number into an audio signal having a second channel number; and a mixer configured to mix the rendered object audio signal with the audio signal having the second channel number. 1. An audio providing method comprising:receiving a plurality of input channel signals;aligning a difference in phase between correlated input channel signals among the plurality of input channel signals; andconverting the plurality of input channel signals including the correlated input channel signals into a plurality of output channel signals based on an input layout and an output layout,wherein the input layout is a format of the plurality of input channel signals and the output layout is a format of the plurality of output channel signals.2. The method of claim 1 , wherein the output layout is 2D layout.3. The method of claim 1 , wherein the plurality of output channel signals include a virtual output channel signal to reproduce a height input channel signal.4. The method of claim 1 , wherein the plurality of input channel signals comprise information for determining whether to perform virtual 3D rendering on a specific frame. This is a continuation of U.S. application Ser. No. 14/649,824 filed on Jun. 4, 2015, which is a National Stage application under 35 U.S.C. §371 of PCT/KR2013/011182, filed on Dec. 4, 2013, which claims the benefit of U.S. Provisional Application No. 61/732,938, filed on Dec. 4, 2012 in the United States Patent and Trademark Office, and U.S. Provisional Application No. 61/732,939, filed on Dec. 4, 2012 in the United States Patent and Trademark Office, all the disclosures of which are incorporated herein in their entireties by reference.Apparatuses and methods ...

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04-01-2018 дата публикации

METHOD FOR DECODING A HIGHER ORDER AMBISONICS (HOA) REPRESENTATION OF A SOUND OR SOUNDFIELD

Номер: US20180007484A1

When compressing an HOA data frame representation, a gain control () is applied for each channel signal before it is perceptually encoded (). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (β) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to 1. A method for decoding a compressed Higher Order Ambisonics (HOA) sound representation of a sound or sound field , the method comprising:receiving a bit stream containing the compressed HOA representation, and{'sub': e', 'e, 'claim-text': {'br': None, 'sub': e', '2', '2', 'MAX, 'i': 'K', 'β=┌log(┌log(√{square root over ()}·O)┐+1)┐.'}, 'decoding the compressed HOA representation based on a lowest integer number β, wherein the lowest integer number βis determined based on'}{'sub': MAX', '1≦N≦N', {'sub2': 'MAX'}, '1', 'O', 'MAX', '1', 'O', '2, 'sup': (N)', '(N)', '(N)', '(N)', '2', '2, 'wherein K=maxK(Ω, . . . , Ω), N is an order of the compressed HOA representation, Nis a maximum order of interest of the compressed HOA representation, Ω, . . . , Ωare directions of virtual loudspeakers, O=(N+1)is a number of HOA coefficient sequences, and K is a ratio between the squared Euclidean norm ∥Ψ∥of a mode matrix and O,'}{'sub': 'MAX', 'wherein e>0, and'}{'sub': 'MAX', 'wherein √{square root over (K)}=1.5.'}2. An apparatus for decoding a compressed Higher Order Ambisonics (HOA) sound representation of a sound or sound field , the apparatus comprising:{'sub': 'e', 'a processor configured to receive a bit stream containing the compressed HOA representation, the processor further configured to ...

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04-01-2018 дата публикации

SOUND SOURCE RENDERING IN VIRTUAL ENVIRONMENT

Номер: US20180007488A1
Принадлежит:

Embodiments are directed to sound source rendering in a virtual reality (VR) system that executes an immersive virtual environment including at least one virtual sound source. A motion sensor to produces a measurement of a motion of the user, and an imaging sensor is used to produce an indication of at least one physical feature of the user relevant to sound perception. A sound rendering engine determines and applies a head-related transfer function (HRTF) based on the at least one physical feature of the user, and effects a source direction of sound from the at least one virtual sound source according to a frame of reference of the user based on the motion of the user. 1. A system for sound source rendering in a virtual reality (VR) system , the system comprising:a modeling engine to execute an immersive virtual environment (VE) that includes at least one virtual sound source;a motion assessor to read an output of a motion sensor to produce a measurement of a motion of the user during execution of the VE;a physical feature assessor to read an output of an imaging sensor and produce an indication of at least one physical feature of the user relevant to sound perception by the user, wherein the physical feature assessor is to read an output of an imaging sensor and produce an indication of at least one physical feature of the user during execution of the VE;a sound rendering engine to determine and apply a head-related transfer function (HRTF) based on the at least one physical feature of the user, and to effect a source direction of sound from the at least one virtual sound source according to a frame of reference of the user based on the motion of the user during execution of the VE; anda sound output device to produce a user-perceptible sound from the virtual sound source based on an output of the sound rendering engine during execution of the VE, the sound having directional properties based on the HRTF and source direction.2. The system of claim 1 , further ...

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04-01-2018 дата публикации

Spatial audio processing

Номер: US20180007490A1
Принадлежит:

An apparatus comprising at least one processor and at least one memory, the memory comprising machine-readable instructions, that when executed cause the apparatus to: store in a non-volatile memory multiple sets of predetermined spatial audio processing parameters for differently moving sound sources; provide in a man machine interface an option for a user to select one of the stored multiple sets of predetermined spatial audio processing parameters for differently moving sound sources; and in response to the user selecting one of the stored multiple sets of predetermined spatial audio processing parameters for differently moving sound sources, the apparatus is further caused to use the selected one of the stored multiple sets of predetermined spatial audio processing parameters to spatially process audio from one or more sound sources. 1. An apparatus comprising at least one processor and at least one memory , the memory comprising machine-readable instructions , that when executed cause the apparatus to:store in a non-volatile memory multiple sets of predetermined spatial audio processing parameters for differently moving sound sources;provide in a man machine interface an option for a user to select one of the stored multiple sets of predetermined spatial audio processing parameters for differently moving sound sources; andin response to the user selecting one of the stored multiple sets of predetermined spatial audio processing parameters for differently moving sound sources, the apparatus is further caused to use the selected one of the stored multiple sets of predetermined spatial audio processing parameters to spatially process audio from one or more sound sources.2. An apparatus as claimed in claim 1 , wherein each set of predetermined spatial audio processing parameters for differently moving sound sources comprises one or more parameters that change relatively.3. An apparatus as claimed in claim 1 ,wherein storing in a non-volatile memory multiple sets of ...

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03-01-2019 дата публикации

Switching Binaural Sound

Номер: US20190007776A1
Принадлежит:

A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound. 120.-. (canceled)21. A method executed by one or more electronic devices in a computer system to switch binaural sound to one of stereo sound and mono sound during an electronic communication between a person and a user , the method comprising:executing, by the one or more electronic devices in the computer system, the electronic communication that provides a voice of the user in binaural sound to the person such that the voice of the user in the binaural sound externally localizes to the person to a sound localization point (SLP) that is at least three feet away from a head of the person;determining, by the one or more electronic devices in the computer system during the electronic communication, when an object enters an area of the SLP;switching, by the one or more electronic devices in the computer system during the electronic communication, the binaural sound to the one of stereo sound and mono sound when the object enters the area of the SLP; andproviding, by the one or more electronic devices in the computer system during the electronic communication and in response to the switching, the voice of the user to the person in the one of stereo sound and mono sound.22. The method of claim 21 , further comprising:determining, by the one or more electronic devices in the computer system during the electronic communication, when a packet loss is above a threshold; andswitching, by the one or more electronic devices in the computer system during the electronic communication and in response to the determining that the packet loss is above the threshold, the binaural sound to the one of stereo sound and mono sound.23. The method of claim 21 , wherein ...

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03-01-2019 дата публикации

Audio processing device and method and program

Номер: US20190007783A1
Принадлежит: Sony Corp

The present technology relates to an audio processing device and method and a program, in which sound can be more efficiently reproduced. An audio processing device includes: a matrix generation unit which generates a vector for each time-frequency with a head-related transfer function obtained by spherical harmonic transform by spherical harmonics as an element by using only the element corresponding to a degree of the spherical harmonics determined for the time-frequency or on the basis of the element common to all users and the element dependent on an individual user; and a head-related transfer function synthesis unit which generates a headphone drive signal of a time-frequency domain by synthesizing an input signal of a spherical harmonic domain and the generated vector. The present technology can be applied to an audio processing device.

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02-01-2020 дата публикации

ACTIVE NOISE CANCELLATION SYSTEM

Номер: US20200007984A1
Принадлежит:

An active noise cancellation system for reducing unwanted noise in a target area by attenuating a disturbance noise signal (d(n)), which is the remaining noise in the target area originated from an ambient noise signal (x(n)) present in the vicinity of the target area that is transferred to the target area via a main path described by a transfer function P(z)), the active noise cancellation system including a processing unit that implements an ANC-controller which is configured to provide a control signal (y′(n)) for controlling a speaker in the target area in order to generate an acoustic signal (y(n)) that destructively overlaps with the disturbance noise signal (d(n)) and thereby attenuates the same. 13004005002222183004005003104105102022310410510. An active noise cancellation system ( , , ) for reducing unwanted noise in a target area () by attenuating a disturbance noise signal (d(n)) , which is the remaining noise in the target area () originated from an ambient noise signal (x(n)) present in the vicinity of the target area () that is transferred to the target area via a main path described by a transfer function (P(z)) , the active noise cancellation system ( , , ) comprising a processing unit that implements an ANC-controller ( , , ) which is configured to provide a control signal (y′(n)) for controlling a speaker () in the target area () in order to generate an acoustic signal (y(n)) that destructively overlaps with the disturbance noise signal (d(n)) and thereby attenuates the same , wherein the control signal (y′(n)) is transferred into the acoustic signal (y(n)) via the secondary path described by the transfer function (S(z)) , and wherein the ANC-controller provides a system transfer function (H(z)) , which minimizes a residual error signal (e(n)) , wherein the residual error signal (e(n)) represents the difference between the acoustic signal (y(n)) and the disturbance noise signal (d(n)) after a destructive overlap of the same , wherein the ANC- ...

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02-01-2020 дата публикации

METHOD AND APPARATUS FOR GENERATING 3D AUDIO CONTENT FROM TWO-CHANNEL STEREO CONTENT

Номер: US20200008001A1

For generating 3D audio content from a two-channel stereo signal, the stereo signal (x(t)) is partitioned into overlapping sample blocks and is transformed into time-frequency domain. From the stereo signal directional and ambient signal components are separated, wherein the estimated directions of the directional components are changed by a predetermined factor, wherein, if changes are within a predetermined interval, they are combined in order to form a directional centre channel object signal. For the other directions an encoding to Higher Order Ambisonics HOA is performed. Additional ambient signal channels are generated by de-correlation and rating by gain factors, followed by encoding to HOA. The directional HOA signals and the ambient HOA signals are combined, and the combined HOA signal and the centre channel object signals are transformed to time domain. 1. A method for determining 3D audio scene and object based content from two-channel stereo based content , comprising:receiving the two-channel stereo based content, wherein the two-channel stereo based content is represented by at least a time/frequency (T/F) tile;determining, for each T/F tile, ambient power, direct power, source directions and mixing coefficients of a corresponding T/F tile;determining, for each T/F tile, a directional signal and at least an ambient T/F channel based on the ambient power, the direct power, and the mixing coefficients of the corresponding T/F tile;determining the 3D audio scene and the object based content based on the directional signal and the ambient T/F channel.2. The method of claim 1 , wherein claim 1 , for each T/F tile claim 1 , a new source direction is determined based on the source direction claim 1 , and claim 1 ,when there is a determination that the new source direction is within a predetermined interval, a directional center channel object signal is determined based on the directional signal, the directional center channel object signal corresponding to ...

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20-01-2022 дата публикации

Multi-device audio adjustment coordination

Номер: US20220021978A1
Принадлежит: Apple Inc

This relates to intelligent automated assistants and, more specifically, to the intelligent coordination of audio signal output adjustments among multiple electronic devices. An example method includes, generating a local audio intent object associated with a software application stored on a first electronic device, the local audio intent object including one or more local audio parameters; determining that a second electronic device that is outputting an audio signal is proximate to the first electronic device; generating a proximate audio intent object corresponding to the second electronic device based on the one or more local audio adjustment parameters and a round-trip time (RTT) of a communication connection between the first electronic device and the second electronic device; and transmitting the proximate audio intent object to the second electronic device via the communication connection, wherein the proximate audio intent object causes the second electronic device to adjust the output of the audio signal.

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27-01-2022 дата публикации

METHOD OF RENDERING ONE OR MORE CAPTURED AUDIO SOUNDFIELDS TO A LISTENER

Номер: US20220030370A1

A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener. 19-. (canceled)10. A method for binaurally rendering a soundfield , the method comprising:receiving at least a virtual loudspeaker feed representing the soundfield;transforming the virtual loudspeaker feed representing the soundfield into a stereo binaural output signal, wherein the transforming comprises applying a time delay to the virtual speaker loudfeed and wherein the transforming further comprises applying a gain prior to the time delay; andrendering the stereo binaural output signal.11. The method of claim 10 , wherein transforming the virtual loudspeaker feed further comprises varying claim 10 , with a plurality of gain blocks coupled to a plurality of time delay blocks claim 10 , a plurality of gains related to the stereo binaural output signal.12. The method of claim 11 , wherein the plurality of time delay blocks and the plurality of gain blocks are configured to add energy related to a reflected energy field of the stereo binaural output signal.13. The method of claim 12 , wherein the reflected energy field corresponds to a rear energy field and a side energy field.14. The method of claim 10 , wherein the soundfield is a B-Format soundfield.15. The method of claim 10 , wherein the soundfield is an Ambisonics soundfield.16. The method ...

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27-01-2022 дата публикации

Method of Processing Audio Signal and Audio Signal Processing Apparatus

Номер: US20220030374A1
Принадлежит:

A method of processing an audio signal includes inputting an audio signal, obtaining position information of a sound source of the audio signal, receiving an input of movement information, moving a position of the sound source around a predetermined axis, based on the movement information, changing a sound image localization position of the sound source, and performing localization processing on the audio signal so as to localize a sound image of the sound source at a changed sound image localization position. 1. A method of processing an audio signal , the method comprising:inputting an audio signal;obtaining position information of a sound source of the audio signal;receiving of the input of movement information;moving a position of the sound source around a predetermined axis, based on the movement information;changing a sound image localization position of the sound source; andperforming localization processing on the audio signal so as to localize a sound image of the sound source at a changed sound image localization position.2. The method according to claim 1 , wherein:the receiving an input of movement information includes receiving an input of three-dimensional movement information; andthe changing of the sound image localization position includes moving the position of the sound source, based on the three-dimensional movement information.3. The method according to claim 2 , wherein:the movement information includes rotation information; andthe changing of the sound image localization position includes rotating the position of the sound source, based on the rotation information.4. The method according to claim 3 , further comprising arranging the position of the sound source on a predetermined spherical surface and rotating the position of the sound source along the spherical surface.5. The method according to claim 1 , further comprising moving positions of a plurality of sound sources and changing sound image localization positions of the plurality of ...

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12-01-2017 дата публикации

APPARATUS AND METHOD FOR SURROUND AUDIO SIGNAL PROCESSING

Номер: US20170011750A1
Автор: Liu Zongxian, Tanaka Naoya
Принадлежит:

An apparatus for decoding surround audio signal, includes a Bitstream De-multiplexer for unpacking a bitstream into spatial parameters and core parameters, a set of Core Decoder for decoding the core parameters into a set of core signal, a matrix derivation unit for deriving the rendering matrix from the spatial parameters and playback speaker layout information, a renderer for rendering of the decoded core signal to playback signals using the rendering matrix. 1. An apparatus for decoding a surround audio signal , comprising:a Bitstream De-multiplexer for unpacking a bitstream into predominant sound parameters, ambiance parameters, channel assignment parameters and core parameters;a set of Core Decoders for decoding the core parameters into a set of core signals;a predominant sound ambiance switch for assigning the decoded core signal to predominant sound and ambiance according to the channel assignment parameters;a matrix derivation unit for deriving a predominant sound rendering matrix from the predominant sound parameters and playback speaker layout information;a matrix derivation unit for deriving an ambiance rendering matrix from the ambiance parameters and playback speaker layout information;a predominant sound renderer for rendering of the predominant sound to playback signals using the predominant sound rendering matrix;an ambiance renderer for rendering of ambient sound to the playback signals using the ambiance rendering matrix; andan output signal composition unit for composing the playback signals using the rendered predominant sound and the rendered ambient sound.2. An apparatus according to claim 1 , wherein said core decoder corresponds to MPEG-1 Audio Layer III or AAC or HE-AAC or Dolby AC-3 or MPEG USAC standard.3. An apparatus according to claim 1 , wherein said surround audio signal is High Order Ambisonics signal.4. An apparatus according to claim 1 , wherein said spatial parameters comprising of Principal Component Analysis (PCA) or Singular ...

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14-01-2021 дата публикации

SYSTEMS AND METHODS FOR PROVIDING AUDIO TO A USER BASED ON GAZE INPUT

Номер: US20210011682A1
Принадлежит: Tobii AB

According to the invention, a method for providing audio to a user is disclosed. The method may include determining, with an eye tracking device, a gaze point of a user on a display. The method may also include causing, with a computer system, an audio device to produce audio to the user, where content of the audio may be based at least in part on the gaze point of the user on the display. 1. A system for modifying a volume of audio provided to a user gazing at a display , the system comprising:an eye tracking device for at least determining a gaze point of a user on a display; determining audio content associated with a virtual sound source in a virtual environment, wherein the audio content has a first volume and includes one or more virtual sounds in the virtual environment;', 'determining a virtual distance and a virtual direction from the virtual sound source to the gaze point of the user on the display;', 'determining a sound channel produced by an audio device, wherein the sound channel is associated with the virtual distance and the virtual;', 'causing the audio content to be produced to the user, wherein the audio content is produced at the first volume via the sound channel, and the first volume is based at least in part on (i) the gaze point of the user on the display, and (ii) at least one of the virtual distance or the virtual direction;', 'determining a change in the gaze point of the user;', 'determining, based on the changed gaze point, a modified virtual distance and a modified virtual direction from the virtual sound source to the changed gaze point; and', 'causing a change in the first volume of the audio content, wherein the change is based at least in part on the modified virtual distance and the modified virtual direction., 'a processor configured for at least2. The system for modifying the volume of audio provided to the user gazing at the display of claim 1 , the system further comprising:a wearable device, wherein the wearable device is ...

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14-01-2016 дата публикации

Signal processing device, headphone, and signal processing method

Номер: US20160012816A1
Автор: Morishige Fujisawa
Принадлежит: Yamaha Corp

A signal processing device includes: an input unit that accepts an input of a sound-source signal; a sound acquisition unit that acquires ambient sound to generate a sound-acquisition signal; a localization processing unit that processes at least one of the sound-source signal and the sound-acquisition signal so that a first position and a second position are different from each other, and mixes the sound-source signal and the sound-acquisition signal at least one of which is processed, to generate an addition signal, the first position being where a sound image based on the sound-source signal is localized, the second position being where a sound image based on the sound-acquisition signal being localized; and an output unit that outputs the addition signal.

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14-01-2016 дата публикации

AUDIO ENCODER AND DECODER

Номер: US20160012825A1
Принадлежит: DOLBY INTERNATIONAL AB

The present disclosure provides methods, devices and computer program products for encoding and decoding a multi-channel audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and discrete representation of the processed multi-channel audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates. 129-. (canceled)31. The decoding method of wherein the step of combining each of the N waveform-coded downmix signals comprising spectral coefficients corresponding to frequencies between a first and a second cross-over frequency with a corresponding one of the N downmix signals comprising spectral coefficients corresponding to frequencies up to the first cross-over frequency into N combined downmix is performed in a frequency domain.32. The decoding method of claim 30 , wherein the step of extending each of the N combined downmix signals to a frequency range above the second cross-over frequency is performed in a frequency domain.33. The decoding method of claim 30 , wherein the step of combining the M upmix signals comprising spectral coefficients corresponding to frequencies above the first cross-over frequency with the M waveform-coded signals comprising spectral coefficients corresponding to frequencies up to the first cross-over frequency is performed in a frequency domain.34. The decoding method of claim 30 , wherein the step of performing a parametric upmix of the N frequency extended combined downmix signals into M upmix signals is performed in a frequency domain.35. The decoding method of claim 30 , wherein the step of downmixing the M waveform-coded signals into N downmix signals comprising spectral coefficients corresponding to frequencies up to the first cross-over frequency is performed in a frequency domain.36. The decoding method of claim 31 , wherein the frequency domain is a Quadrature Mirror Filters claim 31 , QMF claim 31 , domain.37. The ...

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11-01-2018 дата публикации

DEVICE AND METHOD FOR CALCULATING LOUDSPEAKER SIGNALS FOR A PLURALITY OF LOUDSPEAKERS WHILE USING A DELAY IN THE FREQUENCY DOMAIN

Номер: US20180012612A1
Принадлежит:

A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals. 1. A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources , each audio source comprising an audio signal , said device comprising:a forward transform stage configured to transform each audio signal, block-by-block, to a spectral domain so as acquire for each audio signal a plurality of temporally consecutive short-term spectra;a memory configured to store a plurality of temporally consecutive short-term spectra for each audio signal;a memory access controller configured to access a specific short-term spectrum among the plurality of temporally consecutive short-term spectra for a combination comprising a ...

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10-01-2019 дата публикации

INTER-CHANNEL ENCODING AND DECODING OF MULTIPLE HIGH-BAND AUDIO SIGNALS

Номер: US20190013030A1
Принадлежит:

A device includes an encoder and a transmitter. The encoder is configured to generate a first high-band portion of a mid signal based on a left signal and a right signal, to generate a first synthesized signal based on a first gain and linear predictive coefficient (LPC) parameters, and to generate a second synthesized signal based at least in part on a second gain and the LPC parameters. The encoder is also configured to generate a set of first gain parameters based on a comparison of the first synthesized signal and the mid signal, and to generate a set of adjustment gain parameters based on the second synthesized signal and a high-band non-reference signal. The transmitter is configured to transmit information corresponding to the first high-band portion of the mid signal. The transmitter is also configured to transmit the set of adjustment gain parameters corresponding to the high-band non-reference signal. 1. A device comprising: generate a first high-band portion of a mid signal based on a left signal and a right signal;', 'generate a first synthesized signal based at least in part on a first gain and linear predictive coefficient (LPC) parameters;', 'generate a second synthesized signal based at least in part on a second gain and the LPC parameters;', 'generate a set of first gain parameters based on a comparison of the first synthesized signal and the first high-band portion of the mid signal; and', 'generate a set of adjustment gain parameters based at least in part on the second synthesized signal and a high-band non-reference signal, the high-band non-reference signal corresponding to one of a left high-band portion of the left signal or a right high-band portion of the right signal; and, 'an encoder configured to transmit information corresponding to the first high-band portion of the mid signal, wherein the information includes the LPC parameters, the set of first gain parameters, or a combination thereof; and', 'transmit the set of adjustment gain ...

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12-01-2017 дата публикации

CONTROL APPARATUS, MEASUREMENT SYSTEM, CONTROL METHOD, AND STORAGE MEDIUM

Номер: US20170013389A1
Автор: Kitazawa Kyohei
Принадлежит:

A control apparatus comprises a first obtaining unit configured to obtain a result of sound collection performed by a sound collection apparatus that collects a measurement signal as a sound, the measurement signal being output from an output apparatus; a detection unit configured to detect a position of the output apparatus or the sound collection apparatus relative to a measurement-subject person; and a second obtaining unit configured to obtain a head-related transfer function of the measurement-subject person based on the result of sound collection obtained by the first obtaining unit and the position detected by the detection unit. 1. A control apparatus comprising:a first obtaining unit configured to obtain a result of sound collection performed by a sound collection apparatus that collects a measurement signal as a sound, the measurement signal being output from an output apparatus;a detection unit configured to detect a position of the output apparatus or the sound collection apparatus relative to a measurement-subject person; anda second obtaining unit configured to obtain a head-related transfer function of the measurement-subject person based on the result of sound collection obtained by the first obtaining unit and the position detected by the detection unit.2. The control apparatus according to claim 1 ,wherein the second obtaining unit obtains a head-related transfer function that has been obtained based on the result of sound collection obtained by the first obtaining unit, and that has been compensated based on the position detected by the detection unit.3. The control apparatus according to claim 1 ,wherein the sound collection apparatus is attached to at least one ear of the measurement-subject person, andthe detection unit detects the position of the output apparatus relative to the measurement-subject person.4. The control apparatus according to claim 1 ,wherein the second obtaining unit obtains a head-related transfer function that has been ...

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14-01-2021 дата публикации

DECODING OF AUDIO SCENES

Номер: US20210012781A1
Принадлежит: DOLBY INTERNATIONAL AB

Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder. 1. A method for decoding an audio scene represented by N audio signals , the method comprising:receiving a bit stream comprising L auxiliary signals, M downmix signals and matrix elements of a reconstruction matrix, wherein the matrix elements are transmitted as side information in the bit stream, and wherein at least some of the M downmix signals are formed from two or more of the N audio signals;generating the reconstruction matrix using the matrix elements; and 'wherein M is less than N, and M is equal or greater than one.', 'reconstructing the N audio signals from the M downmix signals and the L auxiliary signals using the reconstruction matrix, wherein approximations of the N audio signals are obtained as linear combinations of the M downmix signals with the matrix elements of the reconstruction matrix as coefficients in the linear combinations,'}2. The method of claim 1 , wherein at least some of the N audio signals are rendered to generate a three-dimensional audio environment.3. The method of claim 1 , wherein the audio scene comprises a three-dimensional audio environment which includes audio elements being associated with positions in a three-dimensional space that can be rendered for playback on an audio system.4. The method of claim 1 , wherein the M downmix signals are arranged in a first portion of a bit stream using a first format and the matrix elements are arranged in a second field of the bit stream using a second format.5. The method of claim 1 , wherein the linear combinations are formed by multiplying a matrix of the M ...

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14-01-2021 дата публикации

Processing of a monophonic signal in a 3D audio decoder, delivering a binaural content

Номер: US20210012782A1
Автор: Grégory Pallone
Принадлежит: ORANGE SA

A method for processing a monophonic signal in a 3D audio decoder, including processing binauralizing decoded signals intended to be delivered spatially by a headset. The method is such that, on detection, in a datastream representative of the monophonic signal, of an indication of non-binauralization processing, which indication is associated with spatial delivery position information, the decoded monophonic signal is directed to a stereophonic rendering engine, which takes into account the position information to construct two delivery channels that are directly processed via a direct mixing that sums these two channels with a binauralized signal output from the binauralization processing, in order to be delivered via the headset. A decoder device that implements the processing method is also provided.

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09-01-2020 дата публикации

EMBEDDING ENHANCED AUDIO TRANSPORTS IN BACKWARD COMPATIBLE AUDIO BITSTREAMS

Номер: US20200013414A1
Принадлежит:

In general, techniques are described by which to embed enhanced audio transports in backward compatible bitstreams. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store the backward compatible bitstream, which conforms to a legacy transport format. The processor(s) may obtain, from the backward compatible bitstream, legacy audio data that conforms to a legacy audio format, and obtain, from the backward compatible bitstream, extended audio data that enhances the legacy audio data. The processor(s) may also obtain, based on the legacy audio data and the extended audio data, enhanced audio data that conforms to an enhanced audio format, and output the enhanced audio data to one or more speakers. 1. A device configured to process a backward compatible bitstream , the device comprising:one or more memories configured to store at least a portion of the backward compatible bitstream, the backward compatible bitstream conforming to a legacy transport format; andone or more processors configured to:obtain, from the backward compatible bitstream, legacy audio data that conforms to a legacy audio format;obtain, from the backward compatible bitstream, extended audio data that enhances the legacy audio data;obtain, based on the legacy audio data and the extended audio data, enhanced audio data that conforms to an enhanced audio format; andoutput the enhanced audio data to one or more speakers.2. The device of claim 1 , wherein the legacy transport format comprises a psychoacoustic codec transport format.3. The device of claim 2 , wherein the psychoacoustic coded transport format comprises an Advanced Audio Coding (AAC) transport format or an AptX transport format.4. The device of claim 1 ,wherein the legacy transport format comprises an Advanced Audio Coding transport format or an AptX transport format,wherein the one or more processors are configured to obtain the enhanced audio data from one or more fill ...

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11-01-2018 дата публикации

METHOD FOR PROCESSING SOUND ON BASIS OF IMAGE INFORMATION, AND CORRESPONDING DEVICE

Номер: US20180014135A1
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

A method of processing an audio signal including at least one audio object based on image information includes: obtaining the audio signal and a current image that corresponds to the audio signal; dividing the current image into at least one block; obtaining motion information of the at least one block; generating index information including information for giving a three-dimensional (3D) effect in at least one direction to the at least one audio object, based on the motion information of the at least one block; and processing the audio object, in order to give the 3D effect in the at least one direction to the audio object, based on the index information. 1. A method of processing an audio signal comprising at least one audio object based on image information , the method comprising:obtaining the audio signal and a current image that corresponds to the audio signal;dividing the current image into at least one block;obtaining motion information of the at least one block;generating index information comprising information for giving a three-dimensional (3D) effect in at least one direction to the at least one audio object, based on the motion information of the at least one block;processing the audio object, in order to give the 3D effect in the at least one direction to the audio object, based on the index information; andoutputting the audio signal including the processed the audio object via a speaker.2. The method of claim 1 , wherein the generating of the index information comprisesobtaining motion information of the current image based on the motion information about the at least one block, and generating the index information based on the motion information of the current image.3. The method of claim 1 , wherein the obtaining of the motion information of the at least one block comprises:determining a block, having a lowest pixel value difference from each block of the current image, from among at least one block that is included in an image that is prior or ...

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11-01-2018 дата публикации

Audio Response Based on User Worn Microphones to Direct or Adapt Program Responses System and Method

Номер: US20180014140A1
Принадлежит: BRAGI GmbH

A system, method, and wireless earpieces for communicating with a virtual reality headset. A position and an orientation of a head of a user are detected utilizing at least wireless earpieces. Audio content is received. The audio content is enhanced utilizing the position and the orientation of the head of the user. The audio content is immediately delivered to the user. The method may further include communicating the position and the orientation of the head of the user to the virtual reality headset. The audio content may be based on the orientation and position of the head of the user.

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10-01-2019 дата публикации

METHOD FOR GENERATING A CUSTOMIZED/PERSONALIZED HEAD RELATED TRANSFER FUNCTION

Номер: US20190014431A1
Автор: HII Desmond, LEE Teck Chee
Принадлежит: CREATIVE TECHNOLOGY LTD

A method is provided for generating a personalized Head Related Transfer Function (HRTF). The method can include capturing an image of an ear using a portable device, auto-scaling the captured image to determine physical geometries of the ear and obtaining a personalized HRTF based on the determined physical geometries of the ear. In addition, a system and a method in association with the system are also provided for customizing audio experience. Customization of audio experience can be based on derivation of at least one customized audio response characteristic which can be applied to an audio device used by a person. Finally, methods and systems are provided for rendering audio over headphones with head tracking enabled by, for example, exploiting efficiencies in creating databases and filters for use in filtering 3D audio sources for more realistic audio rendering and also allowing greater head movement to enhance the spatial audio perception. 1. A method for enhancing audio rendering by generating a customized Head Related Transfer Function (HRTF) , the method comprising:acquiring a captured image of at least one ear of an individual using an image capturing device configured for processing a preliminary image that is a preliminary version of the captured image to provide feedback to a user;generating a set of landmarks that correspond to control points from at least the preliminary image by applying a model to the preliminary image;extracting image based properties from the generated set of landmarks for the individual from a finalized representation of the captured image; andproviding the image based properties to a selection processor configured to select a customized HRTF dataset from a plurality of HRTF datasets that have been determined for a plurality of individuals,wherein at least one of acquiring, generating, extracting, or providing steps comprises of a processing method of a processing apparatus.2. The method as recited in claim 1 , further ...

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10-01-2019 дата публикации

MULTI-CHANNEL BINAURAL RECORDING AND DYNAMIC PLAYBACK

Номер: US20190014432A1
Автор: Eriksen Stein Ove
Принадлежит:

Methods and systems are provided for enhanced audio experiences in VR/AR applications. The apparatuses of this disclosure are adapted to record multiple binaural stereo pairs and play back select binaural pairs corresponding to users' head positions. A substantially spherical microarray is utilized in various embodiments for recording multiple binaural stereo pairs. A VR/AR headset is further adapted to track a user's head positions and dynamically play back binaural sound pairs corresponding to the head positions. 1. A system for surround sound recording and binaural playback corresponding to at least two head positions of a user , comprising:a substantially spherical microphone array adapted to record multiple binaural sound pairs;a multi-channel recording processor adapted to process the multiple binaural sound pairs thereby producing multiple pairs of binaural audio;a multi-channel playback processor adapted to retrieve a head position of the user, and relate coordinates of the multiple pairs of binaural audio to coordinates of the head position thereby reproducing a binaural sound pair corresponding to the head position; anda playback unit adapted to playback the binaural sound pair, wherein the substantially spherical microphone array comprises a substantially spherical frame and a plurality of microphones affixed to the frame, wherein each of the plurality of microphones is paired with a microphone on an orthodrome at its opposite side for recording binaural sound pairs.2. The system of claim 1 , wherein the playback unit is one of a headphone and an earphone.3. The system of claim 2 , wherein the one of a headphone and an earphone is wireless.4. The system of claim 1 , wherein the multi-channel playback processor further comprises a tracking unit adapted to track head positions of a user in a 3-D coordinate system.5. The system of claim 4 , wherein the tracking unit is a virtual reality and augmented reality (VR/AR) headset.6. The system of claim 1 , wherein ...

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10-01-2019 дата публикации

AUDIO PROCESSING APPARATUS, METHOD, AND PROGRAM

Номер: US20190014433A1
Принадлежит: SONY CORPORATION

The present technology relates to an audio processing apparatus, a method, and a program that aim at enabling a sound to be reproduced more efficiently. 1. An audio processing apparatus comprising:a head-related transfer function synthesis section configured to synthesize an input signal in an annular harmonic domain or a portion of an input signal in a spherical harmonic domain corresponding to the annular harmonic domain and a diagonalized head-related transfer function; andan annular harmonic inverse transformation section configured to perform an annular harmonic inverse transformation on a signal obtained by the synthesis on a basis of an annular harmonic function to thereby generate a headphone driving signal in a time frequency domain.2. The audio processing apparatus according to claim 1 , whereinthe head-related transfer function synthesis section calculates a product of a diagonal matrix obtained by diagonalizing a matrix including a plurality of head-related transfer functions by an annular harmonic function transformation and a vector including the input signal corresponding to each order of the annular harmonic function and thereby synthesizes the input signal and the diagonalized head-related transfer function.3. The audio processing apparatus according to claim 2 , whereinthe head-related transfer function synthesis section synthesizes the input signal and the diagonalized head-related transfer function by using only an element of the predetermined order settable for each time frequency in a diagonal component of the diagonal matrix.4. The audio processing apparatus according to claim 2 , whereinthe diagonalized head-related transfer function used in common for users is included as an element in the diagonal matrix.5. The audio processing apparatus according to claim 2 , whereinthe diagonalized head-related transfer function that depends on an individual user is included as an element in the diagonal matrix.6. The audio processing apparatus according ...

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09-01-2020 дата публикации

Electronic Glasses that Display a Virtual Image for a Telephone Call

Номер: US20200014792A1
Принадлежит:

Electronic glasses display a virtual image for a telephone call before a first user wearing the electronic glasses answers the telephone call. The electronic glasses display the virtual image at a location in an environment of the first user where a voice of a second user will externally localize in empty space outside the head of the first user when the first user answers the telephone call. 120.-. (canceled)21. A method comprising:receiving, at a smartphone of a first user located in an environment and from a second user, an incoming telephone call; anddisplaying, on a display of electronic glasses worn on a head of the first user and before the first user answers the incoming telephone call, a virtual image at a location in the environment where a voice of the second user will externally localize in empty space outside the head of the first user in the environment when the first user answers the incoming telephone call.22. The method of further comprising:processing, with a digital signal processor (DSP) in the smartphone, the voice of the second user to generate the voice of the second user in binaural sound that externally localizes in empty space outside the head of the first user in the environment;wirelessly transmitting, the voice processed with the DSP from the smartphone to the electronic glasses; andplaying the voice with speakers in the electronic glasses.23. The method of further comprising:displaying, on the display of the electronic glasses and before the first user answers the incoming telephone call, multiple virtual images in the environment that show different options for locations where the voice of the second user will externally localize in empty space outside the head of the first user when one of the options is selected by the first user.24. The method of further comprising:simultaneously displaying, on the display of the electronic glasses and before the first user answers the incoming telephone call, a first virtual image or a first button ...

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14-01-2021 дата публикации

Method and System for Broadcasting a Multichannel Audio Stream to Terminals of Spectators Attending a Sports Event

Номер: US20210014627A1
Автор: Raphael Blouet, Slim ESSID
Принадлежит: Institut Mines Telecom IMT

The invention relates to a method for broadcasting a spatialized audio stream to terminals (TERM) of spectators attending a sports event. The method comprises the acquisition of a plurality of audio streams constituting a soundscape (SS). The soundscape (SS) is analyzed by a server (SERV) in order for the sound spatialization of the audio streams and of the playback thereof on terminals (TERM), depending both on the localization of the audio flows and also the position of the spectators.

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14-01-2021 дата публикации

SPATIAL AUDIO SIGNAL MANIPULATION

Номер: US20210014628A1
Принадлежит:

Described herein is a method () of rendering an audio signal () for playback in an audio environment () defined by a target loudspeaker system (), the audio signal () including audio data relating to an audio object and associated position data indicative of an object position. Method () includes the initial step () of receiving the audio signal (). At step () loudspeaker layout data for the target loudspeaker system () is received. At step () control data is received that is indicative of a position modification to be applied to the audio object in the audio environment (). At step () in response to the position data, loudspeaker layout data and control data, rendering modification data is generated. Finally, at step () the audio signal () is rendered with the rendering modification data to output the audio signal () with the audio object at a modified object position that is between loudspeakers within the audio environment (). 1. A method of rendering an audio signal for playback in an audio environment defined by a target loudspeaker system , the audio signal including object audio data relating to an audio object and associated object position data indicative of a position of the audio object at a given point in time , the method comprising:receiving the object audio data relating to the audio object;receiving loudspeaker layout data for the target loudspeaker system;receiving object rendering control data indicative of a position modification to be applied, at the given point in time, to the audio object in the audio environment; andrendering the audio object, at the given point in time, in response to the position of the audio object at the given point in time, the loudspeaker layout data, and the object rendering control data, to output the audio object, at the given point in time, at a modified object position that is between loudspeakers within the audio environment, wherein thethe object rendering control data determines a degree of position modification ...

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14-01-2021 дата публикации

Method for generating a customized/personalized head related transfer function

Номер: US20210014629A1
Автор: Desmond Hii, Teck Chee Lee
Принадлежит: CREATIVE TECHNOLOGY LTD

A method is provided for generating a personalized Head Related Transfer Function (HRTF). The method can include capturing an image of an ear using a portable device, auto-scaling the captured image to determine physical geometries of the ear and obtaining a personalized HRTF based on the determined physical geometries of the ear. In addition, a system and a method in association with the system are also provided for customizing audio experience. Customization of audio experience can be based on derivation of at least one customized audio response characteristic which can be applied to an audio device used by a person. Finally, methods and systems are provided for rendering audio over headphones with head tracking enabled by, for example, exploiting efficiencies in creating databases and filters for use in filtering 3D audio sources for more realistic audio rendering and also allowing greater head movement to enhance the spatial audio perception.

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14-01-2021 дата публикации

METHOD FOR DETERMINING LISTENER-SPECIFIC HEAD-RELATED TRANSFER FUNCTIONS

Номер: US20210014631A1
Принадлежит:

A method for determining listener-specific head-related transfer functions is described. The method comprising the steps of: A) providing a visual representation of the head and each of the auricles, wherein for each auricle the visual representation includes visual information of the overall shape of the auricle and of anatomical components of the auricle; B) calculating, using said visual representation, three-dimensional polygon meshes, including a head mesh and independent auricle meshes, which respectively model the shapes of the head and auricles, wherein the auricle meshes () preferably include shape information of auricle components such as the entry () of the ear canal, the concha (), the fossa (), and the backside of the auricle (); C) merging the polygon meshes to a three-dimensional combined mesh, in which the auricle meshes are located at proper locations with respect to the head mesh; D) calculating HRTFs based on the combined mesh. 1. Method for determining head-related transfer functions (HRTFs) , wherein said HRTFs are listener-specific with respect to a specific individual , where said HRTFs correlate with physical characteristics of the individual including the shapes of the individual's head and auricles , the method comprising the steps of:A) providing a visual representation of the head and each of the auricles, wherein the visual representation includes visual information of the overall shape of the auricles and of anatomical components of each of the auricles;B) calculating, using said visual representation, a three-dimensional representation formed by polygon meshes, including auricle meshes and a head mesh which are independent of each other, which respectively model the shapes of the head and auricles, the auricle meshes comprising information about the shape of the mentioned anatomical components of the auricle;C) merging the polygon meshes to a three-dimensional combined mesh, in which the auricle meshes are located at proper locations ...

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09-01-2020 дата публикации

ENERGY-RATIO SIGNALLING AND SYNTHESIS

Номер: US20200015028A1
Принадлежит: NOKIA TECHNOLOGIES OY

An apparatus comprising at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: receive at least one audio signal; obtain, associated with the at least one audio signal over at least one frequency band: at least one spatial audio energy ratio parameter; and at least one remainder energy ratio, wherein a sum of the at least one spatial audio energy ratio parameter and the at least one remainder energy ratio over the frequency band equal a determined value; and control a transmission/storage of the at least one spatial audio energy ratio, and the at least one remainder energy ratio. 1. An apparatus comprising at least one processor and at least one memory including a computer program code , the at least one memory and the computer program code configured to , with the at least one processor , cause the apparatus at least to:receive at least one audio signal;obtain, associated with the at least one audio signal over at least one frequency band: at least one spatial audio energy ratio parameter; and at least one remainder energy ratio, wherein a sum of the at least one spatial audio energy ratio parameter and the at least one remainder energy ratio over the frequency band equal a determined value; andcontrol a transmission/storage of the at least one spatial audio energy ratio, and the at least one remainder energy ratio.2. The apparatus as claimed in claim 1 , wherein the apparatus is further caused to obtain associated with the at least one audio signal over at least one frequency band at least one spatial audio direction parameter claim 1 , wherein the apparatus caused to obtain claim 1 , associated with the at least one audio signal over at least one frequency band: at least one spatial audio energy ratio parameter; and at least one remainder energy ratio is further caused to obtain:at least one first type ...

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21-01-2016 дата публикации

Audio Processing

Номер: US20160019899A1
Принадлежит: DOLBY INTERNATIONAL AB

An audio processing system () for spatial synthesis comprises an upmix stage () receiving a decoded m-channel downmix signal (X) and outputting, based thereon, an n-channel upmix signal (Y), wherein 2≦m Подробнее

21-01-2016 дата публикации

System and Method for Adaptive Audio Signal Generation, Coding and Rendering

Номер: US20160021476A1

Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the sound correctly using the available speaker positions in a room that is set up to play the adaptive audio content.

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21-01-2016 дата публикации

Method and apparatus for facilitating spatial audio capture with multiple devices

Номер: US20160021477A1
Принадлежит: NOKIA TECHNOLOGIES OY

A method, apparatus and computer program product are provided in order to facilitate spatial audio capture by utilizing multiple audio recording devices. In the context of a method, spatial characteristics of sound recorded by a first device are determined. The method also mapping, with a processor, sound recorded by a second device to a location and orientation of the first device. As such, the spatial characteristics of the sound recorded by the second device are modified to match the spatial characteristics of the sound recorded by the first device.

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21-01-2016 дата публикации

ROBUST CROSSTALK CANCELLATION USING A SPEAKER ARRAY

Номер: US20160021480A1
Принадлежит:

An audio receiver that performs crosstalk cancellation using a speaker array is described. The audio receiver detects the location of a listener in a room and processes a piece of sound program content to be output through the speaker array using one or more beam pattern matrices. The beam pattern matrices are generated according to one or more constraints. The constraints may include increasing a right channel and decreasing a left channel at the right ear of the listener, increasing a left channel and decreasing a right channel at the left ear of the listener, and decreasing sound in all other areas of the room. These constraints cause the audio receiver to beam sound primarily towards the listener and not in other areas of the room such that crosstalk cancellation is achieved with minimal effects due to changes to the frequency response of the room. Other embodiments are also described.

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19-01-2017 дата публикации

AUDIO SIGNAL PROCESSING METHOD AND APPARATUS

Номер: US20170019746A1

The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal. 1. A method for processing an audio signal , the method comprising:receiving an input audio signal including a multi-channel signal;receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients;obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; andfiltering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information.2. The method of claim 1 , wherein when BRIR filter coefficients having positional information matching with positional information of a specific channel of the input audio signal are present in a BRIR filter set claim 1 , the vector information indicates the relevant BRIR filter coefficients as BRIR filter coefficients corresponding to the specific channel.3. The method of claim 1 , wherein when BRIR filter coefficients having positional information matching with positional information of a specific channel of the input audio signal are not present in a BRIR filter set claim 1 , the vector information indicates BRIR filter coefficients having a minimum geometric distance from the positional ...

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17-01-2019 дата публикации

AMBISONIC ENCODER FOR A SOUND SOURCE HAVING A PLURALITY OF REFLECTIONS

Номер: US20190019520A1
Автор: BERTHET Pierre
Принадлежит:

An ambisonic encoder for a sound wave has a plurality of reflections. The ambisonic encoder makes it possible to improve the sensation of immersion in a 3D audio scene. The complexity of encoding of the reflections of sound sources for an ambisonic encoder is less than the complexity of encoding of the reflections of sound sources of previously known ambisonic encoders. The ambisonic encoder makes it possible to encode a greater number of reflections of a sound source in real time, and makes it possible to reduce the power consumption related to ambisonic encoding, and to increase the life of a battery of a mobile device used for said application. 1. An ambisonic encoder for a sound wave having a plurality of reflections , comprising:a logic for transforming the frequency of the sound wave;a logic for calculating spherical harmonics of the sound wave and of the plurality of reflections on the basis of a position of a source of the sound wave and positions of obstacles to propagation of the sound wave;a plurality of filtering logics in the frequency domain receiving, as input, spherical harmonics of the plurality of reflections, each filtering logic being parameterized by acoustic coefficients and delays of the reflections;a logic for adding spherical harmonics of the sound wave and outputs from the filtering logic.2. The ambisonic encoder as claimed in claim 1 , wherein the logic for calculating spherical harmonics of the sound wave is configured to calculate the spherical harmonics of the sound wave and of the plurality of reflections on the basis of a fixed position of the source of the sound wave.3. The ambisonic encoder as claimed in claim 1 , wherein the logic for calculating spherical harmonics of the sound wave is configured to iteratively calculate the spherical harmonics of the sound wave and of the plurality of reflections on the basis of successive positions of the source of the sound wave.4. The ambisonic encoder as claimed in claim 1 , wherein each ...

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03-02-2022 дата публикации

LOWER LAYER REPRODUCTION

Номер: US20220038838A1
Принадлежит:

An audio processor for reproducing multichannel audio has a normal layer processing and a lower layer processing. The normal layer processing is configured for processing one or more channels of the object-specific audio or the multichannel audio belonging to a normal layer. The lower layer processing is configured for processing at least one channel or more channels of the object-specific audio or the multichannel audio belonging to a lower layer. The lower layer processing is configured to feed at least one signal belonging to the one channel or the more channels of the lower layer to a subwoofer output. 1. An audio processor for reproducing object-specific audio or multichannel audio , comprising:a normal layer processor configured for processing one or more channels of the object-specific audio or the multichannel audio belonging to a normal layer;a lower layer processor configured for processing at least one channel or more channels of the object-specific audio or the multichannel audio belonging to a lower layer, wherein the normal layer is a listening layer laying within a height of a listener's head as seen from a listener's perspective and wherein the normal layer lies above the lower layer as seen from the listener's perspective;wherein the lower layer processor is configured to feed at least one signal belonging to the one channel or the more channels of the lower layer to a subwoofer output, so as to reproduce the one or more channels of the lower layer using a subwoofer wherein frequencies of a frequency portion of the one or more channels of the lower layer are above 150 Hz or even above 300, 500 or 1000 Hz so as to be high enough to enable a localization of the sound generated by the subwoofer.2. The audio processor according to claim 1 , wherein the lower layer processor is configured to feed at least one signal belonging to a first of the one or more channels of the lower layer to a first subwoofer output and to feed at least another signal ...

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03-02-2022 дата публикации

NEAR-FIELD AUDIO RENDERING

Номер: US20220038840A1
Принадлежит:

Examples of the disclosure describe systems and methods for presenting an audio signal to a user of a wearable head device. According to an example method, a source location corresponding to the audio signal is identified. An acoustic axis corresponding to the audio signal is determined. For each of a respective left and right ear of the user, an angle between the acoustic axis and the respective ear is determined. For each of the respective left and right ear of the user, a virtual speaker position, of a virtual speaker array, is determined, the virtual speaker position collinear with the source location and with a position of the respective ear. The virtual speaker array includes a plurality of virtual speaker positions, each virtual speaker position of the plurality located on the surface of a sphere concentric with the user's head, the sphere having a first radius. For each of the respective left and right ear of the user, a head-related transfer function (HRTF) corresponding to the virtual speaker position and to the respective ear is determined; a source radiation filter is determined based on the determined angle; the audio signal is processed to generate an output audio signal for the respective ear; and the output audio signal is presented to the respective ear of the user via one or more speakers associated with the wearable head device. Processing the audio signal includes applying the HRTF and the source radiation filter to the audio signal. 1. A method of presenting an audio signal to a user of a wearable head device , the method comprising:identifying a source location corresponding to the audio signal;determining an acoustic axis corresponding to the audio signal;determining a reference point; determining an angle between the acoustic axis and the respective ear;', 'determining, of a virtual speaker array, a virtual speaker position substantially collinear with the source location and a position of the respective ear, wherein the virtual speaker array ...

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18-01-2018 дата публикации

HEARING ASSISTANCE SYSTEM

Номер: US20180020298A1
Принадлежит:

There is provided a hearing assistance system, comprising a transmission unit comprising a microphone arrangement for capturing audio signals from a voice of a speaker using the transmission unit and being adapted to transmit the audio signals as radio frequency signal via a wireless RF link; a left ear hearing device and a right ear hearing device, each hearing device being adapted to stimulate the user's hearing and to receive an RF signal from the transmission unit via the wireless RF link and comprising a microphone arrangement for capturing audio signals from ambient sound. 1. A system for providing hearing assistance to a user , comprising:a transmission unit comprising a microphone arrangement for capturing audio signals from a voice of a speaker using the transmission unit and being adapted to transmit the audio signals as radio frequency (RF) signal via a wireless RF link;a left ear hearing device to be worn at or at least partially in the user's left ear and a right ear hearing device to be worn at or at least partially in the user's right ear, each hearing device being adapted to stimulate the user's hearing and to receive an RF signal from the transmission unit via the wireless RF link and comprising a microphone arrangement for capturing audio signals from ambient sound; the hearing devices being adapted to communicate with each other via a binaural link, determining a level of the RF signal received by the left ear hearing device and a level of the RF signal received by the right ear hearing device,', 'determining a level of the audio signal captured by the microphone arrangement of the left hearing device and a level of the audio signal captured by the microphone arrangement of the right hearing device,', 'determining, in at least one frequency band, a phase difference between the audio signal received via the RF link from the transmission unit by the left ear hearing device and the audio signal captured by the microphone arrangement of the left ear ...

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18-01-2018 дата публикации

AUDIO PROCESSING APPARATUS WITH CHANNEL REMAPPER AND OBJECT RENDERER

Номер: US20180020310A1

Embodiments are described for rendering spatial audio content through a system that is configured to reflect audio off of one or more surfaces of a listening environment. The system includes an array of audio drivers distributed around a room, wherein at least one driver of the array of drivers is configured to project sound waves toward one or more surfaces of the listening environment for reflection to a listening area within the listening environment and a renderer configured to receive and process audio streams and one or more metadata sets that are associated with each of the audio streams and that specify a playback location in the listening environment. 1. An audio processing apparatus comprising:an input interface for receiving channel metadata, object metadata, multichannel audio signals, and object audio signals;an audio decoder for decoding the multichannel audio signals to produce decoded multichannel audio signals and for decoding the object audio signals to produce decoded object audio signals;a channel remapper for converting the decoded multichannel audio signals from a transmitted channel configuration to a desired reproduction configuration based on the channel metadata to produce converted multichannel audio signals;an object renderer for rendering the decoded object audio signals based on the object metadata to produce rendered object audio signals, the object metadata comprising a set of parameters, wherein the set of parameters includes spatial position and size; andan output interface for outputting the converted multichannel audio signals or rendered object audio signals.2. The audio processing apparatus of wherein the spatial position is defined by one or more metadata parameters of the object metadata using a spherical coordinate system.3. The audio processing apparatus of wherein the object metadata further includes a parameter representing a width of an audio object.4. The audio processing apparatus of wherein the object metadata further ...

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18-01-2018 дата публикации

Systems and Methods for Spatial Audio Adjustment

Номер: US20180020313A1
Принадлежит:

The present disclosure relates to managing audio signals within a user's perceptible audio environment or soundstage. That is, a computing device may provide audio signals with a particular apparent source location within a user's soundstage. Initially, a first audio signal may be spatially processed so as to be perceivable in a first soundstage zone. In response to determining a high priority notification, the apparent source location of the first audio signal may be moved to a second soundstage zone and an audio signal associated with the notification may be spatially processed so as to be perceivable in the first soundstage zone. In response to determining user speech, the apparent source location of the first audio signal may be moved to a different soundstage zone. 1. A method comprising:driving an audio output device of a computing device with a first audio signal;receiving, via at least one microphone, audio information;determining user speech based on the received audio information; and spatially processing the first audio signal for perception in a soundstage zone; and', 'driving the audio output device with the spatially-processed first audio signal, such that the first audio signal is perceivable in the soundstage zone., 'in response to determining user speech2. The method of claim 1 , wherein receiving the audio information comprises receiving the audio information via a microphone array claim 1 , wherein the method further comprises directing claim 1 , by the microphone array claim 1 , a listening beam toward a user of the computing device claim 1 , wherein determining user speech further comprises determining that a signal-to-noise ratio of the audio information is above a threshold ratio.3. The method of claim 1 , wherein determining user speech comprises analyzing the audio information with a speech recognition algorithm.4. The method of claim 1 , wherein determining user speech comprises analyzing the audio information with a speech recognition ...

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16-01-2020 дата публикации

METHODS, APPARATUS AND SYSTEMS FOR ENCODING AND DECODING OF MULTI-CHANNEL AMBISONICS AUDIO DATA

Номер: US20200020344A1

Conventional audio compression technologies perform a standardized signal transformation, independent of the type of the content. Multi-channel signals are decomposed into their signal components, subsequently quantized and encoded. This is disadvantageous due to lack of knowledge on the characteristics of scene composition, especially for e.g. multi-channel audio or Higher-Order Ambisonics (HOA) content. A method for decoding an encoded bitstream of multi-channel audio data and associated metadata is provided, including transforming the first Ambisonics format of the multi-channel audio data to a second Ambisonics format representation of the multi-channel audio data, wherein the transforming maps the first Ambisonics format of the multi-channel audio data into the second Ambisonics format representation of the multi-channel audio data. A method for encoding multi-channel audio data that includes audio data in an Ambisonics format, wherein the encoding includes transforming the audio data in an Ambisonics format into encoded multi-channel audio data is also provided. 1. A method for decoding an encoded bitstream of multi-channel audio data and associated metadata , the method comprising:decoding the encoded bitstream of multi-channel audio data into multi-channel audio data;detecting that the multi-channel audio data includes a first Ambisonics format; andtransforming the first Ambisonics format of the multi-channel audio data to a second Ambisonics format representation of the multi-channel audio data, wherein the transforming maps the first Ambisonics format of the multi-channel audio data into the second Ambisonics format representation of the multi-channel audio data,wherein the associated metadata further describes re-mixing information and wherein the transforming the first Ambisonics format is based on the re-mixing information indicated by the associated metadata.2. A non-transitory computer program product storing a computer program claim 1 , the computer ...

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16-01-2020 дата публикации

DECODING OF AUDIO SCENES

Номер: US20200020345A1
Принадлежит: DOLBY INTERNATIONAL AB

Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder. 1. A method for decoding an audio scene represented by N audio signals , the method comprising:receiving a bit stream comprising L auxiliary signals, M downmix signals and matrix elements of a reconstruction matrix, wherein the matrix elements are transmitted as side information in the bit stream;generating the reconstruction matrix using the matrix elements; andreconstructing the N audio signals from the M downmix signals and the L auxiliary signals using the reconstruction matrix, wherein approximations of the N audio signals are obtained as linear combinations of the M downmix signals with the matrix elements of the reconstruction matrix as coefficients in the linear combinations,wherein M is less than N, and M is equal or greater than one.2. The method of wherein at least some of the M downmix signals are formed from two or more of the N audio signals.3. The method of wherein at least some of the N audio signals are rendered to generate a three-dimensional audio environment.4. The method of wherein the audio scene comprises a three-dimensional audio environment which includes audio elements being associated with positions in a three-dimensional space that can be rendered for playback on an audio system.5. The method of wherein the M downmix signals are arranged in a first portion of a bit stream using a first format and the matrix elements are arranged in a second field of the bit stream using a second format.6. The method of wherein the linear combinations are formed by multiplying a matrix of the M downmix signals with the reconstruction ...

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17-01-2019 дата публикации

MICROPHONE AND MANUFACTURING METHOD THEREOF

Номер: US20190020961A1
Автор: Yoo Ilseon
Принадлежит:

A microphone includes: a case that is vibrated by a vibration signal, a sound inlet through which a sound signal is input being formed at a portion of the case; a first sound element that is formed in the case at a position corresponding to the sound inlet and receives the sound signal and the vibration signal to output a first initial signal; a second sound element that is formed to be adjacent to the first sound element and receives the vibration signal to output a second initial signal; and a semiconductor chip that is connected to the first sound element and the second sound element and receives the first initial signal and the second initial signal to output a final signal. 1. A microphone comprising:a case that is vibrated by a vibration signal, a sound inlet through which a sound signal is input being formed at a portion of the case;a first sound element that is formed in the case at a position corresponding to the sound inlet and receives the sound signal and the vibration signal to output a first initial signal;a second sound element that is formed to be adjacent to the first sound element and receives the vibration signal to output a second initial signal; anda semiconductor chip that is connected to the first sound element and the second sound element and receives the first initial signal and the second initial signal to output a final signal.2. The microphone of claim 1 , wherein the semiconductor chip: i) divides the first initial signal into a sound signal and a vibration signal claim 1 , ii) modulates a phase of the second initial signal claim 1 , iii) merges the first initial signal with the divided sound signal and vibration signal claim 1 , and iv) merges the second initial signal with the phase-modulated signal to cancel the vibration signal and extract the sound signal.3. The microphone of claim 1 , wherein an air passage is formed at a side of a lower portion of the second sound element.4. The microphone of claim 1 , wherein the case includes:a ...

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17-01-2019 дата публикации

Synthesis of signals for immersive audio playback

Номер: US20190020963A1
Принадлежит:

A method for synthesizing sound includes receiving one or more first inputs (), each including a respective monaural audio track (). One or more second inputs are received, indicating respective three-dimensional (3D) source locations having azimuth and elevation coordinates to be associated with the first inputs. Each of the first inputs is assigned respective left and right filter responses based on filter response functions that depend upon the azimuth and elevation coordinates of the respective 3D source locations. Left and right stereo output signals () are synthesized by applying the respective left and right filter responses to the first inputs. 1. A method for synthesizing sound , comprising:receiving one or more first inputs, each first input comprising a respective monaural audio track;receiving one or more second inputs indicating respective three-dimensional (3D) source locations having azimuth and elevation coordinates to be associated with the first inputs;assigning to each of the first inputs respective left and right filter responses based on filter response functions that depend upon the azimuth and elevation coordinates of the respective 3D source locations; andsynthesizing left and right stereo output signals by applying the respective left and right filter responses to the first inputs.2. The method according to claim 1 , wherein the one or more first inputs comprise a plurality of first inputs claim 1 , and wherein synthesizing the left and right stereo output signals comprises applying the respective left and right filter responses to each of the first inputs to generate respective left and right stereo components claim 1 , and summing the left and right stereo components over all of the first inputs.3. The method according to claim 2 , wherein summing the left and right stereo components comprises applying a limiter to the summed components in order to prevent clipping upon playback of the output signals.4. The method according to claim 1 , ...

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17-01-2019 дата публикации

Sub-band Spatial Audio Enhancement

Номер: US20190020966A1
Автор: Seldess Zachary
Принадлежит:

An audio system provides for spatial enhancement of an audio signal including a left input channel and a right input channel. The system may include a spatial frequency band divider, a spatial frequency band processor, and a spatial frequency band combiner. The spatial frequency band divider processes the left input channel and the right input channel into a spatial component and a nonspatial component. The spatial frequency band processor applies subband gains to subbands of the spatial component to generate an enhanced spatial component, and applies subband gains to subbands of the nonspatial component to generate an enhanced nonspatial component. The spatial frequency band combiner combines the enhanced spatial component and the enhanced nonspatial component into a left output channel and a right output channel. In some embodiments, the spatial component and nonspatial component are separated into spatial subband components and nonspatial subband components for the processing. 1. A method for enhancing an audio signal having a left input channel and a right input channel , comprising:processing the left input channel and the right input channel into a spatial component and a nonspatial component, the spatial component including a difference between the left input channel and the right input channel and the nonspatial component including a sum of the left input channel and the right input channel;applying first subband gains to subbands of the spatial component to generate an enhanced spatial component;applying second subband gains to subbands of the nonspatial component to generate an enhanced nonspatial component; andcombining the enhanced spatial component and the enhanced nonspatial component into a left output channel and a right output channel.2. The method of claim 1 , wherein:processing the left input channel and the right input channel into the spatial component and the nonspatial component includes processing the left input channel and the right input ...

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17-01-2019 дата публикации

Ambisonics sound field navigation using directional decomposition and path distance estimation

Номер: US20190020967A1
Автор: Andrew Allen
Принадлежит: Google LLC

Techniques of rendering sound for a listener involve determining, at a first position, a set of directions from which some specified fraction of the sound field energy originates and reconstruct the sound field at a second position based on the set of directions. Along these lines, a sound rendering computer may receive sound data representing sound emanating from positions on a sphere centered on a microphone. Nevertheless, only a small fraction of these points on the sphere contribute much of the sound field energy directed at the listener. The sound rendering computer determines such directions using a directional decomposition in which those directions having the highest fraction of the sound energy are identified. When the listener moves away from the microphone's position, the sound rendering computer, after identifying such directions, applies an amplitude and phase factor to the sound fields corresponding to those directions.

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