Abstract: A system for enabling a shared three-dimensional (“3D”) audio bed available to multiple software applications is provided. The system manages bed metadata defining a number of speaker objects of a 3D audio bed. The bed metadata also associates each speaker object with a location, which in some configurations, is defined by a three-dimensional coordinate system. The bed metadata is communicated to a plurality of applications. The applications can then generate custom 3D audio data that associates individual audio streams with individual speaker objects of the 3D audio bed. The applications can then communicate the custom 3D audio data to a 3D audio bed engine, which causes the processing and rendering of the custom 3D audio data to an output device utilizing a selected spatialization technology. Aspects of the 3D bed can be altered when the spatialization technology or the output device changes.
Type:
Grant
Filed:
June 13, 2017
Date of Patent:
September 17, 2019
Assignee:
Microsoft Technology Licensing, LLC
Inventors:
Robert Norman Heitkamp, Philip Andrew Edry, Ziyad Ibrahim, Paul J. Radek, Steven Wilssens, Brian C. McDowell
Abstract: 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.
Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.
Abstract: Methods, systems, and apparatuses are disclosed for generating a spatial audio format. An input audio source may include one or more individual channels. The one or more individual channels may be designated to be played by a corresponding one or more speakers. The one or more individual channels of the audio source may be separated. The one or more individual tracks may be input into a modeling space representing a multi-dimensional space. The modeling space may include a plurality of emitters at various locations in a vector space. Each of the one or more individual channels may be panned to one or more of the plurality of emitters. The panning may be based on a normalized proximity of the one or more individual channels in the modeling space to the plurality of emitters. The one or more of the plurality of emitters may be encoded into a single multichannel file.
Type:
Grant
Filed:
May 1, 2018
Date of Patent:
August 20, 2019
Assignee:
MACH 1, CORP.
Inventors:
Dra{hacek over (z)}en Bo{hacek over (s)}njak, Dylan J. Marcus
Abstract: An audio enhancement device including an audio-calculating module, a ratio-calculating module, a minimum-tracking module, a weighting-calculating module and a mixing module is provided. The audio-calculating module calculates a mid signal and a side signal according to a sum and a difference of an input first channel signal and an input second channel signal. The ratio-calculating module calculates a side-mid ratio of the side signal relative to the mid signal. The minimum-tracking module tracks a side-mid ratio minimum. The weighting-generating module determines a first and a second weighting values according to the side-mid ratio minimum. The mixing module weights the mid signal and the side signal based on the first and the second weighting values respectively and adjusts the input first channel signal and the input second channel signal accordingly to generate an enhanced first channel signal and an enhanced second channel signal.
Abstract: An ear shape analyzer includes: a sample ear analyzer configured to generate, for each of N sample ears, an ear shape data set that represents a difference between a point group representative of a three-dimensional shape of a reference ear and a point group representative of a three-dimensional shape of one of the N sample ears; an averaging calculator configured to generate averaged shape data by averaging N ear shape data sets generated by the sample ear analyzer; an ear shape identifier configured to identify an average ear shape of the N sample ears by translating coordinates of respective points of the point group representing the three-dimensional shape of the reference ear, by using the averaged shape data.
Abstract: Provided are methods and apparatus for enhancing a signal-to-noise ratio. In an example, provided is an apparatus configured to modify audio to better match the way the human brain processes audio by modifying the audio to a form which takes advantage of human echolocation capabilities. When humans listen to audio, they subconsciously listen for an echo and thus subconsciously focus on listening to, and for, meaningful information in audio. The focus causes humans to ignore noise in the audio, which results in enhancing a signal-to-noise ratio. In an example, the provided apparatus compensates for shortcomings of a device to which the apparatus is coupled by adjusting a respective amplitude of at least one constituent audio frequency of an output digital audio stream of the apparatus.
Abstract: The present technology relates to a signal processing device, a signal processing method, and a program, which are capable of reproducing an acoustic field more appropriately in accordance with content. A decoding unit decodes a multiplexed signal, and obtains a multichannel sound collection signal obtained by performing sound collection through a linear microphone array and spatial correction information for selecting a spatial correction scheme for correcting a spatial transfer characteristic. A spatial correction scheme selecting unit selects the spatial correction scheme on the basis of the spatial correction information, and a spatial transfer characteristic matrix generating unit outputs a spatial transfer characteristic matrix indicated by a selection result of the spatial correction scheme. A drive signal generating unit generates a speaker drive signal of a spatial frequency domain on the basis of the multichannel sound collection signal and the spatial transfer characteristic matrix.
Abstract: A propagation delay tune correction apparatus comprising a means for generating a frequency spectrum signal by performing short-term Fourier transform on an audio signal; a means for setting a propagation delay time for each of a plurality of predetermined frequency bands; a means for calculating a phase control amount for each of the plurality of predetermined frequency bands on a basis of the propagation delay time set for each of the plurality of predetermined frequency bands; a means for generating a phase control signal by smoothing the calculated phase control amount for each of the plurality of predetermined frequency bands; a means for controlling a phase of the frequency spectrum signal for each of the plurality of predetermined frequency bands on a basis of the generated phase control signal; and a means for generating an audio signal on which a propagation delay correction is performed by performing inverse short-term Fourier transform on the frequency spectrum signal of which the phase is controll
Abstract: An audio apparatus includes an analog module that receives a digital left (L) channel signal and a digital right (R) channel signal and outputs first and second analog L signals and first and second analog R signals, a first output port that includes first to fifth conductors, through which the first and second analog L signals, the first and second analog R signals, and a ground voltage are provided, and a second output port that includes sixth to ninth conductors, through which the first and second analog L signals and the first and second analog R signals are provided. When an audio jack including first to third terminals is inserted into the first output port, the first and second conductors are connected to the first terminal, the third and fourth conductors are connected to the second terminal, and the fifth conductor is connected to the third terminal.
Abstract: A stereo parameter conditioner performs a conditioning operation on a first value of a stereo parameter and a second value of the stereo parameter to generate a conditioned value of the stereo parameter. The first value is associated with a first frequency range, and the second value is associated with a second frequency range. The conditioned value is associated with a particular frequency range that is a subset of the first frequency range or a subset of the second frequency range.
Abstract: Example embodiments disclosed herein relates to upmixing of audio signals. A method of upmixing an audio signal is described. The method includes decomposing the audio signal into a diffuse signal and a direct signal, generating an audio bed at least in part based on the diffuse signal, the audio bed including a height channel, extracting an audio object from the direct signal, estimating metadata of the audio object, the metadata including height information of the audio object; and rendering the audio bed and the audio object as an upmixed audio signal, wherein the audio bed is rendered to a predefined position and the audio object is rendered according to the metadata. Corresponding system and computer program product are described as well.
Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which at least comprises one or more audio objects (106a). The encoder (108, 110) generates a bit stream (116) which comprises downmix signals (112) and side information which includes individual matrix elements (114) of a reconstruction matrix which enables reconstruction of the one or more audio objects (106a) in the decoder (120).
Type:
Grant
Filed:
June 21, 2018
Date of Patent:
July 9, 2019
Assignee:
Dolby International AB
Inventors:
Heiko Purnhagen, Lars Villemoes, Leif Jonas Samuelsson, Toni Hirvonen
Abstract: Disclosed is an audio signal processing device for performing binaural rendering on an input audio signal. The audio signal processing device includes a reception unit configured to receive the input audio signal, a binaural renderer configured to generate a 2-channel audio by performing binaural rendering on the input audio signal, and an output unit configured to output the 2-channel audio. The binaural renderer performs binaural rendering on the input audio signal based on a distance from a listener to a sound source corresponding to the input audio signal and a size of an object simulated by the sound source.
Abstract: The present invention relates to a neckband-type wireless sound transducer, and more particularly, to a neckband-type wireless sound transducer that prevents sound from being transferred to the surroundings of the wearer. The neckband-type wireless sound transducer according to the present invention, includes: a main body part configured to be seated on a human body; a main speaker part mounted on the main body part and configured to emit sound to the inside of the main body part; an auxiliary speaker part mounted on the main body part and configured to emit sound for leakage reduction to offset the sound emitted by the main speaker part and leaked to the outside of the main body part; and a controller configured to apply an electric signal for sound emission to the main speaker part and an electric signal for leakage reduction to the auxiliary speaker part.
Abstract: An audio navigation device comprising an input means for inputting two or more audio pieces into the navigation device; a spatialization means for allocating a position in the form of a unique spatial co-ordinate to each audio piece and arranging the audio pieces in a multi-dimensional arrangement; a generating means for generating a binaural audio output (3) for each audio piece, wherein the audio output (3) simulates sounds that would be made by one or more physical sources located at the given position of each audio piece; an output means for simultaneously outputting multiple audio pieces as binaural audio output (3) to a user (5); a navigation means (1) for enabling a user (5) to navigate around the audio output (3) in the multi-dimensional arrangement; a selection means (A) for allowing a user (5) to select a single audio output (3).
Type:
Grant
Filed:
April 21, 2016
Date of Patent:
June 25, 2019
Assignee:
III HOLDINGS 1, LLC
Inventors:
Mark Brian Sandler, Rebecca Lynne Stewart
Abstract: Provided are a method and device for improving sound quality of stereo sound and a terminal. The method includes: an original left channel signal and an original right channel signal are acquired; phases, frequency spectrums and amplitudes of the original left channel signal and original right channel signal are acquired; a left calibrated signal is acquired according to the phase, frequency spectrum and amplitude of the original left channel signal, and a right calibrated signal is acquired according to the phase, frequency spectrum and amplitude of the original right channel signal; the left calibrated signal and the original right channel signal are superposed to generate a final right channel output signal; the right calibrated signal and the original left channel signal are superposed to generate a final left channel output signal; the final right channel output signal and the final left channel output signal are combined to form a PCM signal.
Abstract: An enhanced Stereo Dimensional Array loudspeaker system 250 preferably including a mirror image pair of loudspeaker enclosures 280L, 280R configurable by a user or installer as a left-channel loudspeaker and a right channel loudspeaker each having a driver array aiming configuration with first and second angled baffle facets carrying main and effects drivers on separate facets and a Head Shadow filter signal processing system and method for driving the main and effects drivers to achieve a psycho-acoustically expanded image breadth by Head Shadow filter compensated inter-aural crosstalk cancellation.
Abstract: 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.
Abstract: A spatial audio processor for providing spatial parameters based on an acoustic input signal has a signal characteristics determiner and a controllable parameter estimator. The signal characteristics determiner is configured to determine a signal characteristic of the acoustic input signal. The controllable parameter estimator for calculating the spatial parameters for the acoustic input signal in accordance with a variable spatial parameter calculation rule is configured to modify the variable spatial parameter calculation rule in accordance with the determined signal characteristic.
Type:
Grant
Filed:
January 20, 2017
Date of Patent:
June 18, 2019
Assignee:
Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
Inventors:
Oliver Thiergart, Fabian Kuech, Richard Schultz-Amling, Markus Kallinger, Giovanni Del Galdo, Achim Kuntz, Dirk Mahne, Ville Pulkki, Mikko-Ville Laitinen
Abstract: A method of producing a spatialized stereo audio file from an original stereo audio file comprises creating a data base of impulse responses realized in at least one physical space divided into left, right, front, back, up and down sides relative to a sound acquisition position, with at least one pair of acquisition microphones placed at the sound acquisition position, with at least two pairs of source loudspeakers placed at sound source positions; the sound acquisition position is situated at the left-right median plane of the physical space, the sound source positions are distributed symmetrically by pairs relative to the sound acquisition position, the data base of impulse responses comprising at least one left/right impulse response pair, the left and right impulse responses being obtained by a deconvolution of the direct acquired signal from all the source loudspeakers distributed at the respective left and right side of the physical space.
Type:
Grant
Filed:
December 9, 2016
Date of Patent:
June 11, 2019
Assignee:
AXD Technologies, LLC
Inventors:
Franck Vincent Rosset, Jean-Luc Haurais
Abstract: 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.
Abstract: A transition between a stereophonic presentation and a monophonic presentation of a stereophonic input signal that includes a left channel signal and a right channel signal extracts content that is present at similar levels but not in-phase between the left and right channel signals to produce at least one of a left enhancement signal and a right enhancement signal. The left channel signal, the right channel signal, and only one of the left and right enhancement signals are combined to produce a monophonic signal for the monophonic presentation. Cross-fading between the left channel signal and the monophonic signal and between the right channel signal and the monophonic signal may be used to transition between the stereophonic presentation and the monophonic presentation. The stereophonic input signal may be up-mixed to produce enhancement signal. A similar transition between a multichannel presentation and a monophonic presentation of a multichannel signal is described.
Type:
Grant
Filed:
December 18, 2017
Date of Patent:
May 28, 2019
Assignee:
Apple Inc.
Inventors:
Sylvain J. Choisel, Afrooz Family, Brandon J. Rice
Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix.
Abstract: A method, apparatus and computer program product are provided to cause AN audio source to be modified in a manner consistent with the corresponding video images once the user and/or a display upon which the images are rendered has been tilted. In regards to a method, an initial virtual position of an audio source is determined. The method also includes determining a tilt angle that defines an angle that an apparatus embodying a display for rendering images has been tilted relative to a reference orientation of the apparatus with respect to a user of the apparatus. The method may also include modifying a virtual position of an audio source based upon the tilt angle and an initial virtual position of the audio source. The method may also include causing the audio source to be rendered in accordance with the virtual position as modified.
Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.
Abstract: An equalizing device has: a low-frequency zero-point circuit having a zero point in a low-frequency band of a before-equalization frequency characteristic of a communication medium; a high-frequency zero-point circuit having a zero point in a high-frequency band of the before-equalization frequency characteristic of the communication medium; and an intermediate-frequency zero-point circuit having a zero point in an intermediate-frequency band present between the low-frequency band and the high-frequency band, wherein an inclination of a waveform of the before-equalization frequency characteristic of the communication medium changes in the intermediate-frequency band; wherein the equalizing device equalizes the signal transmitted through the communication medium so as to restrain an amount of change in an inclination of a waveform of the after-equalization frequency characteristic.
Abstract: A method of estimating an individualized head-related transfer function (HRTF) and an individualized interaural time difference function (ITDF) of a particular person, comprising the steps of: a) obtaining a plurality of data sets (Li, Ri, Oi), each comprising a left and a right audio sample (Li, Ri) from a pair of in-ear microphones, and orientation information (Oi) from an orientation unit, measured in a test-arrangement whereby an acoustic test signal is rendered via a loudspeaker; b) storing the data sets in a memory; c) estimating the directions of the loudspeaker relative to the person based on the orientation data and the audio data; d) estimating the ITDF based on the data sets and on the estimated relative position/orientation; e) estimating the HRTF, based on the data sets and based on the estimated relative position/orientation.
Type:
Grant
Filed:
February 12, 2016
Date of Patent:
April 9, 2019
Assignee:
UNIVERSITEIT ANTWERPEN
Inventors:
Jonas Reijniers, Herbert Godelieve P Peremans, Bart Wilfried M Partoens
Abstract: An instant-on nothing-to-download “soundscaping” device that provides natural, atonal sounds such as the rolling surf of the ocean, running streams, gurgling brooks, rain, thunder, wind, crowd sounds, et al is provided.
Abstract: Methods and apparatus improve a user experience during telephone calls or other forms of communication in which a listener localizes electronically generated binaural sounds. The sound is convolved or processed to a location that is behind or near a source of the sound so that the listener perceives the location of the sound as originating from the source of the sound.
Abstract: A combo-plug detecting circuit for use in an audio CODEC is provided. The combo-plug detecting circuit is used to determine the contact configuration of the audio plug, to ensure that the audio plug belonging to the differential structure can be compatible with the audio CODEC.
Abstract: Certain aspects of the technology disclosed herein include generating a virtual sound field based on data from an ambisonic recording device. The ambisonic device records sound of a surrounding environment using at least four microphones having a tetrahedral orientation. An omnidirectional microphone having an audio-isolated portion can be used to isolate sound from a particular direction. Sound received from the plurality of microphones can be used to generate a virtual sound field. The virtual sound field include a dataset indicating a pressure signal and a plurality of velocity vectors. The ambisonic recording device can include a wide angle camera and generate wide angle video corresponding to the virtual sound field.
Type:
Grant
Filed:
July 13, 2017
Date of Patent:
February 12, 2019
Assignee:
ESSENTIAL PRODUCTS, INC.
Inventors:
Michael Kolb, Xinrui Jiang, Andrew E. Rubin, Matthew Hershenson, Xiaoyu Miao, Joseph Anthony Tate, Jason Sean Gagne-Keats, David John Evans, V, Rebecca Schultz Zavin
Abstract: Audio systems and methods are provided to reduce echo content in an audio signal. The systems and methods receive an audio signal and sound stage rendering parameter(s), and select a set of filter coefficients to filter the audio signal to provide an estimated echo signal. The estimated echo signal is subtracted from a microphone signal to provide an output signal with reduced echo content. The set of filter coefficients are selected based upon the sound stage rendering parameter(s) from among a plurality of stored sets of filter coefficients.
Type:
Grant
Filed:
August 3, 2017
Date of Patent:
February 5, 2019
Assignee:
BOSE CORPORATION
Inventors:
Cristian M. Hera, Vigneish Kathavarayan, Jeffery R. Vautin, Elie Bou Daher, Paraskevas Argyropoulos
Abstract: The disclosure relates to an audio signal processing apparatus for filtering a left channel input audio signal (L) and a right channel input audio signal (R), a left channel output audio signal (X1) and a right channel output audio signal (X2) to be transmitted over acoustic propagation paths to a listener, wherein transfer functions of the acoustic propagation paths are defined by an acoustic transfer function matrix. The audio signal processing apparatus comprises a decomposer, a first cross-talk reducer, a second cross-talk reducer, and a combiner. The first cross-talk reducer is configured to reduce a cross-talk within a first predetermined frequency band upon the basis of the acoustic transfer function matrix. The second cross-talk reducer is configured to reduce a cross-talk within a second predetermined frequency band upon the basis of the acoustic transfer function matrix.
Abstract: 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.
Abstract: A position locating method and apparatus are applied to the electronic information field, so that shifting an indication cursor in a wide range using a wireless indication device can be avoided, and a user operation can be simplified. The method comprises detecting a line-of-sight orientation of a user, and obtaining a line-of-sight orientation parameter of the user; detecting a line-of-sight source of the user, and obtaining a line-of-sight source position of the user; obtaining a position of a screen, and obtaining a screen position of a line of sight according to the line-of-sight orientation parameter, the position of the screen, and the line-of-sight source position; and receiving an adjustment signal, adjusting, according to the adjustment signal, the position displayed on the screen, and displaying an adjusted position on the screen.
Abstract: An information processing apparatus includes a holding unit configured to hold a plurality of head related transfer functions for outputting directional sound in a plurality of directions, a setting unit configured to set a direction in which a first head related transfer function and a second head related transfer function are switched, based on characteristics of the first head related transfer function and the second head related transfer function, and a switching unit configured to switch a head related transfer function used to output the directional sound between the first head related transfer function and the second head related transfer function in the set direction.
Abstract: Provided are an audio signal processing method and apparatus for adjusting a location of an audio object in correspondence to a location of a visual object. The audio signal processing apparatus includes a matching unit configured to select an audio object corresponding to a visual object extracted from a video signal among at least one audio object extracted from an audio signal, a location adjusting unit configured to adjust a location of a sound image of the audio signal based on a location of the selected audio object and a location of a visual object corresponding to the selected audio, and an output unit configured to output an audio signal whose the location of the sound image is adjusted.
Type:
Grant
Filed:
March 13, 2017
Date of Patent:
December 18, 2018
Assignee:
Gaudio Lab, Inc.
Inventors:
Hyunoh Oh, Jeonghun Seo, Taegyu Lee, Yonghyun Baek
Abstract: Provided herein are a headphone amplifier circuit for a headphone driver, an operation method thereof, and a universal serial bus (USB) interfaced headphone device using the same. The output stage of the headphone amplifier circuit is improved to have a differential output structure so as to effectively solve the crosstalk issue and increase the isolation between the left and the right channels such that the audio content presents a spatial perception and a distance perception more specifically.
Abstract: 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.
Type:
Grant
Filed:
August 24, 2017
Date of Patent:
December 4, 2018
Assignee:
SAMSUNG ELECTRONICS CO., LTD.
Inventors:
Sang-bae Chon, Sun-min Kim, Jae-ha Park, Sang-mo Son, Hyun Jo, Hyun-joo Chung
Abstract: A directivity control apparatus is provided which controls a directivity of a sound collected by a first sound collector including a plurality of microphones. The directivity control apparatus forms a directivity of the sound in a direction toward a position of a monitoring target in an image displayed on a display. Information on the position of the monitoring target in the image displayed on the display is obtained. The directivity of the sound is changed toward the position of the monitoring target in accordance with a movement of the monitoring target by referring to the obtained information on the position of the monitoring target.
Abstract: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating a panning function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating a mode matrix (?) from the loudspeaker positions, calculating a pseudo-inverse mode matrix (?+) and decoding the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the panning function (W) and the pseudo-inverse mode matrix (?+).
Abstract: A system for monitoring two or more persons in an area with an entry point having an access allowed indication or an access denied indication. a token device in communication with at least one wireless proximity detection device which determines a relative location of the token device to provide a detection data point for each of the wireless proximity detection devices and a set of detection data points for the group of detection data points; a system computing device which calculates the shared proximity of the token device according to the set of detection data points and determines that the token device contains at least two valid tickets or does not, if there are valid tickets and the shared proximity of the token device is within a predetermined area the access allowed indication will display to allow entry for a number of persons corresponding to the number of the tickets.
Type:
Grant
Filed:
June 26, 2015
Date of Patent:
November 13, 2018
Inventors:
Micah Bergdale, Nicholas Ihm, Gregory Valyer
Abstract: A system and method for joint acoustic echo control and adaptive array processing, comprising the decomposition of a captured sound field into N sub-sound fields, applying linear echo cancellation to each sub-sound field, selecting L sub-sound fields from the N sub-sound fields, performing L channel adaptive array processing utilizing the L selected sub-sound fields, and applying non-linear audio echo cancellation.
Abstract: An electronic device receives a first distance signal from a sensor embraced by a left receiver of a headset, and a second distance signal from a sensor embraced by a right receiver of the headset. The electronic device obtains a first distance between the left receiver and a left ear of the person according to the first distance signal, and obtains a second distance between the right receiver and a right ear of the person according to the second distance signal. The electronic device executes a specific operation according to the first distance and the second distance.
Abstract: A domain description is received, by a processor, the domain description identifying a domain associated with a sensor input. The domain description is formatted according to a hierarchical naming structure. A training data set is selected from a plurality of training data sets based upon the received domain description and sensor input. A combination of a subset of classifiers for classifying the sensor input is selected from a set of classifiers based upon the selected training data set.
Type:
Grant
Filed:
April 7, 2017
Date of Patent:
November 6, 2018
Assignee:
INTERNATIONAL BUSINESS MACHINES CORPORATION
Inventors:
Mandis S. Beigi, Seraphin B. Calo, Dinesh C. Verma, Shiqiang Wang, David A. Wood
Abstract: Disclosed herein are system, method, and tangible computer readable medium for creating a desired audio effect for a user. The method includes operations including: causing a plurality of speakers to play test signals, each test signal being specific to one of the speakers; receiving from a remote device recorded frequency responses of the speakers resulting from the playing of the test signals; creating one or more filters to match an audio profile selected by a user; applying the filters to the recorded frequency responses to obtain filtered transformations of the speakers; and transmitting the filtered transformations to the speakers; wherein the filtered transformations are applied at the speakers to thereby achieve the user audio profile.
Type:
Grant
Filed:
July 30, 2015
Date of Patent:
October 2, 2018
Assignee:
Roku, Inc.
Inventors:
Gregory Mack Garner, Patrick Alan Brouillette
Abstract: A binaural hearing system (“system”) preserves and/or enhances interaural level differences (“ILDs”) between first and second signals. The system includes audio detectors each associated with an ear of a user. The audio detectors detect an audio signal presented to the user and generate the first and second signals to represent the audio signal as detected at each ear. The system also includes sound processors associated with each ear that each receive the first and second signals from the audio detectors directly and/or by way of a communication link from the other sound processor. The sound processors each perform operations with respect to the first and second signals to preserve and/or enhance the ILDs between the signals. In so doing, the sound processors perform a contralateral gain synchronization operation to a first degree at the first sound processor and to a distinct second degree at the second sound processor.
Abstract: An audio decoder device for decoding a compressed input audio signal having at least one core decoder having one or more processors for generating a processor output signal based on a processor input signal, wherein a number of output channels of the processor output signal is higher than a number of input channels of the processor input signal, wherein each of the one or more processors has a decorrelator and a mixer, wherein a core decoder output signal having a plurality of channels has the processor output signal, and wherein the core decoder output signal is suitable for a reference loudspeaker setup; at least one format converter device configured to convert the core decoder output signal into an output audio signal, which is suitable for a target loudspeaker setup; and a control device configured to control at least one or more processors in such way that the decorrelator of the processor may be controlled independently from the mixer of the processor, wherein the control device is configured to contro
Type:
Grant
Filed:
January 22, 2016
Date of Patent:
September 25, 2018
Assignee:
Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
Inventors:
Christian Ertel, Johannes Hilpert, Andreas Hoelzer, Achim Kuntz, Jan Plogsties, Michael Kratschmer