Abstract: A device includes a memory configure to store instructions and one or more processors configured to execute the instructions to obtain spatial audio data at a first audio output device. The one or more processors are further configured to perform data exchange, between the first audio output device and a second audio output device, of exchange data based on the spatial audio data. The one or more processors are also configured to generate first monaural audio output at the first audio output device based on the spatial audio data.

Abstract: Audio signals from microphones of a mobile device are received. Each audio signal is generated by a respective microphone of the microphones. First microphones are selected from among the microphones to generate a front audio signal. Second microphones are selected from among the microphones to generate a back audio signal. A first audio signal portion, which is determined based at least in part on the back audio signal, is removed from the front audio signal to generate a modified front audio signal. A second audio signal portion is removed from the modified front audio signal to generate a left-front audio signal. A third audio signal portion is removed from the modified front audio signal to generate aright-front audio signal.

Abstract: The present disclosure relates to a system and a processing method in association with the system 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. The customized audio response characteristic(s) can be unique to the person.

Abstract: A method or apparatus switches between binaural sound and stereo sound in response to head movements of a listener. An electronic device provides the listener with binaural sound at a sound localization point (SLP) in a field-of-view of the listener. Binaural sound at the SLP switches to one of mono or stereo sound when head movements of the listener cause the SLP to move outside the field-of-view.

Abstract: One embodiment provides a method of automatic spatial calibration. The method comprises estimating one or more distances from one or more loudspeakers to a listening area based on a machine learning model and one or more propagation delays from the one or more loudspeakers to the listening area. The method further comprises estimating one or more incidence angles of the one or more loudspeakers relative to the listening area based on the one or more propagation delays. The method further comprises applying spatial perception correction to audio reproduced by the one or more loudspeakers based on the one or more distances and the one or more incidence angles. The spatial perception correction comprises delay and gain compensation that corrects misplacement of any of the one or more loudspeakers relative to the listening area.

Abstract: A machine learning model can determine output frequency response at different directions relative to a target audio output format, based on input including frequency response at directions relative to microphones of a capture device. A spatial filter determined based on the output frequency responses is applied to one or more of the microphone signals to map the spatial information from the microphone signals to the target audio output.

Abstract: A method improves performance of a computer that provides binaural sound to a listener. A memory stores coordinate locations that follow a path of how the head of the listener moves. This path is retrieved in anticipation of subsequent head movements of the listener to improve computer performance of executing binaural sound.

Abstract: Spatial audio is received from an audio server over a first communication link. The spatial audio is converted by a cloud spatial audio processing system into binaural audio. The binauralized audio is streamed from the cloud spatial audio processing system to a mobile station over a second communication link to cause the mobile station to play the binaural audio on the personal audio delivery device.

Abstract: 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 first input audio signal is received, the first input audio signal corresponding to a source location in a virtual environment presented to the user via the wearable head device. The first input audio signal is processed to generate a left output audio signal and a right output audio signal. The left output audio signal is presented to the left ear of the user via a left speaker associated with the wearable head device. The right output audio signal is presented to the right ear of the user via a right speaker associated with the wearable head device.

Abstract: An acoustic signal processing device calculates a signal waveform that a microphone receives when at least one of a sound source and the microphone is moving. The acoustic signal processing device includes a coefficient calculation unit configured to model a steering coefficient gk,m representing how much an amplitude of a sound source signal emitted at an mth discrete time, where m is an integer between 1 and M and M is a length of the sound source signal, is transferred to an amplitude of a signal that the microphone receives at a kth discrete time, where k is an integer between 1 and K and K is a length of a recording signal, using N-order Fourier series expansion where N is an integer of 1 or more, and a recording signal calculation unit configured to calculate the signal waveform that the microphone receives using the modeled steering coefficient gk,m.

Abstract: An information processing device (100) according to the present disclosure includes: an acquisition unit (141) configured to acquire a first image including a content image of an ear of a user; and a calculation unit (142) configured to calculate, based on the first image acquired by the acquisition unit (141), a head-related transfer function corresponding to the user by using a learned model having learned to output a head-related transfer function corresponding to an ear when an image including a content image of the ear is input.

Abstract: A system for rotating sound or selective listening to sound provides for the ability of the apparent direction of sound sources in a listening environment to remain in consistent orientations in space despite rotations of the microphones used to capture the sound and despite rotations of the head of the listener, even when wearing headphones. Modules are provided in the system to distinguish the sound sources and their apparent directions, as well as to optionally rotate the sound sources in response to detected rotations of the listener's head and/or detected rotations of the microphones.

Abstract: An audio communication system for communication between vehicle occupants in a vehicle, including an image capturing device configured to monitor a first vehicle occupant, a processor configured to receive an image of the first vehicle occupant from the image capturing device and determine whether the first vehicle occupant is attracting attention from a second vehicle occupant, a first microphone associated to the first vehicle occupant configured to receive an audio input from the first vehicle occupant in response to the determination of the first vehicle occupant attracting the second vehicle occupant's attention, and a first speaker associated to the second vehicle occupant configured to activate an audio augmentation of the received audio input in the first speaker and output the augmented audio input.

Abstract: When three-dimensional audio is produced by using headphones, particular HRTF-filters are used to modify sound for the left and right channels of the headphone. As the morphology of every ear is different, it is beneficial to have HRTF-filters particularly designed for the user of headphones. Such filters may be produced by deriving ear geometry from a plurality of images taken with an ordinary camera, detecting necessary features from images and fitting said features to a model that has been produced from accurately scanned ears comprising representative values for different sizes and shapes. Taken images are sent to a server (52) that performs the necessary computations and submits the data further or produces the requested filter.

Abstract: Channels of audio data in a spatial audio object are associated with any one or more of a direction and a location of one or more recorded sounds, which channels are to be reproduced as spatial sound. A visualized spatial sound object represents a snapshot/thumbnail of the spatial sound. To preview the spatial sound (by experiencing its snapshot or thumbnail), a user manipulates the orientation of the visualized spatial sound object, and a weighted downmix of the channels is rendered for output as a spatial preview sound, e.g., a single output audio signal is provided to a spatial audio renderer; one or more of the channels that are oriented toward the user are emphasized in the preview sound, more than channels that are oriented away from the user. Other aspects are also described and claimed.

Abstract: Acoustic waveguides can be used to improve audio performance of playback devices, such as a soundbar. Such a playback device can include an elongated body defining an outer perimeter with a forward surface, an upper surface, and a rounded edge between the forward surface and the upper surface. An up-firing transducer is configured to direct sound along an axis that has a vertical oblique angle with respect to a forward axis. A waveguide in fluid communication with the up-firing transducer includes a sidewall extending circumferentially around the transducer, the sidewall having a first end adjacent the up-firing transducer and a second end adjacent the outer perimeter, such that an opening defined by the sidewall has a larger area at the second end than at the first end. A rear portion of the sidewall is more steeply angled with respect to the axis than a forward portion of the sidewall.

Abstract: An apparatus including circuitry configured for: defining at least one parameter field associated with an input multi-channel audio signals, the at least one parameter field configured to describe at least one characteristic of the multi-channel audio signals; determining at least one spatial audio parameter associated with the multi-channel audio signals; and controlling a rendering of the multi-channel audio signals by processing the input multichannel audio signals using at least the at least one characteristic of the multi-channel audio signals and the at least one spatial audio parameter.

Abstract: This application discloses a device to project sound to a specific person while keeping the sound away from others nearby. The device has various parametric speakers arranged in a pattern to project sounds in a narrow field while preventing dispersion of sounds to nearby areas. The device can be connected to other electronic equipment by wires or wirelessly, and can be removed and used in different settings. This device has particular application in delivering sounds to infants and toddlers to sooth them while the same sounds are not heard by adults nearby.

Abstract: A device includes one or more processors configured to execute instructions to obtain, at a first audio output device, first spatial audio data and a first reference time associated, and to cause the first reference time and data representing at least a portion of the first spatial audio data to be transmitted from the first audio output device. The instructions further cause the one or more processors to receive, at the first audio output device from a second audio output device, second spatial audio data and a second reference time. The instructions further cause the one or more processors to, based on the first reference time and the second reference time, time-align the first spatial audio data and the second spatial audio data to generate combined audio data representing a three-dimensional (3D) sound field and to generate audio output based on the combined audio data.

Abstract: There is provided a method 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.

Abstract: A system that performs sound source localization (SSL) using acoustic wave decomposition (AWD) or an approximation. When a device detects a wakeword represented in audio data, the device performs SSL processing in order to determine a position of the user relative to the device (e.g., estimate angle of the user). The device calculates noise statistics based on first audio data representing the wakeword and second audio data preceding the wakeword. Thus, upon detecting the wakeword, the device calculates the noise statistics and a signal quality metric corresponding to the wakeword. In addition, the device uses Multi-Channel Linear Prediction Coding (MCLPC) coefficients to average out the room impulse response. Using the noise statistics, the MCLPC coefficients, and the audio data, the device performs AWD processing to decompose the sound field to disjoint acoustic plane waves, enabling the device to identify the most likely direction for the line-of-sight component of speech.

Abstract: The description relates to rendering directional sound. One implementation includes receiving directional impulse responses corresponding to a scene. The directional impulse responses can correspond to multiple sound source locations and a listener location in the scene. The implementation can also include encoding the directional impulse responses to obtain encoded departure direction parameters for individual sound source locations. The implementation can also include outputting the encoded departure direction parameters, the encoded departure direction parameters providing sound departure directions from the individual sound source locations for rendering of sound.

Abstract: The present disclosure describes a method and a system of panning and vector reproduction of 4 audio channels, which is based on what in this document is defined as a vector panning, which allows audio panning that is not limited to placing sounds on a simple horizontal line, said vector panning, creates an audio image within a panoramic field delimited by X and Y axes. Said audio image is structured on 8 stereo panning lines which in turn allow us to place sounds in at least 25 panning points. The present disclosure comprises a vector audio reproduction equipment, which must have certain specific characteristics for the proper functioning of the disclosure, such as using only mono audio reproduction systems, same which must emit a full frequency range each, among other characters Energetic, which can be positioned in a configuration of specific angles and distances in relation to the listener in 3 different embodiments.

Abstract: Systems and methods for spatially emulating a sound source. An apparatus includes a microphone array including microphones; and a sound profiler communicatively connected to the microphone array, the sound profiler including a processing circuitry and a memory which contains instructions that, when executed by the processing circuitry, configure the apparatus to: generate synthesized audio based on sound beam metadata, a sound profile, and target listener location data, wherein the sound beam metadata includes timed sound beams defining a directional dependence of a spatial sound wave, wherein the sound profile includes timed sound coefficients determined based on audio signals captured in a space wherein the target listener location data includes a position and an orientation, wherein the synthesized audio emulates sound that would be heard by a listener at the position and orientation of the target listener location data; and providing the synthesized audio for projection.

Abstract: Systems and methods are described for extended reality environment interaction. An extended reality environment including a first object is generated for display, and the first object is identified in a field of view based on received input from one or more sensors. An eyelid motion is detected based on the received input, and in response to detecting the eyelid motion, the first object is regenerated for display with a modified level of detail.

Abstract: Disclosed is an audio signal processing apparatus for rendering an input audio signal. The audio signal processing apparatus may include a processor configured to obtain an input audio signal including an ambisonics signal and a non-diegetic channel difference signal, render the ambisonics signal to generate a first output audio signal, mix the first output audio signal and the non-diegetic channel difference signal to generate a second output audio signal, and output the second output audio signal.

Abstract: A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

Abstract: In one embodiment, a method for determining a dynamic head-related transfer function for a subject includes receiving audio recordings of a sound captured by audio sensors. The sound is emitted by an in-ear speaker worn by the subject. Additionally, a reference signal captured by a microphone coupled to the in-ear speaker and one or more images captured by image sensors are received. The one or more images depict a body pose of the subject while the sound is emitted by the in-ear speaker and may be used to generate a pose representation of the body pose of the subject. A head-related transfer function for each audio sensor is determined based on the pose representation, the audio recordings of the sound, and the reference signal. The dynamic head-related transfer function is determined based on the head-related transfer function for each audio sensor.

Abstract: This application discloses a sound reproduction method performed at a computing device. The method includes: detecting, by the computing device, a sound triggering event that corresponds to a first virtual object, the sound triggering event carrying sound source feature information used for matching a sound source; determining, by the computing device according to the sound source feature information, a sound source position at which the sound source is located, and obtaining a first transmission distance between the sound source position and a first position at which the first virtual object is located; determining, by the terminal according to the first transmission distance, a target sound of the sound source at the first position; and generating, by the terminal, the target sound at the first position in the virtual scene. This application resolves a technical problem that accuracy of sound reproduction is relatively low in a sound reproduction method.

Abstract: Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.

Abstract: A voice-activated electronic device including a first portion with a first internal surface having a first attachment structure, and a second portion with a second internal surface having a second attachment structure. The first and second internal surfaces have compatible shapes that permit the first and second portions to be moved from a separated position to a joined position, where when the first portion and the second portion are in the joined position the first and second internal surfaces form a nested arrangement. The first and second attachment structures form a secure but separable connection to one another when the first and second portions are in the joined position. The first and second portions are configured to be joined securely and separated through manual human manipulation of one or both of the first portion and the second portion to move the first and second portions between separated and joined positions.

Abstract: An acoustic device includes binaural speakers, a storing portion for storing a plurality of head-related transfer functions, and a signal processing portion for processing a sound signal with a head-related transfer function. Two or more function candidates are selected from the plurality of head-related transfer functions. Prescribed test sounds are emitted from the speaker. A function candidate to apply to the user is selected based on a discrepancy between a user-perceived location direction and the sound generation location direction for each function candidate.

Abstract: A method of reproducing a multi-channel audio signal including an elevation sound signal in a horizontal layout environment is provided, thereby obtaining a rendering parameter according to a rendering type and configuring a down-mix matrix, and thus effective rendering performance may be obtained with respect to an audio signal that is not suitable for applying virtual rendering. A method of rendering an audio signal includes receiving a multi-channel signal includes a plurality of input channels to be converted into a plurality of output channels; determining a rendering type for elevation rendering based on a parameter determined from a characteristic of the multi-channel signal; and rendering at least one height input channel according to the determined rendering type, wherein the parameter is included in a bitstream of the multi-channel signal.

Abstract: A system for generating individualized HRTFs that are customized to a user of a headset. The system includes a server and an audio system. The server determines the individualized HRTFs based in part on acoustic features data (e.g., image data, anthropometric features, etc.) of the user and a template HRTF. The server provides the individualized HRTFs to the audio system. The audio system presents spatialized audio content to the user using the individualized HRTFs.

Abstract: Systems and methods for rendering spatial audio in accordance with embodiments of the invention are illustrated. One embodiment includes a spatial audio system, including a primary network connected speaker, including a plurality of sets of drivers, where each set of drivers is oriented in a different direction, a processor system, memory containing an audio player application, wherein the audio player application configures the processor system to obtain an audio source stream from an audio source via the network interface, spatially encode the audio source, decode the spatially encoded audio source to obtain driver inputs for the individual drivers in the plurality of sets of drivers, where the driver inputs cause the drivers to generate directional audio.

Abstract: Disclosed are an audio encoding method, to which BRIR/RIR parameterization is applied, and a method and device for reproducing audio by using parameterized BRIR/RIR information.

Abstract: 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 first input audio signal is received, the first input audio signal corresponding to a source location in a virtual environment presented to the user via the wearable head device. The first input audio signal is processed to generate a left output audio signal and a right output audio signal. The left output audio signal is presented to the left ear of the user via a left speaker associated with the wearable head device. The right output audio signal is presented to the right ear of the user via a right speaker associated with the wearable head device.

Abstract: This disclosure relates to techniques for generating physically accurate auralization of sound propagation in complex environments, while accounting for important wave effects, such as sound absorption, sound scattering, and airborne sound insulation between rooms. According to some embodiments, techniques may be utilized to determine more accurate, e.g., “acoustically-effective” room volumes that account for open windows, open doors, acoustic dead space, and the like. According to other embodiments disclosed herein, techniques may be utilized to perform optimized hybrid acoustical ray tracing, including grouping coherent rays by processing core.

Abstract: An audio adjusting method and an audio adjusting device are provided. The audio adjusting method includes the steps of: receiving a plurality of directional audio signals; calculating a displacement amount through a positioning module; calculating a plurality of gain values corresponding to the displacement amount through a processing module; adjusting volume levels of the plurality of directional audio signals according to the plurality of gain values; converting the plurality of directional audio signals into a two- or multi-channel audio signal according to the quantity of two or more channels of a head-mounted playback device; and playing the two-channel or multi-channel audio signal through the head-mounted playback device.

Abstract: An apparatus for spatial audio signal processing, the apparatus including at least one processor configured to: receive captured audio content, the captured audio content captured within a capture environment with defined geometry; determine audio directions and distances associated with the captured audio content; determine a listening space area geometry; map the audio directions and distances associated with the captured audio content relative to the defined geometry to the listening space area geometry; and synthesize a volumetric audio based on the mapped audio directions and distances to generate an audio content experience within the listening space area geometry.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: The invention relates to a method of determining latency information in a wireless audio system, the wireless audio system comprising at least one wireless receiver, a wireless transmitter and a sound recorder, the method comprising the following steps: transmitting at least one test signal from the wireless transmitter to the wireless receiver, the test signal defining a time reference; playing the test signal at the wireless receiver; acquiring an audio record by the sound recorder while the test signal is played by the wireless receiver; determining, by the sound recorder, a latency information related to the wireless receiver at least based on the time reference and based on the audio record, and the wireless transmitter and the sound recorder being synchronized; transmitting the determined latency information to the wireless transmitter.

Abstract: A method improves performance of a computer that provides binaural sound to a listener. A memory stores coordinate locations that follow a path of how the head of the listener moves. This path is retrieved in anticipation of subsequent head movements of the listener to improve computer performance of executing binaural sound.

Abstract: An in-ear device includes a housing shaped to hold the in-ear device in an ear of a user, and an audio package, disposed in the housing, to emit augmented sound. A first set of one or more microphones is positioned to receive external sound, and a controller is coupled to the audio package and the first set of one or more microphones. The controller includes a low-latency audio processing path, digital control parameters, and logic that when executed by the controller causes the in-ear device to perform operations. The operations may include receiving the external sound with the first set of one or more microphones to generate a low-latency sound signal; augmenting the low-latency sound signal by passing the low-latency sound signal through the low-latency audio processing path to produce an augmented sound signal; and outputting, with the audio package, the augmented sound based on the augmented sound signal.

Abstract: A system and method for performing perceptually-transparent RTF estimation for a two-channel system with low estimation errors. The system includes two loudspeakers for the left and right channels positioned in a room and a single microphone, positioned in the vicinity of a listener in the room. The system performs spectral splitting using a filter bank and a complementary filter bank, to obtain separation in frequency between the two channels. Calibration signals being uncorrelated across the two channels are, generated using the original audio signals and the constructed filters. Then, the calibration signals are played back by the loudspeakers, instead of playback of the original audio signals. At the end, the system records the playback of the calibration signals by the microphone and simultaneously estimates the two-channel RTF using the recorded played back calibration signals.

Abstract: The present disclosure relates to a system and a processing method in association with the system 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. The customized audio response characteristic(s) can be unique to the person.

Abstract: A device to perform audio signal equalization includes one or more processors configured to receive impulse response data corresponding to multiple audio channels. Each audio channel is associated with a corresponding microphone of multiple microphones of an audio device and indicative of sound propagation from one or more speakers of the audio device to the corresponding microphone. The one or more processors are configured to generate equalization filter data that is based on the impulse response data and that is indicative of multiple equalization filters. Each of the equalization filters is associated with a corresponding audio channel of the multiple audio channels. The one or more processors are also configured to process the equalization filter data to determine a playback equalization filter to be applied to an audio playback signal prior to playout at the one or more speakers.

Abstract: The current invention related to methods and systems for panning audio objects on multichannel loudspeaker setups. The invention relates to a method of processing an audio object along an axis, said audio object comprising an audio object abscissa and an audio object spread, for spatialized restitution thereof over a plurality of sound transducers, N in number, aligned along said axis; each of said sound transducers comprising a transducer abscissa; N being at least equal to two; said method comprising a plurality of steps.

Abstract: In general, techniques are described by which to render different portions of audio data using different renderers. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store audio renderers. The processor(s) may obtain a first audio renderer of the plurality of audio renderers, and apply the first audio renderer with respect to a first portion of the audio data to obtain one or more first speaker feeds. The processor(s) may next obtain a second audio renderer of the plurality of audio renderers, and apply the second audio renderer with respect to a second portion of the audio data to obtain one or more second speaker feeds. The processor(s) may output, to one or more speakers, the one or more first speaker feeds and the one or more second speaker feeds.

Abstract: A positioning system has an initiator device configured for emitting a high-speed wireless signal, at least one reference device configured for receiving the high-speed wireless signal and emitting a low-speed wireless signal after receiving the high-speed wireless signal, at least one target device each having one or more components for receiving the low-speed wireless signals, and at least one engine configured for determining the position of each of the at-least-one target device by calculating the distance between the target device and each of the at-least-one reference device based on at least the times-of-arrival of the low-speed wireless signals, each time-of-arrival being the time that the corresponding low-speed wireless signal being received by the target device, and determining the position of the target device based on the calculated distances.