Abstract: Processing sound in an enhanced reality environment can include generating, based on an image of a physical environment, an acoustic model of the physical environment. Audio signals captured by a microphone array, can capture a sound in the physical environment. Based on these audio signals, one or more measured acoustic parameters of the physical environment can be generated. A target audio signal can be processed using the model of the physical environment and the measured acoustic parameters, resulting in a plurality of output audio channels having a virtual sound source with a virtual location. The output audio channels can be used to drive a plurality of speakers. Other aspects are also described and claimed.

Abstract: Provided are a computer system for transmitting audio content to realize a user-customized being-there and a method thereof. The computer system may be configured to detect audio files that are generated for a plurality of objects at a venue, respectively, and metadata including spatial features that are set for the objects at the venue, respectively, and to transmit the audio files and the metadata for a user. An electronic device of the user may realize a being-there at the venue by rendering the audio files based on the spatial features in the metadata. That is, the user may feel a user-customized being-there as if the user directly listens to audio signals generated from corresponding objects at a venue in which the objects are provided.

Abstract: A system and method includes generating a virtual n-dimensional space that includes one or more positions of one or more source nodes and a position of a receiver node; executing a plurality of simulations including simulating acoustic signals emanating from the one or more source nodes within the virtual n-dimensional room; estimating a measure of the acoustic signals received at the receiver node; computing a plurality of acoustic signal data samples based on the estimation for each of the plurality of simulations; and creating a training data corpus for training an artificial neural network, the training data corpus including at least a sampling of the plurality of acoustic data samples, and the artificial neural network, once trained, is configured to generate an inference indicating a likely intended sound to a target receiver of a mixture of acoustic signals.

Abstract: Aspects of the disclosure provide methods and apparatuses for audio processing. In some examples, an apparatus for media processing includes processing circuitry. The processing circuitry receive audio inputs associated with a layered description for a space of interest in an audio scene. The space of interest includes a plurality of subspaces. The layered description includes a first layer and a second layer. The first layer has a common node with a first value that is a common attribute value of two or more subspaces in the plurality of subspaces. The second layer has individual nodes respectively associated with each of the plurality of subspaces. The processing circuitry determines the plurality of subspaces of the space of interest based on the layered description, and renders an audio output based on the audio inputs in response to a location of a subject of the audio scene being in the space of interest.

Abstract: A method by a computer system including generating audio files based on respective audio signals, the audio signals having been respectively generated from a plurality of objects at a venue, generating metadata including spatial features at the venue that are respectively set for the objects, and transmitting the audio files and the metadata for the objects to a first electronic device to cause the first electronic device to realize a being-there at the venue by rendering the audio files based on the spatial features in the metadata may be provided.

Abstract: Methods and systems are provided for integrating media cues into virtual reality scenes presented on a head mounted display (HMD) is disclosed. The method includes presenting a virtual reality scene on a display of an HMD. The method further includes receiving sensor data from one or more sensors in a real-world space in which the HMD is located. Then, identifying an object location of an object in the real-world space that produces a sound. The method includes generating a media cue in the virtual reality scene presented in the HMD. The media cue is presented in a virtual location that is correlated to the object location of the object in the real-world space.

Abstract: A microphone device comprising, for example, a mid microphone cartridge and two side microphone cartridges. The mid microphone cartridge may be, for example, a cardioid microphone cartridge, and the side microphone cartridges may each be, for example, a bidirectional microphone cartridge. Each of the mid microphone cartridge and of the two side microphone cartridges may be orthogonal to the other two microphone cartridges. The microphone device, which may be referred to herein as a mid dual-side microphone, may provide a combined pickup pattern, such as a beam and/or a toroid, that may be steerable.

Abstract: A sound signal processing method includes: obtaining a plurality of sound signals respectively collected by a plurality of microphones arranged in a space; adjusting respective levels of the plurality of sound signals in accordance with respective positions of the plurality of microphones; mixing the plurality of sound signals having the adjusted respective levels to thereby obtain a mixed signal; and generating a reflected sound by using the obtained mixed signal.

Abstract: In at least one embodiment, a system for synchronizing an audio stream is provided. The system includes a first loudspeaker and an audio controller. The first loudspeaker plays back a first audio output signal including first signature information. The audio controller provides a first audio input signal and superimpose the first signature information on the first audio input signal. The audio controller receives the first audio output signal including the first audio packets and the first signature information and to detect the first signature information. The audio controller determines a delay attributed to a transmission of the first audio input signal and the first audio output signal based on the first signature information and synchronizes the transmission of a second audio input signal from the audio controller to the first loudspeaker with the playback of another audio output signal from a second loudspeaker based at least on the delay.

Abstract: An audio signal processing method detects a position, in a sound space, of a sound source that generates an audio signal, the sound space being divisible into a plurality of areas, and generates an initial reflected sound control signal by convolving (i) an impulse response of an initial reflected sound linked to a first area of the sound space, among the plurality of areas of the sound space, corresponding to the detected position of the sound source to (ii) the audio signal of the sound source without convolving (iii) an impulse response of an initial reflected sound linked to any of the plurality of areas of the sound space other than the first area of the sound space corresponding to the detected position of the sound source to (ii) the audio signal of the sound source.

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: The description relates to representing acoustic characteristics of real or virtual scenes. One method includes generating directional impulse responses for a scene. The directional impulse responses can correspond to sound departing from multiple sound source locations and arriving at multiple listener locations in the scene. The method can include processing the directional impulse responses to obtain coherent sound signals and incoherent sound signals. The method can also include encoding first perceptual acoustic parameters from the coherent sound signals and second perceptual acoustic parameters from the incoherent sound signals, and outputting the encoded first perceptual acoustic parameters and the encoded second perceptual acoustic parameters.

Abstract: A method processes binaural sound to externally localize to a first user at a first location. This location is shared such that an electronic device processes the binaural sound to externally localize to a second user at a second location. The first and second locations occur at a same or similar location such that the first and second users hear the binaural sound as originating from the same or similar location.

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: 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: 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 method of reproducing an audio signal in a car cabin via a car audio system, the car audio system being configured to define at least one specific listening zone (LZ1, LZ2) of the audio signal for a specific listener position and a specific non-listening zone (NLZ1, NLZ2) of the audio signal for other listener positions within a car cabin, said car audio system comprising an audio signal source and a plurality of audio output devices associated with the audio signal source.

Abstract: Systems and methods for managing playback devices in accordance with embodiments of the invention are illustrated. One embodiment includes a method for modifying a system that includes several devices. The method includes steps for measuring a first signal pattern for wireless signals between the several devices, measuring a second signal pattern for the wireless signals after measuring the first signal pattern between the several devices, determining an updated state of the system based on a difference between the second signal pattern and the first signal pattern, and modifying state variables of one or more devices of the playback system based on the determined updated state.

Abstract: Provided is an information processing apparatus that includes an output processing unit, a recognized-position acquisition unit, and a comparing unit. The output processing unit generates a sound by using a head-related transfer function of a user measured in advance, the sound being to be output from a virtual sound source located at a first position in a space around the user. The recognized-position acquisition unit acquires information of a second position in the space, the second position being recognized as a position of the virtual sound source by the user who has listened to the sound. The comparing unit compares the first position and the second position with each other. The output processing unit controls the position of the virtual sound source of the sound output toward the user on the basis of a result of the comparison by the comparing unit.

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: 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: 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 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: 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: 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: 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.