Abstract: Techniques are described for handling duress input. For example, techniques are described for handling duress input provided to a mobile application that controls a monitoring system located at a fixed property.

Abstract: A wearable electronic device (WED) includes a display that displays a virtual image on a physical object located in an area around a user and speakers that play sound as binaural sound that originates from the virtual image. One or more processors scan the area around the user to locate the physical object, divide the area around the user into different zones, assign room impulse responses (RIRs) to the zones, and process the sound with the RIRs.

Abstract: A speaker device includes left and right speakers and a correction part. The correction part corrects frequency characteristics of a stereo audio signal supplied to the left and right speakers so that transfer characteristics of the left and right speakers become predetermined transfer characteristics.

Abstract: In a system that includes one or more audio output components, positioning information associated with positioning of a user within a three-dimensional space around the user maybe determined. The one or more audio output components may be controlled based on the determined positioning information, with the controlling including providing a three-dimensional audio environment according to the user. The controlling may include configuring least one audio output related parameter or function for optimizing, based on the determined positioning information, directionality of at least one of the one or more audio output components. The determined positioning information may be generated or adjusted based on sensory data relating to the user.

Abstract: An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes a signal analyzer for estimating a signal characteristic of the first channel signal or the second channel signal or both signals or a signal derived from the first channel signal or the second channel signal; a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a weighter for weighting a smoothed or non-smoothed cross-correlation spectrum to obtain a weighted cross correlation spectrum using a first weighting procedure or using a second weighting procedure depending on a signal characteristic estimated by the signal analyzer, wherein the first weighting procedure is different from the second weighting procedure; and a processor for processing the weighted cross-correlation spectrum to obtain the inter-channel time difference.

Abstract: An apparatus including circuitry configured to: receive a spatial audio signal, the spatial audio signal including at least one audio signal and spatial metadata associated with the at least one audio signal; obtain a room effect control indication; and determine, based on the room effect control indication, whether a room effect is to be applied to the at least one audio signal, wherein the circuitry is configured, when the room effect is to be applied to the spatial audio signal, to: generate a first part binaural audio signal based on the at least one audio signal and spatial metadata; generate a second part binaural audio signal based on the at least one audio signal, at least the second part binaural audio signal is generated with at least in part the room effect so as to have a different response than a response of the first part binaural audio signal; and combine the first part binaural audio signal and the second part binaural audio signal to generate a combined binaural audio signal.

Abstract: A system, method, and wireless earpieces for implementing a virtual assistant in response to user preferences. User preferences associated with a user of the wireless earpieces are received. Data and information about the user and an environment of the user are automatically captured by the wireless earpieces based on the user preferences. A determination is made whether to provide automatic assistance to the user based on the user preferences utilizing the virtual assistant of the wireless earpieces. The automatic assistance is generated through the virtual assistant of the wireless earpieces utilizing the data and the information. The automatic assistance is communicated to the user through the virtual assistant of the wireless earpieces.

Abstract: An apparatus for synthesizing a spatially extended sound source includes: a spatial information interface for receiving a spatial range indication indicating a limited spatial range for the spatially extended sound source within a maximum spatial range; a cue information provider for providing one or more cue information items in response to the limited spatial range; and an audio processor for processing an audio signal representing the spatially extended sound source using the one or more cue information items.

Abstract: [Problem] To generate acoustic profile information using a simple operation and without increasing the cost of a multichannel audio device. [Solution] A wireless recording playback terminal 5 is positioned at a listening point of a multichannel audio device 2 as a measuring microphone. The wireless recording playback terminal 5 records, as channel playback data in accordance with a test signal playback command received from the multichannel audio device 2, a test signal outputted from a speaker 3 that corresponds to the channel designated by the command. The wireless recording playback terminal 5 also plays back the sound source of the test signal and records the played-back sound source as sound source playback data without being outputted to the outside from the speaker.

Abstract: A method of controlling radio transmission in a wireless communication network is described, the method comprising a radio device (10; 400; 700) receiving an indication of a set of first signals (31, 32, 33, 34, 35, 36) for associating the set of first signals (31, 32, 33, 34, 35, 36) with a second signal. The method comprises the radio device (10; 400; 700) receiving at least one of the first signals (31; 32, 33, 34, 35, 36); and in response to the indication, the radio device (10; 400; 700) estimating at least one parameter of the second signal based on the at least one received first signal. The method further comprises, based on the at least one estimated parameter, the radio device (10; 400; 700) processing at least one radio transmission (21) from the wireless communication network.

Abstract: An apparatus (303) comprising means configured to: obtain at least one audio signal (112); determine at least one direction parameter (506) of at least one propagating sound associated with the at least one audio signal (112); determine at least one direction (302) of at least one sound source associated with the at least one audio signal (112); modify the at least one direction parameter (506) in accordance with the determined at least one direction (302); and output the modified at least one direction parameter (508).

Abstract: A device for managing sound playback includes one or more processors configured to receive an indication of a user-device interaction between a user and an audio interface device during a sound playback operation of a multi-speaker audio playback system. The one or more processors are also configured to, based on receiving the indication of the user-device interaction, initiate a selective adjustment of the sound playback operation to reduce a playback sound of the multi-speaker audio playback system based on a position of the user.

Abstract: A sound signal downmix method includes an inter-channel relationship information obtaining step of obtaining an inter-channel correlation value and an inter-channel time difference in an approximate manner, and a downmix step of obtaining a downmix signal based on the obtained information. In the inter-channel relationship information obtaining step, multiple channel signals are sorted such that signals of adjacent channels are similar to each other, the inter-channel correlation value and the inter-channel time difference are determined only between adjacent channels after the sorting, the inter-channel correlation value between non-adjacent channels is obtained by a product or a geometric mean of the inter-channel correlations between the adjacent channels, and the inter-channel time difference between non-adjacent channels is obtained by adding up the inter-channel time differences of adjacent channels.

Abstract: Arrangement for distributing head related transfer function filters. In the arrangement a user device sends a request for a head related transfer function filter to the service being used. The service verifies if the user of the device has a subscription for a head related transfer function filters in the service being used and retrieves a filter as a response to a positive verification result. The service may filter audio channels and transmit filtered audio further. In an alternative embodiment the service transmits the filter to the user device for filtering the audio.

Abstract: A signal processing method includes: receiving, by a first reception unit, an acoustic signal transmitted via a first transmission path, the acoustic signal received by the first reception unit being a first acoustic signal; receiving, by a second reception unit, the acoustic signal transmitted via a second transmission path different from the first transmission path, the second transmission path having a longer delay time for transmission than the first transmission path, and the acoustic signal received by the second reception unit being a second acoustic signal; calculating, by a delay amount calculation unit, a transmission delay amount that is a relative delay amount between the first acoustic signal and the second acoustic signal; and delaying, by a delay amount addition unit, the first acoustic signal based on the calculated transmission delay amount and outputting the delayed first acoustic signal.

Abstract: An augmented reality headset is provided that includes a first microphone mounted on a headset and configured to receive a first audio signal from a speaker, and a second microphone remotely located from the headset and configured to receive a second audio signal from the speaker. The augmented reality headset also includes a memory storing multiple instructions, and one or more processors configured to execute the instructions. When the one or more processors execute the instructions, they cause the augmented reality headset to: synchronize the first audio signal and the second audio signal, form an enhanced audio signal with the first audio signal and the second audio signal, and provide the enhanced audio signal to a user of the augmented reality headset.

Abstract: An apparatus includes one or more processors configured to receive orientation data and to select, based on the orientation data, a particular filter from among multiple filters. The one or more processors are configured to perform signal processing operations associated with three-dimensional (3D) sound data based on the particular filter.

Abstract: A method of audio processing includes generating harmonics in a hybrid complex quadrature mirror filter domain. Generating the harmonics may include multiplication, using a feedback delay loop, and dynamic compression. The harmonics may be generated based on one or more hybrid sub-bands of the complex transform domain signal.

Abstract: There is provided a sound tracing method. The method includes: a setup processing step of setting ray information; a ray generation step of generating a sound ray; a traversal/intersection test step of generating hit triangle information; a propagation path validation (PPV) step of searching a sound path; a guide plane sort step of generating and sorting a guide plane; a Reverb Geometry Collect/Reverb Plane Sort (RGC/RPS) step of generating and sorting a reverb plane; and an impulse response calculation step of calculating an impulse response (IR) and storing the calculated IR in a valid path buffer.

Abstract: A sound source separation program causes a computer to acquire an acoustic signal, convert the acquired acoustic signal from a time region to a frequency region, and perform sound source separation on the acoustic signal converted to the frequency region by performing updating based on elementary row operation on a demixing matrix to iteratively minimize an objective function including a quadratic form of a separation vector and a determinant of the demixing matrix.

Abstract: A sound reproduction method includes: obtaining a first audio signal indicating a first sound which arrives at a listener from a first range and a second audio signal indicating a second sound which arrives at the listener from a predetermined direction; when the first range and the predetermined direction are determined to be included in a second range which is a back range relative to a front range in the direction that the head part of the listener faces, performing a correction process on at least one of the first audio signal or the second audio signal so that intensity of the second audio signal becomes higher than intensity of the first audio signal; and performing mixing of the at least one of the first audio signal or the second audio signal, and outputting, to an output channel, the first and second audio signals.

Abstract: A method including: obtaining at least two audio signals for reproduction, each of the at least two audio signals associated with a respective one of at least two reproduction locations within an audio reproduction space; obtaining within the audio reproduction space at least two zones; obtaining at least one location for a user's position within the audio reproduction space, the at least one location being relative to at least one of the at least two zones and the at least two reproduction locations; and processing the at least two audio signals based on the obtained at least one location for the user's position within the audio reproduction space to generate at least one output audio signal, the at least one output audio signal is reproduced from at least one of the at least two reproduction locations.

Abstract: A method (1900) for audio signal filtering. The method includes generating (s1902) a pair of filters for a certain location specified by an elevation angle ? and an azimuth angle ?, the pair of filters consisting of a right filter (hr(?,?)) and a left filter (hl(?, ?)); filtering (s1904) an audio signal using the right filter; and filtering (s1906) the audio signal using the left filter. Generating the pair of filters comprises: i) obtaining at least a first set of elevation basis function values at the elevation angle: ii) obtaining at least a first set of azimuth basis function values at the azimuth angle; iii) generating the right filter using: a) at least the first set of elevation basis function values, b) at least the first set of azimuth basis function values, and c) right filter model parameters; and iv) generating the left filter using: a) at least the first set of elevation basis function values, b) at least the first set of azimuth basis function values, and c) left filter model parameters.

Abstract: Systems and methods for audio in accordance with embodiments of the invention are illustrated. One embodiment includes a method for upmixing audio, including receiving an audio track which includes an input plurality of channels, each channel having an encoded audio signal, decoding the audio signal, calculating a first frequency spectrum for a low frequency component of the signal using a first window, calculating a second frequency spectrum for a high frequency component of the signal using a second window, determining at least one direct signal by estimating panning coefficients, estimating at least one ambient signal based on the at least one direct signal, and generating an output plurality of channels based on the at least one direct signal and the at least one ambient signal.

Abstract: A hearing system comprises a) a multitude of M of microphones providing M corresponding electric input signals xm(n), m=1, . . . , M, and n representing time, b) a processor connected to said multitude of microphones and providing a processed signal in dependence thereof, c) an output unit for providing an output signal in dependence of said processed signal, and d) a database (?) comprising a dictionary (?pd) of previously determined acoustic transfer function vectors (ATFpd).

Abstract: An apparatus for generating a description of a combined audio scene, includes: an input interface for receiving a first description of a first scene in a first format and a second description of a second scene in a second format, wherein the second format is different from the first format; a format converter for converting the first description into a common format and for converting the second description into the common format, when the second format is different from the common format; and a format combiner for combining the first description in the common format and the second description in the common format to obtain the combined audio scene.

Abstract: An artificial reality headset enhances audio signals from a target sound source using information from other devices in the local area. A primary headset broadcasts a location of a target sound source to secondary headsets in a local area. The secondary headsets transmit audio signals to the primary headset to enhance the audio content presented by the primary headset to a user. The secondary headset may select an array transfer function for the location of the target sound source. The secondary headsets correlate known transfer functions in the target direction with estimated transfer functions. The secondary headset may perform beamforming on the target sound source and transmit the output audio signal to the primary headset. In some embodiments, the secondary headset may transmit the array transfer function and a raw audio signal to the primary headset. The primary headset generates audio content based on the received audio signal.

Abstract: This disclosure describes a system and method for Head-Related Transfer Function (HRTF) interpolation when an HRTF dataset does not contain a particular direction associated with a desired source. The disclosed HRTF interpolation uses a finite set of HRTFs from a dataset to obtain the HRTF of any possible direction and distance, even if the direction/distance doesn't exist on the current dataset.

Abstract: Provided is a signal processing device including: a reverberation sound signal generation unit that generates a reverberation sound signal according to a sound source position of a virtual sound source and a distance to a reference point; and a drive signal generation unit that generates a drive signal for a speaker array by a wavefront synthesis filter, in which the drive signal generation unit generates the drive signal on the basis of a signal obtained by performing wavefront synthesis filtering processing on a signal obtained by convolving the reverberation sound signal with a signal of the virtual sound source and/or a signal obtained by performing wavefront synthesis filtering processing on the reverberation sound signal to make the reverberation sound signal into a virtual sound source.

Abstract: A data processing device includes: a digital signal processor; at least one processor; and at least one memory device configured to store a plurality of instructions, which when executed by the at least one processor, cause the at least one processor to operate to: output a first determination result relating to a scene of content through use of sound data; select processing for the sound data by a first selection method based on the first determination result; determine an attribute of the content from among a plurality of attribute candidates; and select the processing by a second selection method, which is different from the first selection method, based on a determination result of the attribute, wherein the digital signal processor is configured to execute the processing selected by the at least one processor on the sound data.

Abstract: Disclosed is a method and apparatus for label encoding in a multi-sound event interval. The method includes identifying an event interval in which a plurality of sound events occurs in a sound signal, separating a sound source into sound event signals corresponding to each sound event by performing sound source separation on the event interval, determining energy information for each of the sound event signals, and performing label encoding based on the energy information.

Abstract: Various embodiments comprise systems and methods to monitor operations of a data pipeline. In some examples, a data pipeline receives data inputs, processes the data inputs, and responsively generates and transfers data outputs. Data monitoring circuitry monitors the operations of the data pipeline circuitry, identifies an input change between an initial one of the data inputs and a subsequent one of the data inputs, and identifies an output change between an initial one of the data outputs and a subsequent one of the data outputs. The data monitoring circuitry correlates the input change to the output change, determines a quality threshold for the output change based on the correlation, and determines when the output change falls below the quality threshold. When the output change falls below the quality threshold, the data monitoring circuitry generates and transfers an alert that indicates the input change and the output change.

Abstract: A processing load at a receiving side is reduced in a case where a plurality of classes of audio data is transmitted. A predetermined number of audio streams including coded data of a plurality of groups is generated and a container of a predetermined format having this predetermined number of audio streams is transmitted. Command information for creating a command specifying a group to be decoded from among the plurality of groups is inserted into the container and/or the audio stream. For example, a command insertion area for the receiving side to insert a command for specifying a group to be decoded is provided in at least one audio stream among the predetermined number of audio streams.

Abstract: A method for rendering a soundscape in an environment in form of a real-world indoor scene is provided. The method comprises the steps of recording a present soundscape in the environment, identifying an empty area in the environment, selecting a virtual object to be placed at the empty area, rendering an updated soundscape in the environment using the present soundscape and sound-absorbing properties of the selected virtual object, playing the rendered updated soundscape, and displaying the selected virtual object placed at the empty area.

Abstract: Methods and systems for a product quality inspection system. One aspect is a method of conveying a quality of a furnishing with an e-commerce system, the method comprising presenting with the e-commerce system a listing for a furnishing including a scene with an image of the furnishing, the image of the furnishing having a depth value corresponding to a location of the image of the furnishing in the scene, receiving an audio playback request, and presenting in response to the audio playback request an audio recording of an interaction with the furnishing, wherein the audio recording is indicative of a material composition of the furnishing and the audio recording includes a sound effect based on the depth value.

Abstract: There are disclosed several examples of encoding and decoding technique. In particular, an audio synthesizer for generating a synthesis signal from a downmix signal includes: an input interface for receiving the downmix signal, the downmix signal having a number of downmix channels and side information, the side information including channel level and correlation information of an original signal, the original signal having a number of original channels; and a synthesis processor for generating, according to at least one mixing rule, the synthesis signal using: channel level and correlation information of the original signal; and covariance information associated with the downmix signal.

Abstract: The present technology relates to an information processing device and method, a reproduction device and method, and a program that are capable of performing gain correction more easily. An information processing device includes a gain correction value decision unit that decides a correction value of a gain value for performing gain correction on an audio signal of an audio object in accordance with a direction of the audio object viewed from a listener. The present technology can be applied to a gain decision device and a reproduction device.

Abstract: Systems and methods for enhancing a headset user's own voice include at least two outside microphones, an inside microphone, audio input components operable to receive and process the microphone signals, a voice activity detector operable to detect speech presence and absence in the received and/or processed signals, and a cross-over module configured to generate an enhanced voice signal. The audio processing components includes a low frequency branch comprising low pass filter banks, a low frequency spatial filter, a low frequency spectral filter and an equalizer, and a high frequency branch comprising highpass filter banks, a high frequency spatial filter, and a high frequency spectral filter.

Abstract: A signal processing system and method for delivering spatialized sound by optimizing sound waveforms from a sparse array of speakers to the ears of a user. The system can provide listening areas within a room or space, to provide spatialization sounds to create a 3D audio effect. In a binaural mode, a binary speaker array provides targeted beams aimed towards a user's ears.

Abstract: An apparatus comprises a receiver (201) receiving a multi-channel audio signal representing audio for a scene. An extractor (203) extracts at least one directional audio component by applying a spatial filtering to the multi-channel signal where the spatial filtering is dependent on the multi-channel audio signal. A feature processor (205) determines a set of features for the first directional audio component and a categorizer (207) determines a first audio source category out of a plurality of audio source categories for the directional audio signal in response to the set of features. An assigner (209) assigns a first audio source property to the first directional audio component from a set of audio source properties for the first audio source category. The apparatus may provide very advantageous categorization and characterization of individual audio sources/components present in a multi-channel signal. This may be advantageous e.g. for visualization of audio events.

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: An example may include receiving a page code identifier, determining a priority of the page code identifier, queuing the page code identifier in a paging queue, retrieving content associated with the page code, and forwarding the content to one or more audio devices identified by the page code identifier when the page code has reached a top of the queue.

Abstract: According to some embodiments, a control device for a premises security system that is configured to monitor a premises using a plurality of premises devices is provided. The control device is configured to obtain a plurality of acoustic samples for the premises, detect a sound anomaly in at least one of the plurality of acoustic samples, obtain a verification that the sound anomaly is expected, generate an acoustic model for the premises based at least on the plurality of acoustic samples and the verification that the sound anomaly is expected, receive data representing a detected sound during monitoring of the premises, compare the detected sound with the acoustic model for the premises to determine that the detected sound is unexpected, and initiate a premises security system alert based at least on the detected sound being unexpected.

Abstract: Example embodiments disclosed herein relate to audio object clustering. A method for metadata-preserved audio object clustering is disclosed. The method comprises classifying an audio object into at least a category based rendering mode information metadata. The method further comprises assigning a predetermined number of clusters to the categories and rendering the audio object based on the rendering mode. Corresponding system and computer program product are also disclosed.

Abstract: Methods, systems, and apparatus, for a wearable aural development medical device and system. A medical device is worn by an infant while the infant is in a neonatal intensive care unit (NICU). In particular, the device protects the child's hearing by active filtering, while, at the same time, provides aural stimulation for development. In some implementations, a remote server system also enables people associated the child, e.g., parents and siblings, to provide recordings, such as recordings of their voices, for playback to the child.

Abstract: An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

Abstract: Disclosed are a sound tracing apparatus and a sound tracing method, and the sound tracing apparatus includes a first acceleration structure generation unit configured to generate a first acceleration structure for a static scene in a sound space, an intersection test execution unit configured to perform an intersection test on each of a plurality of dynamic objects constituting a dynamic scene in the sound space to detect whether or not the dynamic object affects a sound propagation path, a second acceleration structure generation unit configured to select the dynamic objects that affect the sound propagation path as a result of the intersection test and then generate the second acceleration structure for the dynamic scene, and a sound generation unit configured to generate a 3D sound by performing sound tracing based on the first and second acceleration structures.

Abstract: A system for sound localization can include a first electronic device having a microphone to detect a sound, and a second electronic device. A processor can be in communication with the first electronic device and the second electronic device. The processor can receive a first signal from the first electronic device corresponding to the detected sound, determine a location of origin of the detected sound based at least in part on the first signal, and provide a second signal to the second electronic device based at least in part on the location of origin.

Abstract: An audio processing method includes processing, by M first virtual speakers, a to-be-processed audio signal to obtain M first audio signals; processing, by N second virtual speakers, the to-be-processed audio signal to obtain N second audio signals; obtain M first head-related transfer functions (HRTFs) centered at a left ear position and N second HRTFs centered at a right ear position; obtain a first target audio signal based on the M first audio signals and the M first HRTFs; and obtain a second target audio signal based on the N second audio signals and the N second HRTFs.

Abstract: Disclosed herein, among other things, are methods and apparatus for providing a time-stamp based controller for synchronization of sink or source sampling rate with external packet rate. A method for wireless communications includes receiving a transmission of a packet using a wireless transceiver of an electronic device, and using a processor of the electronic device to read a first value of a system timer and store the first value as an arrival time-stamp. The packet is decoded and processed by the processor, and sent to an output. When the processed packet is sent, a second value of the system timer is read, adjusted and stored as a departure time-stamp. The arrival time-stamp and the departure time-stamp are used to calculate an adjustment stimulus for a sample rate actuator of the electronic device. The sample rate actuator is configured to maintain synchronization of sampling rate with an external packet rate.