Abstract: Examples are disclosed relating to providing spatialized audio to multiple users. In one example, a computing device presents spatialized audio to multiple users within an environment via communicative connection to one or more wearable spatial audio output devices. For each communicatively connected wearable spatial audio output device, a user-specific subset of audio tracks is generated from a set of audio tracks for a dynamic audio object positioned within the environment based on one or more user-specific parameters. A location of the wearable spatial audio output device is determined relative to the dynamic audio object, and based upon this location, a device-specific spatialized audio mix is generated that includes the user-specific subset of audio tracks. The device-specific spatialized audio mixes are sent to the wearable spatial output devices, and playback of the device-specific spatialized audio mixes are synchronously initiated at each wearable spatial audio output device.

Abstract: A method for controlling a scene sound effect and related products are disclosed. The method comprises: a monitoring service is enabled after electronic equipment is turned on; the electronic equipment monitors a newly created view in the electronic equipment by means of the monitoring service to determine whether there is the newly created view in the electronic equipment, a characteristic of the newly created view having a mapping relationship with an application in the electronic equipment; when the electronic equipment determines that there is the newly created view in the electronic equipment, the electronic equipment determines the application having the mapping relationship with the characteristic of the newly created view in the electronic equipment according to the mapping relationship; and the electronic equipment acquires a scene sound effect corresponding to the application and sets a sound effect of the electronic equipment as the scene sound effect.

Abstract: A signal processing device includes a prior distribution storage that stores a prior distribution group that is a set of prior distributions representing representative spectral shapes of spectral bases of an acoustic event specified as a detection target, and a basis generation unit that, using as input a spectrogram for basis generation, generates an acoustic event basis group that is a set of the spectral bases of the acoustic event specified as a detection target. The basis generation unit performs sound source separation on the spectrogram for basis generation and thereby generates an acoustic event basis group, using respective prior distributions included in a prior distribution group as a constraint for spectral bases of a corresponding acoustic event.

Abstract: A signal processing apparatus and method are provided for separating a plurality of mixture signals from a MIMO system to iteratively obtain a plurality of output signals. The plurality of mixture signals are a response of the MIMO system to a plurality of source signals. The signal processing apparatus comprises a plurality of blind source separators including a first blind source separator based on a first blind source separation technique or algorithm and a second blind source separator based on a second blind source separation technique or algorithm, wherein the first blind source separator is configured to compute a first plurality of preliminary output signals on the basis of a first set of coefficients describing the MIMO system and wherein the second blind source separator is configured to compute a second plurality of preliminary output signals on the basis of a second set of coefficients describing the MIMO system.

Abstract: Methods, systems, and computer readable media for simulating sound propagation are disclosed. According to one method, the method includes decomposing a virtual environment scene including at least one object into a plurality of surface regions, wherein each of the surface regions includes a plurality of surface patches. The method further includes organizing sound rays generated by a sound source in the virtual environment scene into a plurality of path tracing groups, wherein each of the path tracing groups comprises a group of the rays that traverses a sequence of surface patches. The method also includes determining, for each of the path tracing groups, a sound intensity by combining a sound intensity computed for a current time with one or more previously computed sound intensities respectively associated with previous times and generating a simulated output sound at a listener position using the determined sound intensities.

Abstract: An earpiece includes an earpiece housing, wherein the earpiece housing is configured to substantially encompass an opening of an ear canal to physically block ambient sound from entering the ear canal. The earpiece further includes a processor disposed within the earpiece housing, at least one microphone operatively connected to the processor and configured to sense ambient sound, and a speaker operatively connected to the processor. The processor is configured to process the ambient sound from the microphone and add additional ambient sound to produce a modified ambient sound at the speaker.

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: Audio data processor, having: a receiver interface for receiving encoded audio data and metadata related to the encoded audio data; a metadata parser for parsing the metadata to determine an audio data manipulation possibility; an interaction interface for receiving an interaction input and for generating, from the interaction input, interaction control data related to the audio data manipulation possibility; and a data stream generator for obtaining the interaction control data and the encoded audio data and the metadata and for generating an output data stream, the output data stream having the encoded audio data, at least a portion of the metadata, and the interaction control data.

Abstract: A recording method, a recording play method, apparatuses and terminals, where the recording method comprises obtaining, by a terminal, recording data in all sound source directions input by at least three microphones, and generating, by the terminal, a recording file according to the obtained recording data, wherein the recording file saves the recording data in all the sound source directions.

Abstract: A rendering system including a plurality of loudspeakers, at least one microphone and a signal processing unit. The signal processing unit is configured to determine at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between the plurality of loudspeakers and the at least one microphone using a rendering filters transfer function matrix using which a number of virtual sources is reproduced with the plurality of loudspeakers.

Abstract: In general, techniques are described by which to provide priority information for higher order ambisonic (HOA) audio data. A device comprising a memory and a processor may perform the techniques. The memory stores HOA coefficients of the HOA audio data, the HOA coefficients representative of a soundfield. The processor may decompose the HOA coefficients into a sound component and a corresponding spatial component, the corresponding spatial component defining shape, width, and directions of the sound component, and the corresponding spatial component defined in a spherical harmonic domain. The processor may also determine, based on one or more of the sound component and the corresponding spatial component, priority information indicative of a priority of the sound component relative to other sound components of the soundfield, and specify, in a data object representative of a compressed version of the HOA audio data, the sound component and the priority information.

Abstract: Techniques for determining positions of devices within an environment are described herein. In some instances, an environment, such as a home or office of a user, may include an array of devices, some or all of which may couple to a network or to other devices via short-range wireless connections (e.g., Bluetooth®, Zigbee®, etc.). These devices may capture an array of data for providing to a central service, which is configured to analyze the data and, based on this analysis, determine a location of the devices relative to one another. That is, the central service may analyze the data to determine relative distances and orientations between the identified devices within the environment.

Abstract: A processing method and a processing system of an audio multichannel output speaker and a mobile phone are disclosed. The method includes: retrieving an audio and video file and establishing four channel signals for audio signals in the audio and video file with independent channel indications; performing frequency-division processing on the four channel signals respectively, and storing data of the four channel signals after frequency-division processing in order; performing digital-to-analog conversion on the data of the four channel signals respectively, and respectively sending converted analog signal data to speakers corresponding to the four channel signals one-to-one.

Abstract: A signal detection device 10 includes a compression unit 11 which compresses an activation matrix by adding to each column an element of the row corresponding to a basis mapped to the information of the same acoustic element in an activation matrix computed by non-negative matrix factorization using a basis matrix, using the information of acoustic elements making up an acoustic event, mapped to the basis constituting the basis matrix.

Abstract: A system and method for providing user location-based multi-zone media. As a non-limiting example, various aspects of this disclosure provide a system and method that flexibly presents media content (e.g., audio content) to a user as the user moves throughout a premises. As a non-limiting example, various aspects of this disclosure provide a system and method that flexibly presents media content (e.g., audio content) to a user as the user moves throughout a premises.

Abstract: An audio system is provided. The audio system has a first speaker assembly having a first steerable audio beam and a second speaker assembly having a second steerable audio beam. A control system is in communication with the first and second speaker assemblies and is programmed to receive a sweet-spot input signal indicative of a sweet-spot area having stereo surround sound. The controller commands the first and second speaker assemblies to adjust at least one of parameter of the first and second steerable audio beams so that the second steerable audio beam intersects the first steerable audio beam to form the sweet-spot area at the intersection.

Abstract: A method provides a voice in binaural sound to a first user during an electronic communication between the first user wearing a head mounted display (HMD) and a second user. The method includes designating a sound localization point (SLP) with a handheld portable electronic device (HPED), processing the voice of the second user with a digital signal processor (DSP) in the HMD, and moving the voice of the second user in response to a microphone in the HMD receiving a voice command from the first user.

Abstract: Provided are an encoding method of a multichannel signal, an encoding apparatus to perform the encoding method, a multichannel signal processing method, and a decoding apparatus to perform the decoding method. The decoding method may include identifying an N/2-channel downmix signal derived from an N-channel input signal; and generating an N-channel output signal from the identified N/2-channel downmix signal using a plurality of one-to-two (OTT) boxes. If a low frequency effect (LFE) channel is absent in the output signal, the number of OTT boxes may be equal to N/2 where N/2 denotes the number of channels of the downmix signal.

Abstract: In at least one embodiment, an apparatus for enhancing speech prompts in a vehicle is provided. The apparatus includes an audio processor that is electrically coupled to a plurality of loudspeakers in a vehicle. The audio processor being programmed to provide entertainment data to the plurality of loudspeakers for playback in the vehicle and to receive a speech prompt indicative of only a spoken audio output to a driver in the vehicle. The audio processor is further programmed to mute the entertainment data that is played back on a first loudspeaker of the plurality of loudspeakers in response to the speech prompt and to provide the speech prompt to the first loudspeaker for playback to the driver in response to the speech prompt.

Abstract: Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal.

Abstract: A method of presenting an audio signal to a user of a mixed reality environment is disclosed. According to examples of the method, an audio event associated with the mixed reality environment is detected. The audio event is associated with a first audio signal. A location of the user with respect to the mixed reality environment is determined. An acoustic region associated with the location of the user is identified. A first acoustic parameter associated with the first acoustic region is determined. A transfer function is determined using the first acoustic parameter. The transfer function is applied to the first audio signal to produce a second audio signal, which is then presented to the user.

Abstract: A sound adjustment method applied to a sound adjustment system is disclosed. The sound adjustment system includes a sound receiving module, a sound identification module, a sound frequency conversion module and a sound equalizer. The sound adjustment method includes the steps of: receiving a sound signal via the sound receiving module; identifying the sound signal via the sound identification module to determine a type of the sound signal; if the sound signal is a voice signal, executing a frequency conversion of the voice signal via the sound frequency conversion module such that the voice signal becomes a frequency-converted voice signal; if the sound signal is a non-voice signal, adjusting the non-voice signal via the sound equalizer such that the non-voice signal becomes an equalizer-adjusted sound signal.

Abstract: A method expedites processing and playing of binaural sound during an electronic communication between a first user and a second user. An electronic device of the first user convolves sound into binaural sound for the second user before the binaural sound transmits to the electronic device of the second user. In this way, the binaural sound is already convolved and ready to play upon receipt at the electronic device of the second user.

Abstract: The present disclosure relates to systems and methods for directing the audio output of a wearable device having a plurality of speakers. In one implementation, the system may include an image sensor configured to capture one or more images from an environment of the user of the wearable apparatus, a plurality of speakers, and at least one processing device. The at least one processing device may be configured to analyze the one or more images to determine at least one indicator of head orientation of the user of the wearable apparatus, select at least one of the plurality of speakers based on the at least one indicator of head orientation, and output the audio to the user of the wearable apparatus via the selected at least one of the plurality of speakers.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: The description relates to parametric directional propagation for sound modeling and rendering. One implementation includes receiving virtual reality space data corresponding to a virtual reality space. The implementation can include using the virtual reality space data to simulate directional impulse responses for initial sounds emanating from multiple moving sound sources and arriving at multiple moving listeners. The implementation can include using the virtual reality space data to simulate directional impulse responses for sound reflections in the virtual reality space. The directional impulse responses can be encoded and used to render sound that accounts for a geometry of the virtual reality space.

Abstract: Perspective-based dynamic audio volume adjustment. A method obtains multimedia content, the multimedia content including video frames of a scene and audio content for playing in conjunction with display of the video frames. The audio content includes audio portions corresponding to objects in the scene. Based on a selection of a viewing perspective from which to view the scene, the method determines an audio mix for the audio portions given the selected viewing perspective. Determining the audio mix includes determining, for each object of the objects in the scene, a respective audio volume level for an audio portion, of the audio portions, corresponding to the object. The respective audio volume level is based at least in part on a position of the object relative to the selected viewing perspective.

Abstract: An acoustic crosstalk canceller is determined for an asymmetric audio playback device, by determining a transfer function of an acoustic stereo playback path having asymmetries defined by speakers of the playback device. The transfer function is inverted to determine an inverse transfer function. The inverse transfer function is regularised by applying frequency dependent regularisation parameters to obtain an acoustic crosstalk canceller. Also, the inverse transfer function could be regularised for symmetric playback paths by applying aggregated frequency dependent regularisation parameters to obtain an acoustic crosstalk canceller without band branching.

Abstract: A method improves computer performance to provide binaural sound in a telephone call. The method includes providing the telephone call with a voice of a calling party and head-related transfer functions (HRTFs) providing where a voice of the calling party localizes as binaural sound to the called party. The voice of the calling party is convolved with the HRTFs received so the voice externally localizes as the binaural sound to the called party.

Abstract: This disclosure relates to speakers and more specifically to an array speaker for distributing music uniformly across a room. A number of audio drivers can be radially distributed within a speaker housing so that an output of the drivers is distributed evenly throughout the room. In some embodiments, the exit geometry of the audio drivers can be configured to bounce off a surface supporting the array speaker to improve the distribution of music throughout the room. The array speaker can include a number of vibration isolation elements distributed within a housing of the array speaker. The vibration isolation elements can be configured reduce the strength of forces generated by a subwoofer of the array speaker.

Abstract: A method provides a voice in binaural sound to a user. The method includes designating a sound localization point (SLP) with a handheld portable electronic device (HPED) and tracking the SLP with electronic glasses worn on a head of the user. A processor processes the voice with head-related transfer functions (HRTFs) to generate the voice in the binaural sound that localizes at the SLP at a location in empty space at physical object.

Abstract: Sound captured by a first microphone proximate to a severely-impaired first ear is processed to compensate for the hearing loss of the other, less-impaired second ear and then provided to the second ear through a sound transducer or speaker located inside or adjacent the second ear. The same sounds picked up by the first microphone are processed differently for the first and more severely impaired first ear and then provided to the first, severely-impaired ear through its own transducer. Sounds captured by a second microphone proximate to a less-impaired second ear are processed according to the second ear impairment and provided into the second ear by the second and different transducer for the second ear.

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

Abstract: The technology described in this document can be embodied in a method for processing an input signal that represents a signal of interest captured in the presence of noise to generate a de-noised estimate of the signal of interest. The method includes receiving an input signal representing a signal of interest captured in the presence of noise. The method also includes processing at least a portion of the input signal using a digital filter to generate a filtered signal, the digital filter configured to suppress at least a portion of spectrum of the noise. The method further includes processing the filtered signal using a first neural network to generate a de-noised estimate of the signal of interest, wherein the first neural network is trained to compensate for distortions introduced by the digital filter in the signal of interest.

Abstract: The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

Abstract: Systems and methods are described for providing an immersive listening area for one or more listeners. To improve the immersive experience a rear sound bar is placed behind the listeners and the input channels of the rear sound bar receive customized processing to create a virtual rear sound stage. The virtual rear sound stage and a front sound stage created by a front sound bar, combine to create an overall sound stage to encompass the listeners, providing the listeners with an immersive listening experience.

Abstract: To provide an information processing device, an information processing method, and a program capable of sharing a space while maintaining the degree of freedom of a visual line. An information processing device according to the present disclosure includes: a control unit configured to perform control in a manner that a display image generated based on image information which is generated through imaging of an imaging device mounted on a moving object moving in a space, imaging-device posture information which is information regarding a posture of the imaging device, and user view information which is obtained from a user manipulation device manipulated by a user and specifies a region that the user desires to view is displayed in a display region viewed by the user.

Abstract: The present technology relates to an audio processing apparatus and method and a program that make it possible to obtain sound of higher quality. An acquisition unit acquires an audio signal and metadata of an object. A vector calculation unit calculates, based on a horizontal direction angle and a vertical direction angle included in the metadata of the object and indicative of an extent of a sound image, a spread vector indicative of a position in a region indicative of the extent of the sound image. A gain calculation unit calculates, based on the spread vector, a VBAP gain of the audio signal in regard to each speaker by VBAP. The present technology can be applied to an audio processing apparatus.

Abstract: Some disclosed methods may involve receiving audio reproduction data and determining, based on the audio reproduction data, a sound source location at which a sound is to be rendered. A near-field gain and a far-field gain may be based, at least in part, on a sound source distance between the sound source location and a reproduction environment location. Room speaker feed signals may be based, at least in part, on room speaker positions, the sound source location and the far-field gain. Near-field speaker feed signals may be based, at least in part, on the near-field gain, the sound source location and a position of near-field speakers.

Abstract: An information processing apparatus includes a processor that performs a separate process to separate a content signal into a primary component which is an objective sound and a secondary component which is other than the objective sound, a speaker that outputs the primary component, and a transmitter that transmits the secondary component to another apparatus.

Abstract: Embodiments relate to obtaining filter coefficients for a binaural synthesis filter; and applying a compensation filter to reduce artefacts resulting from the binaural synthesis filter; wherein the filter coefficients and compensation filter are configured to be used to obtain binaural audio output from a monaural audio input. The filter coefficients and compensation filter may be applied to the monaural audio input to obtain the binaural audio output. The compensation filter may comprise a timbre compensation filter.

Abstract: One or more chat servers receives voice signals and pose (location and orientation) signals from devices such as VR headsets associated with respective chat participants. For each participant, the server renders a single stream representing the voices of the other participants, with the voice data in each stream being modified to account for the orientation of the head of the receiving participant. The server sends the streams to the participants for whom the streams are tailored. The voice information representing the chat of the other participants in a stream intended for a particular participant can also be modified to account for the distances between participants and orientations of speakers' heads relative to the particular participant for whom the stream is tailored.

Abstract: A system of crosstalk cancelled zone creation in audio playback comprising: main transducers emitting stereo soundwaves of an audio playback; a local system comprising at least two or more close-proximity-transducers (CPTs), each is arranged proximal to one of left and right-side ear canals of a listener. Each of the CPTs comprises: a position tracking device for tracking the relative positions of the main transducers to the CPT and the other CPTs; a control unit for receiving the relative position data from the position tracking device and generating control signal according to the relative position data for the generation of crosstalk cancellation (XTC) soundwaves. Each of the CPTs is configured to generate XTC soundwaves corresponding to the stereo soundwaves arriving at the corresponding ear of the listener. The generated XTC soundwaves are synchronized with the audio playback and with respect to the relative positions.

Abstract: An apparatus for generating a sound field description having a representation of sound field components, including a direction determiner for determining one or more sound directions for each time-frequency tile of a plurality of time-frequency tiles of a plurality of microphone signals; a spatial basis function evaluator for evaluating, for each time-frequency tile of the plurality of time-frequency tiles, one or more spatial basis functions using the one or more sound directions; and a sound field component calculator for calculating, for each time-frequency tile of the plurality of time-frequency tiles, one or more sound field components corresponding to the one or more spatial basis functions evaluated using the one or more sound directions and a reference signal for a corresponding time-frequency tile, the reference signal being derived from one or more microphone signals of the plurality of microphone signals.

Abstract: The present technology relates to a sound processing apparatus, a method, and a program that can reproduce an acoustic field more appropriately. A sound source position correction unit corrects sound source position information indicating a position of each object sound source, on the basis of a hearing position of a sound, and derives corrected sound source position information. A reproduction area control unit calculates, on a basis of an object sound source signal of a sound of an object sound source, the hearing position, and corrected sound source position information obtained by the correction, a spatial frequency spectrum such that a reproduction area is adjusted in accordance with the hearing position provided inside a spherical or annular speaker array. The present technology can be applied to a sound processing apparatus.

Abstract: A sound processing device includes an equalizer that tunes the frequency characteristic so that a frequency characteristic of the sound wave listened in a second environment replicates the frequency characteristic of a sound wave listened in a first environment. A plurality of the equalizers is provided corresponding to a plurality of sound image signals that has respective sound images to be localized in different directions. In addition, the equalizer performs a unique frequency characteristic changing process to the corresponding sound image signal. Each equalizer has a transfer function that cancels the unique change to the frequency characteristic caused in accordance with the sound localization direction of the sound image signal.

Abstract: A method comprising: processing audio signals to rotate a sound scene so that a user selected sound object is automatically at a first position within the rotated sound scene; enabling user editing of audio properties of the selected sound object to create a modified sound object; and causing rendering of the sound scene using the modified sound object.

Abstract: The present invention relates to methods and apparatus for encoding an HOA signal representation (c(t)) of a sound field having an order of N and a number O=(N+1)2 of coefficient sequences to a mezzanine HOA signal representation (wMEZZ(t)) is generated that consists of an arbitrary number I of virtual loudspeaker signals wMEZZ,1(t), wMEZZ,2(t), . . . , wMEZZ,I(t). The present invention further relates to methods and apparatus for decoding a reconstructed HOA signal representation from the mezzanine HOA signal representation.

Abstract: A method and apparatus for providing binaural audio for a headset is provided. In one embodiment, a method includes encoding audio signals to provide binaural audio to a headset. The method includes receiving audio signals from a microphone array comprising a first plurality of elements and applying far-field array processing to the audio signals to generate a first plurality of channels. The channels can be beam channels and each channel is associated with a particular beam angle. The method further includes selecting a second plurality of channels from the first plurality of channels that is a subset of the first plurality of channels. The method includes encoding the audio signals from the selected second plurality of channels with information associated with the particular beam angle for each of the selected second plurality of channels. The encoded audio signals are configured to provide binaural audio to a headset.

Abstract: A method displays an emoji that enables a user to select how or where sound will localize to a listener of the sound associated with the emoji. During an electronic communication between a first user and a second user, the first user selects how or where sound of the emoji will localize to the second user. When the second user receives the emoji, the sound plays to the second user how or where as selected by the first user.