Abstract: A method for channel identification of a multi-channel audio signal comprising X>1 channels is provided. The method comprises the steps of: identifying, among the X channels, any empty channels, thus resulting in a subset of Y?X non-empty channels; determining whether a low frequency effect (LFE) channel is present among the Y channels, and upon determining that an LFE channel is present, identifying the determined channel among the Y channels as the LFE channel; dividing the remaining channels among the Y channels not being identified as the LFE channel into any number of pairs of channels by matching symmetrical channels; and identifying any remaining unpaired channel among the Y channels not being identified as the LFE channel or divided into pairs as a center channel.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).

Abstract: A sound system includes a seat, a bone conduction system configured to be disposed within the seat, and at least one directional speaker configured to provide audio through directional acoustic delivery contemporaneously with the bone conduction system. The bone conduction system configured to provide audio through bone conduction sound.

Abstract: Circuitry for acoustic crosstalk cancellation between first and second acoustic signals, the circuitry comprising: crosstalk cancellation circuitry configured to: receive a first audio signal and, based on the received first audio signal, generate a first crosstalk cancellation signal; receive a second audio signal and, based on the received second audio signal, generate a second crosstalk cancellation signal; combine the first crosstalk cancellation signal with a signal indicative of the second audio signal to generate a first crosstalk cancellation circuitry output signal; and combine the second crosstalk cancellation signal with a signal indicative of the first audio signal to generate a second crosstalk cancellation circuitry output signal; and output stage circuitry configured to: receive the first crosstalk cancellation circuitry output signal and, based on the received first crosstalk cancellation circuitry, generate a first drive signal for driving a first speaker to generate the first acoustic signal

Abstract: The information processing device includes an acquisition unit, a determination unit, and a signal processing unit. The acquisition unit acquires one or a plurality of other sound elements. The determination unit determines importance levels of the sound elements acquired by the acquisition unit. The signal processing unit changes a sound source position of at least one of a sound element of a content being reproduced or another sound element according to the importance levels of the sound elements determined by the determination unit. Consequently, even in a case where the sound element of the reproduced content and the another sound element are simultaneously generated, sound interference between the sound element of the reproduced content and the another sound element can be suppressed.

Abstract: There is provided a signal processing device that includes a relative azimuth prediction unit that predicts, on the basis of a delay time in accordance with a distance from a virtual sound source to a listener, a relative azimuth of the virtual sound source when a sound of the virtual sound source reaches the listener, and a binaural-room impulse response (BRIR) generation unit that acquires a head-related transfer function of the relative azimuth for each one of a plurality of the virtual sound sources and generates a BRIR on the basis of a plurality of the acquired head-related transfer functions.

Abstract: A device including: first and second input units providing first and second input signals of first and second audio tracks, a decomposition unit to decompose the first input audio signal to obtain decomposed signals, a playback unit to start playback of a first output signal obtained from recombining at least first and second decomposed signals at first and second volume levels, respectively, and a transition unit for performing a transition between playback of the first output signal and playback of a second output signal obtained from the second input signal. The transition unit is adapted for reducing the first/second volume levels according to first/second transition functions. The device includes an analyzing unit to analyze an audio signal to determine a song part junction between two song parts. The transition time interval of at least one of the transition functions is set such as to include the song part junction.

Abstract: An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF.

Abstract: A system may provide for the generation of spatial audio for audiovisual conferences, video conferences, etc. (referred to herein simply as “conferences”). Spatial audio may include audio encoding and/or decoding techniques in which a sound source may be specified at a location, such as on a two-dimensional plane and/or within a three-dimensional field, and/or in which a direction or target for a given sound source may be specified. A conference participant's position within a conference user interface (“UI”) may be set as the source of sound associated with the conference participant, such that different conference participants may be associated with different sound source positions within the conference UI.

Abstract: In one aspect, a device includes at least one processor and storage accessible to the at least one processor. The storage includes instructions executable by the at least one processor to track a location of at least a segment of a wire of a virtual reality (VR) headset. The instructions are also executable to, based on the location and responsive to a trigger, present a notification regarding the location on a display of the VR headset.

Abstract: A system to output a cross-talk cancelled spatial audio signal to two speakers, including a user tracking module to provide a location and orientation of a head of a user, and a time-domain dynamic crosstalk cancellation module to receive a spatial audio signal, the location and orientation of the head of the user from the user tracking module, a location for each of the two speakers, and generate the time-domain crosstalk cancelled spatial audio signal based thereon.

Abstract: What is described is a method for loss concealment of spatial audio parameters, the spatial audio parameters having at least a direction of arrival information; the method having the following steps: receiving a first set of spatial audio parameters having at least a first direction of arrival information; receiving a second set of spatial audio parameters, having at least a second direction of arrival information; and replacing the second direction of arrival information of a second set by a replacement direction of arrival information derived from the first direction of arrival information, if at least the second direction of arrival information or a portion of the second direction of arrival information is lost or damaged.

Abstract: A microphone can be provided on a charging case for wireless earbuds. This can be used for quality recording and microphone placement, as well as assisting in echo cancellation algorithms by measuring environmental noise. The microphone can be movably mounted to adjust for noise or echo cancelation on another device such as a cell phone.

Abstract: At least one exemplary embodiment is directed to a method of earphone feature control by using a graphic user interface on a device linked to the earphone.

Abstract: An aspect of the disclosure relates to an apparatus, including an audio interface including first and second outputs; a crosstalk cancelling circuit including first and second inputs coupled to the first and second outputs of the audio interface, respectively; a first summer including a first input coupled to the first output of the audio interface and a second input coupled to a first output of the crosstalk cancelling circuit; a second summer including a first input coupled to the second output of the audio interface and a second input coupled to a second output of the crosstalk cancelling circuit; a first digital to analog converter (DAC) including an input coupled to an output of the first summer; and a second DAC including an input coupled to an output of the second summer.

Abstract: A speaker system includes a first speaker and a second speaker. The first speaker includes a first sound signal input interface configured to acquire a first sound signal, and a first sound emitter configured to output a first sound based on the first sound signal in a state in which a position of a host device is fixed. The second speaker includes a second sound signal input interface configured to acquire a second sound signal, a moving body that includes a position information acquisition interface configured to acquire information relating to position and that is configured to move based on the information relating to position, and a second sound emitter configured to output a second sound based on the second sound signal.

Abstract: A bone conduction speaker is built into a seat of a vehicle and faces the skull of an occupant seated in the seat. An acoustic control ECU individually selects sound to be output from the bone conduction speaker for each of the occupants in the vehicle, according to attribute information of the occupant recorded in a portable terminal device carried by the occupant, and causes the selected sound to be individually output from the bone conduction speaker to each of the occupants.

Abstract: System and methods for providing dynamic positional audio are disclosed. Methods can comprise determining availability of one or more devices to output audio and determining a location of the one or more available devices. Audio information can be received and at least a portion of the audio information can be configured to generate assigned audio information based on the determined location of the available devices. The assigned audio information can be transmitted to the available devices.

Abstract: A system and method that incorporates the subject disclosure may include, for example, receiving a multichannel audio stream; forming a front channel audio stream of the multichannel audio stream, including combining a first subset of audio channels of the multichannel audio stream to form the front channel audio stream; forming a surround channel audio stream of the multichannel audio stream including combining a second subset of audio channels of the multichannel audio stream to form the surround channel audio stream; providing the front channel audio stream to a primary set of speakers positioned in front of a listener and providing the surround channel audio stream to a supplemental speaker positioned behind the listener; and synchronizing the front channel audio stream and the surround channel audio stream. Additional embodiments are disclosed.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

Abstract: An audio apparatus includes a signal receiver configured to receive a left-channel signal and a right-channel signal of a stereo audio signal; at least one filter configured to block a signal in a band exceeding a predetermined frequency based on the frequency bands of each of the received left-channel signal and right-channel signal; and a controller configured to generate a center-channel signal of the stereo audio signal based on the left-channel signal and the right-channel signal passing through the filter, to calculate a signal level sensitivity between the left-channel signal and the right-channel signal passing through the filter, to determine a similarity between the left-channel signal and the right-channel signal based on the calculated signal level sensitivity, and to control the generated center-channel signal to be output in response to the determined similarity.

Abstract: Methods and systems are provided for gaming headset with enhanced off-screen awareness. An audio system that is used for outputting audio signals to a user may be configured for identifying in audio signals, based on parameters or criteria associated with a user of the system, one or more components in the audio signals; determining characteristics of the one or more components based on perception of the audio signals by the user; and adjusting based on the characteristics of the one or more components, the one or more components and/or one or more other components in the audio signals. The audio system may include circuits for handling the required functions. The circuits may include a detection circuit configured for identify the one or more components, one or more adjustment circuits for applying the adjustments, and a controller circuit for controlling applying the adjustments.

Abstract: A device may be configured to: obtain at least a first audio signal and a second audio signal, wherein the first audio signal and the second audio signal are captured with a microphone array comprising at least two microphones; identify a first direction for a plurality of frequency bands; and identify a second direction for the plurality of frequency bands; wherein the first direction and the second direction are identified based, at least partially, on spatial analysis of the first audio signal and the second audio signal.

Abstract: A non-transitory computer-readable medium is provided. The non-transitory computer-readable medium having computer instructions stored therein, which when executed by one or more processors, cause the one or more processors to perform operations. The operations comprise: determining a channel combination scheme for a current frame, where the determined channel combination scheme for the current frame is one of a plurality of channel combination schemes; and determining a coding mode of the current frame based on a channel combination scheme for a previous frame and the channel combination scheme for the current frame, where the coding mode of the current frame is one of a plurality of coding modes.

Abstract: A signal processing device includes: a processor; and a memory having instructions. The instructions, when executed by the processor, cause the signal processing device to perform operations. The operations include performing a modulation processing of modulating a sound signal by using a modulation parameter based on an interaural phase difference at a listening position of the sound signal.

Abstract: Systems and methods for amplifying and/or attenuating audio signals are disclosed. In one implementation, a system for selectively amplifying audio signals may include at least one microphone for capturing sounds from an environment of the user and a processor. The processor may be programmed to receive an audio signal representative of sounds captured by the at least one microphone; determine whether the audio signal comprises speech by a user of the system; subject to the audio signal comprising speech by the user, modify the audio signal by attenuating a first part of the audio signal comprising the speech by the user; subject to the audio signal comprising audio other than speech by the user, modify the audio signal by amplifying a second part of the audio signal comprising audio other than the speech by the user; and transmit the modified audio signal to a hearing interface device.

Abstract: A method of processing audio content including a plurality of audio elements comprises: clustering the plurality of audio elements into a plurality of clusters of audio elements; and for a cluster among the plurality of clusters: for each audio element in the cluster, determining a measure of energy that the audio element contributes to the cluster; for at least one audio element in the cluster, determining a compensation gain based at least in part on the measures of energy for the audio elements in the cluster; and applying the compensation gain to the at least one audio element in the cluster.

Abstract: A 3D sound spatializer provides delay-compensated HRTF interpolation techniques and efficient cross-fading between current and delayed HRTF filter results to mitigate artifacts caused by interpolation between HRTF filters and the use of time-varying HRTF filters.

Abstract: An audio headset receives one or more audio signals carrying one or more audio channels and processes the audio channels to generate stereo signals for output to a left and a right speaker of the audio headset. The processing determines a number of the audio channels carried in the received audio signal(s), adjusts level(s) of the audio channels based on the determined number of audio channels and/or adjusts gain and/or phase of the audio channels to control a perceived location of a listener wearing the headset relative to a source of sounds carried in the stereo signals.

Abstract: A system and method for making categorized music tracks available to end user applications. The tracks may be categorized based on computer-derived rhythm, texture and pitch (RTP) scores for tracks derived from high-level acoustic attributes, which is based on low level data extracted from the tracks. RTP scores are stored in a universal database common to all of the music publishers so that the same track, once RTP scored, does not need to be re-RTP scored by other music publishers. End user applications access an API server to import collections of tracks published by publishers, to create playlists and initiate music streaming. Each end user application is sponsored by a single music publisher so that only tracks capable of being streamed by the music publisher are available to the sponsored end user application.

Abstract: An audio processing system may include at least one memory configured to maintain a plurality of audio signal processing algorithms and audio components, and a server having a processor. The processor may be configured to present an interface displaying at least a portion of the signal processing algorithms and audio components, receive user selection of at least one of the audio components, present the at least one selected audio component via the interface, receive user selection of at least one of signal processing algorithms to apply to the at least one selected audio component, apply the at least one selected algorithm to the at least one selected audio component, and provide a measurement of the at least one selected algorithm and the at least one selected audio component via the interface.

Abstract: The present invention relates to a wireless speaker chair, and more particularly, to a wireless speaker chair capable of providing a sense of space and a three-dimensional effect to a user who sits in the chair through sounds output from a plurality of speaker channels and allowing the plurality of speaker channels to wirelessly receive sound signals and power which are received to output sounds.

Abstract: A method for mapping a plurality of input channels of an input channel configuration to output channels of an output channel configuration includes providing a set of rules associated with each input channel of the plurality of input channels, wherein the rules define different mappings between the associated input channel and a set of output channels. For each input channel of the plurality of input channels, a rule associated with the input channel is accessed, determination is made whether the set of output channels defined in the accessed rule is present in the output channel configuration, and the accessed rule is selected if the set of output channels defined in the accessed rule is present in the output channel configuration. The input channels are mapped to the output channels according to the selected rule.

Abstract: The present disclosure relates to a device for processing an audio signal. The device may include a first acoustic-electric transducer and a second acoustic-electric transducer. The first acoustic-electric transducer may have a first frequency response, and may be configured to detect the audio signal and generate a first sub-band signal according to the detected audio signal. The second acoustic-electric transducer may have a second frequency response, the second frequency response being different from the first frequency response. The second acoustic-electric transducer may be configured to detect the audio signal and generate a second sub-band signal according to the detected audio signal.

Abstract: A method for increasing stability of an inter-channel time difference (ICTD) parameter in parametric audio coding, wherein a multi-channel audio input signal comprising at least two channels is received. The method comprises obtaining an ICTD estimate, ICTDest(m), for an audio frame m and a stability estimate of said ICTD estimate, and determining whether the obtained ICTD estimate, ICTDest(m), is valid. If the ICTDest(m) is not found valid, and a determined sufficient number of valid ICTD estimates have been found in preceding frames, a hang-over time is determined using the stability estimate and a previously obtained valid ICTD parameter, ICTD(m?1), is selected as an output parameter, ICTD(m), during the hang-over time. The output parameter, ICTD(m), is set to zero if valid ICTDest(m) is not found during the hang-over time.

Abstract: Systems and methods provide for advanced video editing techniques using sampling patterns. In one example, a computing device can receive a selection of a clip of a video and a sampling pattern. The computing device can determine a respective number of frames to sample from the clip for each interval of time over a length of time for a new clip. For example, the computing device can determine a function corresponding the pattern that relates time and the number of frames to sample, a histogram corresponding to the pattern, or a definite integral corresponding to the pattern, among other approaches. The computing device can extract these numbers of frames from the clip and generate the new clip from the extracted frames. The computing device can present the new clip as a preview and send the new clip to other computing devices.

Abstract: A system and method for audio combination and playback which replicates the sounds made by a crowd gathered together in one place.

Abstract: Systems and methods for a media guidance application that adjusts output parameters of media assets delivered to output devices based on user preferences of users near the output devices. For example, the media guidance application may adjust the volume to be higher at a speaker near a first user who enjoys a particular media asset and lower at a speaker near a second user who dislikes the media asset.

Abstract: A signal processing method and Automotive Audio System 290 comprising a tripolar loudspeaker configuration housed in at least one automotive head-rest assembly 200 or 500, whose radiation pattern, in conjunction with inter-element delays and other design features, is such that that passengers are afforded temporal and amplitude cues for achieving the desired soundfield appropriate for a variety of audio program material. Optionally, some or all of the headrest assembly transducers are aligned and configured with a Floating Waveguide member 470.

Abstract: A method performed a local device that is communicatively coupled with several remote devices, the method includes: receiving, from each remote device with which the local device is engaged in a communication session, an input audio stream; receiving, for each remote device, a set parameters; determining, for each input audio stream, whether the input audio stream is to be 1) rendered individually or 2) rendered as a mix of input audio streams based on the set of parameters; for each input audio stream that is determined to be rendered individually, spatial rendering the input audio stream as an individual virtual sound source that contains only that input audio stream; and for input audio streams that are determined to be rendered as the mix of input audio streams, spatial rendering the mix of input audio streams as a single virtual sound source that contains the mix of input audio streams.

Abstract: A conference terminal and an echo cancellation method for a conference are provided. In the echo cancellation method, a synthetic speech signal is received. The synthetic speech signal includes a user speech signal of a speaking party corresponding to a first conference terminal of multiple conference terminals and an audio watermark signal corresponding to the first conference terminal. One or more delay times corresponding to the audio watermark signal are detected in a received audio signal. The received audio signal is recorded through a sound receiver of a second conference terminal of the conference terminals. An echo in the received audio signal is canceled according to the delay time.

Abstract: Disclosed are methods of encoding and decoding an audio signal using side information, and an encoder and a decoder for performing the methods. The method of encoding an audio signal using side information includes identifying an input signal, the input signal being an original audio signal, extracting side information from the input signal using a learning model trained to extract side information from a feature vector of the input signal, encoding the input signal, and generating a bitstream by combining the encoded input signal and the side information.

Abstract: An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF.

Abstract: A device for reducing crosstalk in an audio system that has first and second pairs of loudspeakers and first and second audio sources. The device is connected to each audio source and to at least the first pair of loudspeakers. The device includes a module for acquiring first audio signals from the first source and second audio signals from the second source, a module for determining crosstalk reduction filters resulting from a loudspeaker of the second pair, a module for calculating corrective signals, by applying the reduction filters to the second audio signals, and a module for generating corrected audio signals for the first pair, obtained from the first audio signals and corrective signals. Each reduction filter is obtained from transfer functions, each representing an acoustic path between a loudspeaker and a user's ear.

Abstract: A system comprises a sound source to output an audio signal to a plurality of speakers arranged in a plurality of locations in a vehicle. A laser device is disposed on an object representing a human head placed in a seat of the vehicle to scan the locations of the speakers. A microphone is disposed in each ear of the object. A controller routes the audio signal to the speakers one speaker at a time, receives audio signals received via the microphones, and compiles binaural acoustic data for the vehicle based on the audio signals received via the microphones. The controller receives scan data from the laser device and generates geometric data for the vehicle based on the scan data. The controller generates head-related transfer functions (HRTFs) for the object based on the binaural acoustic data and the geometric data for the vehicle.

Abstract: An encoding system encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).

Abstract: A method for supporting a multi-conversation mode for a vehicle of the present disclosure may include: receiving user information of the multi-conversation mode and at least one of a message or speech from a conversation partner participating in the multi-conversation mode; allocating sound spaces on the basis of the user information; and assigning directivity to speech generated on the basis of at least one of the message or the speech and outputting the speech to an allocated space.

Abstract: A gaming unit and producing a game-audio output of a varying volume level and a chat-voice output is connected to headphones adapted to be worn by a player of the unit and connected to the gaming unit for making the chat-voice output and the game-audio output audible to the player. A circuit or software associated with the headphones can increase the chat-voice volume level generally proportionately to the game-audio volume level.

Abstract: A method including: obtaining at least one spatial audio signal including at least one audio signal, wherein the at least one spatial audio signal at least partially defines an audio scene; obtaining at least one augmentation audio signal; determining at least two audio objects based upon the at least one augmentation audio signal; determining audio-object dependency information for the determined at least two audio objects; and augmenting the audio scene based, at least partially, on both the determined at least two audio objects and the determined audio-object dependency information.