Abstract: Systems and processes for operating an intelligent automated assistant are provided. An examples process of operating an intelligent automated assistant includes, at an electronic device with one or more processors and memory, receiving audio input, determining a direct-to-reverberant energy ratio based on the audio input, and determining a head pose of a user based on the direct-to-reverberant energy ratio.

Abstract: An audio processor for providing loudspeaker signals on the basis of input signals, like channel signals and/or object signals, obtains an information about the position of a listener and an information about the position of a plurality of loudspeakers, or sound transducers. The audio processor selects one or more loudspeakers for a rendering of the objects and/or of the channel objects and/or of the adapted signals, derived from the input signals. The selection depends on the information about the position of the listener and about the positions of the loudspeakers and takes into consideration the information about one or more acoustic obstacles. The audio signal processor renders the objects/channel objects/adapted signals derived from the input signals, in dependence on the information about the position of the listener and about positions of the loudspeakers, in order to obtain the loudspeaker signals, such that a rendered sound follows a listener.

Abstract: An audio system in a local area providing an audio signal to a headset of a remote user is presented herein. The audio system identifies sounds from a human sound source in the local area, based in part on sounds detected within the local area. The audio system generates an audio signal for presentation to a remote user within a virtual representation of the local area based in part on a location of the remote user within the virtual representation of the local area relative to a virtual representation of the human sound source within the virtual representation of the local area. The audio system provides the audio signal to a headset of the remote user, wherein the headset presents the audio signal as part of the virtual representation of the local area to the remote user.

Abstract: Methods and apparatus assist listeners in distinguishing between electronically generated binaural sound and physical environment sound while the listener wears a wearable electronic device that provides the binaural sound to the listener. The wearable electronic device generates a visual alert or audio alert when the electronically generated binaural sound occurs.

Abstract: The present disclosure provides a method for playing audio data, and belongs to the field of computer technology. The method includes: in a process of playing audio, partitioning left-channel audio data of the audio into audio data of a preset number of frequency sub-bands and partitioning right-channel audio data of the audio into audio data of the preset number of frequency sub-bands; calibrating the left-channel audio data of respective frequency sub-bands with left-channel calibration parameter values of respective frequency sub-bands to obtain left-channel audio data to be played; calibrating the right-channel audio data of respective frequency sub-bands with right-channel calibration parameter values of respective frequency sub-bands to obtain right-channel audio data to be played; and playing, over a left channel, the left-channel audio data to be played, and playing, over a right channel, the right-channel audio data to be played.

Abstract: An apparatus for mapping a first input channel and a second input channel of an input channel configuration to at least one output channel of an output channel configuration, wherein each input channel and each output channel has a direction in which an associated loudspeaker is located relative to a central listener position, configured to map the first input channel to a first output channel of the output channel configuration; and despite of the fact that an angle deviation between a direction of the second input channel and a direction of the first output channel is less than an angle deviation between a direction of the second input channel and the second output channel and/or is less than an angle deviation between the direction of the second input channel and the direction of the third output channel, map the second input channel to the second and third output channels by panning between the second and third output channels.

Abstract: An audio processor for providing loudspeaker signals on the basis of input signals, like channel signals and/or object signals, obtains an information about the position of a listener and an information about the position of a plurality of loudspeakers, or sound transducers. The audio processor selects one or more loudspeakers for a rendering of the objects and/or of the channel objects and/or of the adapted signals, derived from the input signals. The selection depends on the information about the position of the listener and about the positions of the loudspeakers and takes into consideration the information about one or more acoustic obstacles. The audio signal processor renders the objects/channel objects/adapted signals derived from the input signals, in dependence on the information about the position of the listener and about positions of the loudspeakers, in order to obtain the loudspeaker signals, such that a rendered sound follows a listener.

Abstract: Disclosed is a system configured to determine filter parameters of an audio precompensation filter for the compensation of an associated sound generating system having at least two loudspeakers. The system is configured to obtain sound field models describing the sound field in at least one region of space designated the bright zone and the sound field in at least one region of space designated the dark zone. The system is also configured to obtain a target sound field in the bright zone. Further, the system is configured to determine filter parameters of the audio precompensation filter so that a criterion function is optimized under the constraint of causality and stability of the dynamics of the audio precompensation filter to enable reproduction of a desired target sound field in the bright zone, while reproducing as little sound as possible in the dark zone.

Abstract: Provided is an information processing device that processes a dialogue of an audio agent. The information processing device includes an acquisition unit that acquires audio information of an agent device that is played back through interaction with a user and audio information of other contents different from the audio information of the agent device, and a controller that performs sound field control processing on an audio output signal based on the audio information of the agent device acquired by the acquisition unit. The controller performs wavefront composition of pieces of audio output from a plurality of speakers, controls a sound field of the agent device, and avoids mixing with the other contents different from the audio information of the agent device.

Abstract: Techniques are provided to transform external sound from an external position of an external audio space to an internal position of an internal audio space where the external sound would be heard naturally. The techniques involve capturing the external sound with a microphone and processing the captured external sound to appear as if it was captured at the internal position of the internal audio space, which may correspond to an output of a driver or a user's eardrum. Then the processed captured external sound may be conditioned for the individual user or further combined with conditioned audio content to create an augmented audio experience for the user. Any combination of the techniques can be implemented in real time.

Abstract: Systems and methods are disclosed for modelling of individual acoustic transfer functions relative to the audition of an individual in three-dimensional space. A method is provided for modelling sets of acoustic transfer functions specific to an individual according to a multiplicity of directions in space, where a set of acoustic transfer functions specific to the individual in a given direction is determined depending on the result of a statistical analysis of a plurality of distinct stimuli emitted in the direction of the individual. A stimulus can be dependent on at least one set of predetermined acoustic transfer functions that are associated with the given direction, and on responses received from the individual to each emitted stimulus.

Abstract: A camera input can be used by the computer to support audio spatialization or to improve audio spatialization of an application that already supports it. A computer system may to support audio spatialization, for example, by modifying the relative latency or relative amplitude of the rendered audio packets. If a sound is intended, for example, to be located on the left side of the user, then the audio channel that is rendered on the headset speaker located on the user's left ear may have a somewhat decreased latency and increased amplitude compared to the other audio channel.

Abstract: Embodiments relate to audio processing for opposite facing speaker configurations that results in multiple optimal listening regions around the speakers. A system includes a left speaker and a right speaker in an opposite facing speaker configuration, and a crosstalk cancellation processor connected with the left speaker and the right speaker. The crosstalk cancellation processor applies a crosstalk cancellation to an input audio signal to generate left and right output channels. The left output channel is provided to the left speaker and the right output channel is provided to the right speaker to generate sound including multiple crosstalk cancelled listening regions that are spaced apart.

Abstract: An apparatus includes a processor configured to receive one or more media signals associated with a scene. The processor is also configured to identify a spatial location in the scene for each source of the one or more media signals. The processor is further configured to identify audio content for each media signal of the one or more media signals. The processor is also configured to determine one or more candidate spatial locations in the scene based on the identified spatial locations. The processor is further configured to generate audio to playback as virtual sounds that originate from the one or more candidate spatial locations.

Abstract: An audio signal processing device includes a memory; and a processor configured to execute receiving a command to determine one set of parameters from among parameters calculated according to positions at which a sound is localized, the positions being set relative to positions of audio output devices; and processing a sound to be output from the audio output devices, by using the one set of parameters determined based on the received command.

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

Abstract: 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: Appropriate audio processing is selected. A signal processing apparatus includes: an audio signal output unit that outputs a plurality of test sounds in a superimposed manner; a controller that prompts a listener to select a test sound having a specific sense of localization out of the plurality of test sounds; a receiver that acquires results of the selection made by the listener; and an audio signal processing unit that performs audio signal processing associated with the results of the selection on an input signal.

Abstract: An auditory attention tracking system may modify external sounds in a manner that enables the system to detect, from EEG readings taken while a user listens to the modified sounds, which sound the user is paying attention to. Multiple microphones may record external sounds, and the sounds may be separated. A computer may modify one or more of the sounds, then remix the sounds, and then cause speakers to audibly output the modified, remixed sound. EEG measurements may be taken while the user listens to this modified sound. A computer may detect patterns in the EEG readings and may, based on the patterns, predict which external sound the user is paying attention to. The system may further modify the external sounds, to make the sound of interest more perceptually prominent. Or, based on the detected sound of interest, the system may modify a human-computer interaction.

Abstract: Media content playback is based on an identified geographic location of a sound source. A geolocation of a target venue is determined. A first geolocation of a media playback device is determined. An indication that the media playback device is within a threshold distance of the target venue is received. Media content associated with the target venue is transferred to the media playback device, wherein one or more characteristics of the media content are dynamically modified to simulate audio originating from the target device. The one or more characteristics of the media content can include audio content characteristics associated with a timing filter, a loudness filter, and a timbre filter.

Abstract: An apparatus and method of rendering audio. The method includes deriving filters by defining a binaural error, defining an activation penalty, and minimizing a cost function that is a combination of the binaural error and the activation penalty. In this manner, the listening experience is improved by reducing the signal level output by loudspeakers further from an audio object's desired position.

Abstract: A portable electronic device (PED) divides an area around a user into a three-dimensional (3D) zone. A wearable electronic device worn on a head of the user displays the 3D zone in response to the wearable electronic device detecting that the user is leaving the zone. The wearable electronic device plays binaural sound that emanates to the user from sound localization points (SLPs) inside the zone.

Abstract: A system includes a first communication system and a second communication system, each with a microphone, an array of speakers arranged to encircle a user of the communication system, a controller, and an array of antenna elements. In each communication system, the array of antenna elements transmits and receives over 360 degrees together, a positional correspondence between the array of speakers and the array of antenna elements is fixed, and the controller controls audio to be output only by speakers among the first array of speakers that correspond to antenna elements among the first array of antenna elements that receive communication.

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: This application discloses an image display method performed at an electronic device. The electronic device obtains a target location of a virtual object in a map scene displayed on a screen of the electronic device. After identifying a virtual building within a preset range centered on the target location, the electronic device gradually decreases the virtual building from a display height of the virtual building to a first target display shape of the virtual building. Next, the electronic device displays the first target display shape of the virtual building in the map scene. In response to a movement of the virtual object in the map scene, the electronic device gradually resumes the virtual building from the first target display shape to the display height when the virtual building is outside the preset range centered on the updated target location of the virtual object in the map scene.

Abstract: Electronic gaming machines that optimize audio during game play or otherwise based on a determined ear structure of a player, and in various embodiments also on a determined location of the player's ear in real-time.

Abstract: The present document describes a method (600) for estimating source parameters of audio sources (101) from mix audio signals (102), with. The mix audio signals (102) comprise a plurality of frames. The mix audio signals (102) are representable as a mix audio matrix in a frequency domain and the audio sources (101) are representable as a source matrix in the frequency domain. The method (600) comprises updating (601) an un-mixing matrix (221) which is configured to provide an estimate of the source matrix from the mix audio matrix, based on a mixing matrix (225) which is configured to provide an estimate of the mix audio matrix from the source matrix. Furthermore, the method (600) comprises updating (602) the mixing matrix (225) based on the un-mixing matrix (221) and based on the mix audio signals (102). In addition, the method (600) comprises iterating (603) the updating steps (601, 602) until an overall convergence criteria is met.

Abstract: A sound signal processor includes a memory storing instructions and a processor configured to implement the stored instructions to execute a plurality of tasks, the tasks including a sound signal input task configured to obtain a sound signal, a beat detection task configured to detect a beat in the sound signal, and a processing task configured to perform an effect processing on the sound signal in accordance with a timing of the detected beat.

Abstract: Head-tracking comprises remixing audio input data or channels to discrete device audio reproduction channels, to alter the audio input data or channels' perceived location in 3D space, in a manner compensating for relative head movement (compensatory manner), based upon provided head location data. Phase (delay) of the desired audio input data or channels is altered to shift the perceived sonic image location in 3D space, in a compensatory manner that correlates to a desired location relative to a designated device reference point. Amplitude of the desired audio input data or channels is altered to shift a perceived sonic image location in 3D space, in a compensatory manner that correlates to the desired location relative to the designated device reference point. Altering phase (delay) is performed using a constant delay function for each of the desired audio input data or channels that is independently varied based on current head location data.

Abstract: Systems and methods are described for detecting and remedying potential incongruence in a video conference. A camera of a video conferencing system may capture video images of a conference room. A processor of the video conferencing system may identify locations of a plurality of participants within an image plane of a video image. Using face and shape detection, a location of a center point of each identified participant's torso may be calculated. A region of congruence bounded by key parallax lines may be calculated, the key parallax lines being a subset of all parallax lines running through the center points of each identified participant. When the audio device location is not within the region of congruence, audio captured by an audio device may be adjusted to reduce effects of incongruence when the captured audio is replayed at a far end of the video conference.

Abstract: In general, various aspects of the techniques are directed to audio rendering for low frequency effects. A device comprising a memory and a processor may be configured to perform the techniques. The memory may store audio data representative of a soundfield. The processor may analyze the audio data to identify spatial characteristics of low frequency effects components of the soundfield, and process, based on the spatial characteristics, the audio data to render a low frequency effects speaker feed. The processor may also output the low frequency effects speaker feed to a low frequency effects capable speaker.

Abstract: Examples of the disclosure describe systems and methods for presenting an audio signal to a user of a wearable head device. According to an example method, a source location corresponding to the audio signal is identified. An acoustic axis corresponding to the audio signal is determined. For each of a respective left and right ear of the user, an angle between the acoustic axis and the respective ear is determined. For each of the respective left and right ear of the user, a virtual speaker position, of a virtual speaker array, is determined, the virtual speaker position collinear with the source location and with a position of the respective ear. The virtual speaker array includes a plurality of virtual speaker positions, each virtual speaker position of the plurality located on the surface of a sphere concentric with the user's head, the sphere having a first radius.

Abstract: Disclosed are devices, systems and methods for binaural spatial audio processing based on a pair of head-related transfer functions (HRTFs) for each of a listener's two ears to synthesize a binaural sound that seems to come from a particular point in space. Applications of the disclosed devices, systems and methods include digital audio reproduction, recording, and multimedia applications including virtual reality and augmented reality experiences.

Abstract: Systems and methods for using auditorily-induced vection (AIV) to enhance a person's attitude awareness are provided herein. In at least one embodiment, an auditory object is projected based on the orientation of the person or a vehicle and the projected auditory is provided to the person. By projecting the auditory object, the attitude of the person or the vehicle can be conveyed to the person to enhance the person's attitude awareness.

Abstract: Methods and systems for more efficient analyses of and response to voice commands and queries are provided. The system may be configured to receive one or more of audio files corresponding to a voice query and determine, for each of the audio files, whether the audio file is a first type of audio file capable of being processed based on a characteristic of the audio file or a second type of audio file that cannot, and may require further processing in order to recognize the voice query associated with the audio file. The system may process each of the first type of audio files and respond to the associated voice queries. The system may also determine a priority for each of the second type of audio files for further processing of the second type of audio files.

Abstract: An audio system and method of using the audio system to augment spatial audio rendition is described. The audio system can include a device to receive user inputs designating positions on an augmented reality view of a listening environment. Sound source icons can be presented in the augmented reality view at the designated positions. The sound source icons can visually represent sound sources at locations in the listening environment that correspond to, but are different than, the positions. One or more processors of the audio system can apply head-related transfer functions, which correspond to the locations in the listening environment, to audio input signals to generate binaural audio signals. The audio system can include a headset that uses the binaural audio signals to render spatialized audio localizing sounds to the locations in the listening environment. Other aspects are also described and claimed.

Abstract: A system can include a position sensor configured to output position data of a HWD. The system can include one or more processors configured to identify a first head angle of the HWD using the position sensor, generate an audio signal using the first head angle, identify a second head angle of the HWD using the position sensor, determine an angle error based at least on the first head angle and the second head angle, and apply at least one of a time difference or a level difference to the audio signal based at least on the angle error to adjust the audio signal. The system can include an audio output device configured to output the adjusted audio signal. By adjusting the audio signal using the angle error, the system can correct for long spatial update latencies and reduce the perceptual impact of such latencies for the user.

Abstract: The present technology relates to a sound processing device and a method that can present progress of sound reproduction. The sound processing device includes a control unit for controlling a sound output that aurally expresses progress of sound reproduction with respect to an entirety of the sound reproduction according to the reproduction of a sound. The present technology can be applied to a sound speech progress presentation UI system.

Abstract: An audio system captures audio data of test sounds through a microphone of a headset worn by a user. The test sounds are played by an external speaker, and the audio data includes audio data captured for different orientations of the headset with respect to the external speaker. A set of head-related transfer function (HRTFs) is calculated based at least in part on the audio data of the test sounds at the different orientations of the headset. A portion of the set of HRTFs is discarded to create an intermediate set of HRTFs. The discarded portion corresponding to one or more distortion regions that are based in part on wearing the headset. One or more HRTFs are generated that correspond to the discarded portion using at least some of the intermediate set of HRTFs to create an individualized set of HRTFs for the user.

Abstract: An exemplary acoustics generation system accesses acoustic propagation data representative of characteristics affecting propagation of a virtual sound to an avatar within an extended reality world being experienced by a user associated with the avatar. Based on the acoustic propagation data, the acoustics generation system also generates a binaural audio signal representative of the virtual sound as experienced by the avatar when the propagation of the virtual sound to the avatar is simulated in accordance with the characteristics affecting the propagation. Additionally, the acoustics generation system prepares the binaural audio signal for presentation to the user as the user experiences the extended reality world by way of the avatar. Corresponding methods and systems are also disclosed.

Abstract: An audio signal processing apparatus includes; a first correlation component separating unit configured to predict a first signal from a second signal in a predetermined period to generate a first correlation component signal and to separate the first non-correlation component signal from the first signal by using the first correlation component signal; a second correlation component separating unit configured to predict a second signal from the first signal in the predetermined period to generate a second correlation component signal and to separate the second non-correlation component signal from the second signal by using the second correlation component signal; a correlation component synthesizing unit configured to synthesize the first correlation component signal and the second correlation component signal to generate a synthesized correlation component signal; a first gain multiplying unit configured to multiply the synthesized correlation component signal by a gain to generate a correlation component

Abstract: The present application provides method and apparatus for a binaural hearing assistance system using a monaural audio signal input. The system, in various examples, provides adjustable delay/phase adjustment and sound level adjustment. Different embodiments are provided for receiving the monaural signal and distributing it to a plurality of hearing assistance devices. Different relaying modes are provided. Special functions are supported, such as telecoil functions. The system also has examples that account for a head-related transfer function in providing advanced sound processing for the wearer. Other examples are provided that are described in the detailed description.

Abstract: A speaker driving device includes a first calculation unit, a first driving signal generation unit and a third calculation unit. The first calculation unit outputs a first calculation signal obtained from a first input signal based on response characteristics according to a first parameter defining an equivalent circuit of a first speaker unit. The first driving signal generation unit generates a first driving signal based on a second driving signal and the first calculation signal. The first driving signal drives a first output speaker unit. The third calculation unit generates a third calculation signal from a second input signal based on response characteristics according to a third parameter defining an equivalent circuit of a third speaker unit. The second driving signal generation unit generates the second driving signal based on the first driving signal and the third calculation signal. The second driving signal drive a second output speaker unit.

Abstract: A filter determination system, a filter determination device, a filter determination method and a program with which a filter can be appropriately determined are provided. A processing device according to this embodiment includes a frequency response acquisition unit that acquires a frequency response of a filter for an audio signal, a smoothing unit that smoothes the frequency response and obtains a smoothed response, a candidate point determination unit that determines candidate split points based on a bottom position of the smoothed response, and a split point determination unit that determines one or more band split points from the candidate split points.

Abstract: An audio signal processing method. The audio signal processing method includes: acquiring a first stereo audio signal; splitting the first stereo audio signal into 5.1-channel audio signals; obtaining processed 5.1-channel audio signals by processing the 5.1-channel audio signals based on a speaker box parameter of a three-dimensional surround 5.1-channel virtual speaker box; and synthesizing the processed 5.1-channel audio signals into a second stereo audio signal.

Abstract: A display apparatus is provided. The display apparatus according to the disclosure includes a display, a plurality of speakers, a sensor, and a processor configured to control the display apparatus to: display an image and generate a sound corresponding to the image and output the sound through the plurality of speakers, and based on a user being sensed by the sensor, determine a location of the user, and based on the determined location of the user, change a first weight of a high band sound and a second weight of a low band sound of the sound, and output the changed sounds through the plurality of speakers.

Abstract: A distance measuring unit 331 measures a distance of a path of sound based on the number of map units between the map unit where a sound source is located and the map unit where a player character is located. An effect unit 332 attenuates the volume of a game sound based on the distance calculated by the distance measuring unit 331. Thereby, it is possible to attenuate the volume level of the sound effect depending on the distance where the sound is transmitted, even if the sound comes around an object obstructing sound.

Abstract: Systems and methods for performing noise reduction in a vehicle comprising a noise reduction system and a plurality of zones are herein provided. A noise reduction criterion for the vehicle is obtained. A first one of the plurality of zones for which noise reduction is to be performed is determined, based on the noise reduction criterion. The noise-reduction system is controlled to perform noise reduction in the first zone, wherein performing noise reduction in the first zone causes the noise reduction system to effect a resultant adjustment in noise level in at least a second one of the plurality of zones.

Abstract: A portable audio device that includes a network card able to connect with a WLAN and a wireless modem to connect to a WWAN. The portable audio device communicates with a voices services platform and/or content provider via the network card and WLAN or the wireless modem and WWAN. If the portable audio device does not have access to the WLAN, the portable audio device may process and respond to voice queries by communicating with the voice services platform via the wireless modem and WWAN. The portable audio device also includes a battery that provides power for the various hardware and software components of the portable audio device to perform various functions, such as advanced voice functions. The portable audio device provides true portability and may be used in any environment, such as within a home or building environment or outside of the home or building environment.

Abstract: An apparatus is disclosed, comprising at least one processor and at least one memory including computer program code, the at least one memory and the computer program code configured with the processor to cause the apparatus to perform the following steps. In a first step, the spatial position of a user in a real world space is detected. In another step, one or more audio signals are provided to the user representing audio from spatially-distributed audio sources in a virtual space. In another step, responsive to detecting movement of the user's spatial position from within a first zone to within a second zone within the real world space, the audio signals for selected ones of the audio sources are modified based on their spatial position in the virtual space.