Abstract: An audio customization system operates to enhance a user's audio environment. A user may wear headphones and specify what portion the ambient audio and/or source audio will be transmitted to the headphones or the personal speaker system. The audio signal may be enhanced by application of a spatialized transformation using a spatialization engine such as head-related transfer functions so that at least a portion of the audio presented to the personal speaker system will appear to originate from a particular direction. The direction may be modified in response to movement of the personal speaker system.

Abstract: An information processing device includes circuitry that estimates an attribute of a first person located in at least one of a plurality of areas, and sets a first content corresponding to the attribute of the first person and outputs a sound through wave field synthesis so that a sound field of the set first content is spatially transmitted to at least the one of the plurality of areas via a plurality of speakers.

Abstract: Various implementations include computer-implemented methods and related systems for controlling the phantom center image of an audio output in an automobile. In one implementation, a method includes: receiving at least one user interface command to modify a phantom center image of audio output from the audio system in the automobile, wherein the phantom center image of the audio output includes a designated position of sound produced by a set of speakers in the audio system other than physical locations of the set of speakers in the audio system; and adjusting a perceived location of the phantom center image of the audio output from the audio system based upon the at least one user interface command to modify the phantom center image of the audio output.

Abstract: At least one exemplary embodiment is directed to a method and device for voice operated control with learning. The method can include measuring a first sound received from a first microphone, measuring a second sound received from a second microphone, detecting a spoken voice based on an analysis of measurements taken at the first and second microphone, learning from the analysis when the user is speaking and a speaking level in noisy environments, training a decision unit from the learning to be robust to a detection of the spoken voice in the noisy environments, mixing the first sound and the second sound to produce a mixed signal, and controlling the production of the mixed signal based on the learning of one or more aspects of the spoken voice and ambient sounds in the noisy environments.

Abstract: The present technology relates to a signal processing device, method, and program that can improve encoding efficiency. A signal processing device includes: an acquisition unit that acquires reverb information including at least one of space reverb information specific to a space around an audio object or object reverb information specific to the audio object and an audio object signal of the audio object; and a reverb processing unit that generates a signal of a reverb component of the audio object on the basis of the reverb information and the audio object signal. The present technology can be applied to a signal processing device.

Abstract: A headphone arrangement includes two earphones, wherein each earphone comprises a housing encompassing a low-frequency transducer and an array of at least three high-frequency transducers. The low-frequency transducer of each earphone is disposed on or over an ear canal of a user when the earphone is worn by the user, and is configured to broadcast low-frequency sound that corresponds to low-frequency components of an input signal. The array of at least three high frequency transducers of each array are configured to broadcast high-frequency sound that corresponds to high-frequency components of the input signal, and the array of at least three high frequency transducers of each array is disposed adjacent to the low-frequency transducer and in a lower rostral quadrant of a full circle around the low-frequency transducer when the earphone is worn by the user.

Abstract: The present disclosure relates to a system and a processing method in association with the system for customizing audio experience. Customization of audio experience can be based on derivation of at least one customized audio response characteristic which can be applied to an audio device used by a person. The customized audio response characteristic(s) can be unique to the person.

Abstract: Embodiments of the present disclosure provide systems and methods for synchronizing an unaligned audio stream and a corresponding unaligned video stream of real time streaming media. A non-limiting disclosed method comprises performing, using a video classifier, video reference point classification of a video stream based on an audio-video dataset; performing, using an audio classifier, audio reference point classification of the audio stream based on the audio-video dataset. The system correlates object related reference points in video segments of the video stream and in audio segments of the audio stream to identify a set of audio-video synchronization candidates. The system compares context of the set of audio-video synchronization candidates to identify an audio-video synchronization candidate to synchronize the audio stream and video stream based on reference point alignment.

Abstract: A technology is provided that reduces noise heard when a user sits in a seat of an aircraft, without using earphones or headphones. A sound system includes: a control system that generates a control signal for canceling noise in a place close to a head of a user using a seat of an aircraft, from a signal of the noise (noise signal); and a noise-cancellation speaker system including speaker units that emit sound based on the control signal (first noise-cancellation speaker unit, . . .

Abstract: When a first condition related to a change in a position of a virtual microphone in a virtual space is satisfied, a residual virtual microphone is placed at a position of the virtual microphone before the change in the position, and residual virtual microphone acquisition sound data whose volume is set on the basis of a distance between the residual virtual microphone and a virtual sound source and current virtual microphone acquisition sound data whose volume is set on the basis of a distance between the virtual microphone after the change in the position and the virtual sound source are outputted to a speaker such that an output level of the residual virtual microphone acquisition sound data is gradually decreased and an output level of the current virtual microphone acquisition sound data is gradually increased.

Abstract: A sound effect optimizing method, an electronic device, and a non-transitory computer computer-readable storage medium are provided. The method includes controlling the speaker to play an audio signal emitted by a virtual sound source; receiving a sound source identifying result, the sound source identifying result including a first position relationship, and the first position relationship being a position relationship between the virtual sound source and a user and determined by the audio signal; and adjusting a sound effect parameter until the first position relationship is consistent with a second position relationship in response to the first position relationship being inconsistent with the second position relationship, the second position relationship being an actual position relationship between the virtual sound source and the user.

Abstract: A method or apparatus provides binaural sound that originates to a person at a location inside a field of view (FOV) provided to the person by an electronic device. The electronic device determines when the location of the binaural sound moves outside the FOV and assists the person in determining the location of the binaural sound. A display of the electronic device displays a visual indication that shows a direction to the location of the binaural sound outside the FOV.

Abstract: The present disclosure relates to methods, apparatus and systems for encoding an audio signal into a bitstream, in particular at an encoder, comprising: encoding or including audio signal data associated with 3DoF audio rendering into one or more first bitstream parts of the bitstream, and encoding or including metadata associated with 6DoF audio rendering into one or more second bitstream parts of the bitstream. The present disclosure further relates to methods, apparatus and systems for decoding an audio signal and audio rendering based on the bitstream.

Abstract: An apparatus for rendering a sound scene having reflection objects and a sound source at a sound source position, includes: a geometry data provider for providing an analysis of the reflection objects of the sound scene to determine a reflection object represented by a first polygon and a second adjacent polygon having associated a first image source position for the first polygon and a second image source position for the second polygon, wherein the first and second image source positions result in a sequence including a first visible zone related to the first image source position, an invisible zone and a second visible zone related to the second image source position; an image source position generator for generating an additional image source position such that the additional image source position is placed between the first image source position and the second image source position; and a sound renderer for rendering the sound source at the sound source position and, additionally for rendering the sound

Abstract: Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.

Abstract: An electronic apparatus is provided. The electronic apparatus includes a camera, a processor and a memory configured to store at least one instruction executable by the processor where and the processor is configured to input audio data to an artificial intelligence model corresponding to user information, and obtain output audio data from the artificial intelligence model, and the artificial intelligence model is a model learned based on first learning audio data obtained by recording a sound source with a first recording device, second learning audio data obtained by recording the sound source with a second recording device, and information on a recording device for obtaining the second learning audio data, and the second learning audio data is binaural audio data.

Abstract: In general, the subject matter described in this disclosure can be embodied in methods, systems, and program products for identifying that a first audio stream includes first, second, and third sources of audio. A computing system identifies that a second audio stream includes the first, second, and third sources of audio. The computing system determines that the first and second sources of audio are part of a first conversation. The computing system generates a third audio stream that combines the first source of audio from the first audio stream, the first source of audio from the second audio stream, the second source of audio from the first audio stream, and the second source of audio from the second audio stream, and diminishes the third source of audio from the first audio stream, and the third source of audio from the second audio stream.

Abstract: A device includes one or more processors configured to perform a first boost operation on a left audio signal to generate a boosted left audio signal, perform a second boost operation on a right audio signal to generate a boosted right audio signal, and perform a crosstalk cancellation operation that includes generation of an adjusted left audio signal and an adjusted right audio signal. The one or more processors are also configured to provide a left output signal for playback at a first speaker, the left output signal based at least in part on the adjusted left audio signal, and to provide a right output signal for playback at a second speaker, the right output signal based at least in part on the adjusted right audio signal.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A wearable spatial audio device is provided. The wearable spatial audio device includes one or more audio speakers, one or more processors, and a storage machine holding instructions executable by the one or more processors. Map data is obtained for a real-world environment that includes one or more dynamic audio objects. A device-specific subset of audio tracks is obtained, and a device-specific spatialized audio mix of the device-specific subset of audio tracks that is based on the map data is obtained. An indication of a change in an environmental condition relative to one or more of the dynamic audio objects is received. The device-specific spatialized audio mix is adjusted based on the change in the environmental condition.

Abstract: In one aspect, there is a method for rendering a spatially-heterogeneous audio element. In some embodiments, the method includes obtaining two or more audio signals representing the spatially-heterogeneous audio element, wherein a combination of the audio signals provides a spatial image of the spatially-heterogeneous audio element. The method also includes obtaining metadata associated with the spatially-heterogeneous audio element, the metadata comprising spatial extent information indicating a spatial extent of the audio element. The method further includes rendering the audio element using: i) the spatial extent information and ii) location information indicating a position (e.g. virtual position) and/or an orientation of the user relative to the audio element.

Abstract: The present disclosure pertains to an apparatus comprising circuitry configured to: determine a loudspeaker dependent spread factor for at least one individual loudspeaker of a loudspeaker arrangement, wherein the loudspeaker dependent spread factor depends on a specification of the at least one individual loudspeaker; and 5 control the outputs of the loudspeakers of the loudspeaker arrangement based on the loudspeaker dependent spread factor for the at least one individual loudspeaker to generate at least one virtual sound source.

Abstract: In general, disclosed is a device that includes one or more processors, coupled to the memory, configured to perform an energy analysis with respect to one or more audio objects, in the ambisonics domain, in the first time segment. The one or more processors are also configured to perform a similarity measure between the one or more audio objects, in the ambisonics domain, in the first time segment, and the one or more audio objects, in the ambisonics domain, in the second time segment. In addition, the one or more processors are configured to perform a reorder of the one or more audio objects, in the ambisonics domain, in the first time segment with the one or more audio objects, in the ambisonics domain, in the second time segment, to generate one or more reordered audio objects in the first time segment.

Abstract: Disclosed in the invention is a virtual reproduction method for a multichannel spatial surround sound in three-dimensional space. Multichannel spatial surround sound signals are undergone a sum and difference operation and processed with virtual reproduction signal processing functions, then fed to four actual loudspeakers arranged at left-front-up and right-front-up directions on a high elevation plane of 30°±10° and horizontal left-front and right-front directions for reproduction, and generate a auditory effect of spatial surround sound. The invention simplifies the number and arrangement of loudspeakers required for multichannel spatial surround sound reproduction, and is suitable for cases in which the arrangement of multiple loudspeakers for spatial surround sound is infeasible, such as in a TV set, and so on.

Abstract: Embodiments of the present disclosure disclose an audio processing method, an audio processing apparatus, and storage medium. The audio processing apparatus obtains first audio data of a target speaker acquired by the audio acquisition device on the target speaker, and obtains second audio data acquired by the electronic device on where the target speaker is located. According to a position of the target speaker relative to the electronic device, the first audio data is combined with the second audio data to obtain target audio data, where, when the target audio data is output, voice of the target speaker aligns with the spatial location. Accuracy of the spatial effect of the target speaker's voice in the target audio data is relatively ensured.

Abstract: Various implementations include audio systems and methods for mixed rendering to enhance audio output. Certain implementations include an audio system having: at least one far-field speaker configured to output a first portion of an audio signal; and a pair of non-occluding near-field speakers configured to output a second portion of the audio signal in synchrony with the output of the first portion of the audio signal, where the first portion of the audio signal is distinct from the second portion of the audio signal.

Abstract: A wearable electronic device (WED) worn on a head of a user displays a virtual image that has sound. One or more processors process the sound for the virtual image into binaural sound for the user. The binaural sound has a sound localization point (SLP) with a coordinate location that occurs behind the virtual image while the image is located a near-field distance from the head of the user.

Abstract: Examples of the disclosure describe systems and methods for presenting an audio signal to a user of a wearable head device. In an example, a received first input audio signal is processed to generate a left output audio signal and a right output audio signal presented to ears of the user. Processing the first input audio signal comprises applying a delay process to the first input audio signal to generate a left audio signal and a right audio signal; adjusting gains of the left audio signal and the right audio signal; applying head-related transfer functions (HRTFs) to the left and right audio signals to generate the left and right output audio signals. Applying the delay process to the first input audio signal comprises applying an interaural time delay (ITD) to the first input audio signal, the ITD determined based on the source location.

Abstract: The present disclosure relates to a waterproof open earphone. The waterproof open earphone may include a housing, at least one button, at least one elastic pad, and at least one pair of speaker units. The housing may be placed on a head or at least one ear of a user while not blocking an ear canal of the user. The at least one button may be set on the housing, wherein each of the at least one button corresponds to a button hole. The at least one elastic pad may correspond to the at least one button, respectively, wherein each elastic pad prevents the corresponding button from moving relative to the button hole. Each pair of the at least one pair of speaker units may generate sound within a frequency range from two sound guiding holes through two sound guiding tubes.

Abstract: Disclosed are techniques for performing ray-based acoustic modeling that models scattering of acoustic waves by a surface of a device. The acoustic modeling uses two parameters, a room response representing acoustics and geometry of a room and a device response representing acoustics and geometry of the device. The room response is determined using ray-based acoustic modeling, such as ray tracing. The device response can be measured in an actual environment or simulated and represents an acoustic response of the device to individual acoustic plane waves. The device applies a superposition of the room response and the plane wave scattering from the device response to determine acoustic pressure values and generate microphone audio data. The device can estimate room impulse response (RIR) data using data from the microphones, and can use the RIR data to perform audio processing such as sound equalization, acoustic echo cancellation, audio beamforming, and/or the like.

Abstract: The present technology relates to a signal processing device, method, and program that can improve encoding efficiency. A signal processing device includes: an acquisition unit that acquires reverb information including at least one of space reverb information specific to a space around an audio object or object reverb information specific to the audio object and an audio object signal of the audio object; and a reverb processing unit that generates a signal of a reverb component of the audio object on the basis of the reverb information and the audio object signal. The present technology can be applied to a signal processing device.

Abstract: There is provided a method of controlling an array of loudspeakers. The method comprises: receiving a plurality of input audio signals to be reproduced, by the array, at a respective plurality of control points in an acoustic environment; and generating a respective output audio signal for each of the loudspeakers in the array by applying a set of filters to the plurality of input audio signals. The set of filters is based on: a first plurality of filter elements based on a first approximation of a set of transfer functions, each transfer function in the set of transfer functions being between an audio signal applied to a respective one of the loudspeakers and an audio signal received at a respective one of the control points from the respective one of the loudspeakers; and a second plurality of filter elements based on a second approximation of the set of transfer functions.

Abstract: This disclosure describes embodiments of methods, non-transitory computer-readable media, and systems for detecting that a physical space includes a physical object corresponding to an analogous virtual object from an augmented reality experience and rendering (or otherwise modifying) the augment reality experience to integrate the physical object as part of the experience. In particular, the disclosed systems can determine that a physical object within a physical environment corresponds to an analogous virtual object of an augmented reality experience. Based on this correspondence, the disclosed systems can modify one or more of the virtual graphics, sound, or other features corresponding to the augmented reality experience to represent the virtual object using the physical object. For example, the disclosed systems can modify acoustic features of a sound for the augmented reality experience to simulate the sound originating from (or being affected by) the physical object.

Abstract: Provided is an information processing apparatus including: a speaker array that includes a plurality of speakers, and performs wavefront synthesis by using an output of the plurality of speakers; and a presentation unit that presents visual information indicating a state of waves on a wavefront formed in the wavefront synthesis, or presents visual information based on positional information of a virtual sound image that has been formed in a position that is different from a vicinity of the speaker array in the wavefront synthesis.

Abstract: A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound.

Abstract: A method for reproducing an external sound in an automobile includes recording external sound using external microphones located on an automobile exterior of the automobile; establishing recorded audio signals based on the recorded external sound; determining from the recorded audio signals a relative directionality of external sound sources of the recorded external sound with respect to at least one occupant within the automobile; establishing at least one directionality encoded audio signal by applying at least one frequency dependent directionality filter to the recorded audio signals, wherein at least one frequency dependent directionality filter is based on the determined relative directionality of external sound sources, thereby encoding the relative directionality into said at least one directionality encoded audio signal; reproducing a representation of the external sound based on at least one directionality encoded audio signal using internal loudspeakers located within an automobile interior of the

Abstract: Techniques for calibrating listening devices are disclosed herein. The techniques include emitting a predetermined audio signal using an outward-facing transducer located on a first portion of a head-mounted device worn by the user, receiving the predetermined audio signal at a microphone located on a second portion of the head-mounted device, the second portion being different from the first portion, determining a transfer function for the user based on the received predetermined audio signal, and applying the transfer function to audio signals transmitted to the user.

Abstract: Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.

Abstract: The invention relates to the technical fields of binaural audio rendering and, to this end, estimation of room acoustic parameters like reverberation time. In particular, the invention provides a device and method for estimating such acoustic parameters. The device is configured to record an acoustic signal, particularly a speech signal, to estimate a frequency-dependent reverberation time in a lower frequency range based on the recorded acoustic signal, and to extend the frequency-dependent reverberation time to a higher frequency range based on a predetermined model to obtain an extended frequency-dependent reverberation time. Virtual 3D audio can thus be adapted to a real room.

Abstract: A method of incorporating environmental acoustic sources into a virtual environment by measuring real environment acoustic sources and locations and incorporating them into a virtual environment with virtual acoustic sources.

Abstract: In one embodiment, a method for emitting a sound from an in-ear speaker worn by a subject includes generating, by an audio source of the in-ear speaker, a source audio signal. One or more speakers of the in-ear speaker may emit a sound based on the audio signal and an audio-transport tube of the in-ear speaker may receive the sound. The one or more speakers may be comprised of a singular speaker or a speaker array coupled to a crossover network. The audio-transport tube has an input end coupled to the one or more speakers to receive the sound. An audio reflector of the in-ear speaker may reflect the sound. The audio reflector is coupled to an output end of the audio-transport tube.

Abstract: The disclosed computer-implemented method may include receiving a signal for each channel of an audio transducer array on a wearable device. The method may also include calculating a beamformed signal for each beam direction of a set of beamforming filters for the wearable device. Additionally, the method may include classifying a first beamformed signal from the calculated beamformed signals into a first class of sound and a second beamformed signal from the calculated beamformed signals into a second class of sound. The method may also include adjusting, based on the classifying, a gain of the first beamformed signal relative to the second beamformed signal. Furthermore, the method may include converting the beamformed signals into spatialized binaural audio based on a position of a user. Finally, the method may include transmitting the spatialized binaural audio to a playback device. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: A new real-time spatial audio rendering system includes a real-time spatial audio rendering computer software application adapted to run on a communication device. The application renders stereo audio from mono audio sources in a virtual room of a listener. The listener can be mobile. The stereo audio is rendered for each listener within the room. The real-time spatial audio rendering system has two different modes, with and without reverberation. Reverberation can provide the sense of the dimensions of the room, First, the anechoic processing module produces the anechoic stereo audio that provides the sense of direction and distance of spatial audio. When reverberation is desired, the reverberation processing module is also performed to provide the sense of the room's dimensions by the spatial audio.

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

Abstract: Embodiments relate to binaural spatialization of more than two sound sources on two audio channels of an audio system. Sound signals each emitted from a corresponding sound source are collected, and a respective virtual position within an angular range of a sound scene is assigned to each sound source. Multi-source audio signals are generated by panning each sound signal according to the respective virtual position. A first multi-source audio signal is spatialized to a first direction to generate a first left signal and a first right signal. A second multi-source audio signal is spatialized to a second direction to generate a second left signal and a second right signal. A binaural signal is generated using the first left signal, the second left signal, the first right signal, and the second right signal. The binaural signal is such that each sound source appears to originate from its respective virtual position.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Abstract: An apparatus including at least one processor and at least one memory including computer code for one or more programs, the at least one memory and the computer code configured, with the at least one processor, to cause the apparatus at least to: generate content lock information for a content lock, wherein the content lock information enables control of audio signal processing associated with audio signals related to one or more audio sources based on a position and/or orientation input.

Abstract: An illustrative audio processing system identifies an experience location with which an augmented reality presentation device is associated. The experience location is included within a multizone augmented reality space that is presented by the augmented reality presentation device. The audio processing system determines that the experience location is within both a first sound zone and a second sound zone of the multizone augmented reality space, and, based on the determining that the experience location is within both the first and second sound zones, generates a binaural audio stream for presentation by the augmented reality presentation device. The binaural audio stream includes an environmental audio component implemented by a mix of a first environmental audio stream associated with the first sound zone and a second environmental audio stream associated with the second sound zone. Corresponding methods and systems are also disclosed.

Abstract: The present disclosure provides an acoustic output apparatus. The acoustic output apparatus may include an acoustic output component and a supporting structure forming an acoustically open structure that allows the acoustic output component to acoustically communicate with the surroundings. The acoustic output component may include a plurality of acoustic drivers, each of which may be configured to output a sound with a frequency range. At least one of the acoustic drivers may include a magnetic system for generating a first magnetic field. The magnetic system may include a first magnetic component for generating a second magnetic field and at least one second magnetic component. A magnetic gap may be formed between the first magnetic component and the at least one second magnetic component. A magnetic field intensity of the first magnetic field in the magnetic gap may be greater than that of the second magnetic field in the magnetic gap.

Abstract: A system and method of modifying a binaural signal using headtracking information. The system calculates a delay, a first filter response, and a second filter response, and applies these to the left and right components of the binaural signal according to the headtracking information. The system may also apply headtracking to parametric binaural signals. In this manner, headtracking may be applied to pre-rendered binaural audio.