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 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: 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: 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: 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: 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.

Abstract: Audiovisual presentations, such as film recordings, may have been originally created having an audio soundtrack with multiple audio tracks mixed for a surround sound system that includes a set of speakers physically surrounding a user. The present disclosure presents systems and methods to remix these soundtracks into 3D audio that when presented to the ears of a user can be perceived as a virtual surround sound system that mimics the physical system. What is more, the disclosed systems and methods can enhance the virtual surround sound system by adjusting virtual speakers of the virtual surround sound system according to video content of the audiovisual presentation. Further enhancement may be possible by adjusting the virtual speakers of the virtual surround sound system according to a sensed position of a user.

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: Systems and methods of the present disclosure provide a plurality of audio reactions from a plurality of client devices. The audio reactions are captured by microphones on the client devices and are time-stamped. The method also includes mixing the audio reactions by a mixer server to form a mixed audio reaction, and sending the mixed audio reaction to at least one of the client devices. The client device is adapted to play the mixed audio reaction and a mass media presentation. The mixed audio reaction and the mass media presentation are synchronized to create an audience effect for the mass media presentation. The present technology also provides echo removal, volume balancing, compression, and time stamping of an audio stream by the client device. Reactions from at least one of buttons and gestures to activate synthesized sounds, for example clapping, booing, and cheering, which are mixed into the mixed audio reaction.

Abstract: A test audio pattern is sent to the speaker of the participant computer for outputting by the speaker. A computer receives a microphone input signal from the participant computer that includes the test audio pattern outputted by the speaker of the participant computer, and any ambient noise picked up by the speaker of the participant computer. Ambient noise suppression is performed to cancel out any ambient noise in the microphone input signal picked up by the speaker of the participant computer. The test audio pattern sent to the speaker of the participant computer is compared with the noise-suppressed microphone input signal which includes the test audio pattern outputted by the speaker of the participant computer. An audio signal transform is generated from the comparison.

Abstract: Software and computer processor implemented system and method of providing customized automated responses to different types of incoming electronic messages from various contact sources, particularly useful for preventing distracted driving or distractions at a place of work or at a place of rest and relaxation. Here the invention's software, often in the form of an app, runs on a smartphone or other computerized device configured to automatically connect to various devices, such as automobile associated Bluetooth peripherals or WiFi access points/routers. When operating, the app causes active connections to automatically trigger auto response mode in response to incoming messages. Optionally, different automated responses to be assigned to different connected device identification codes or different incoming message originators.

Abstract: A wearable electronic device (WED) includes one or more sensors and cameras that determine a location of a physical object in a zone where the user is located and that track movement of an electronic device that moves to define a boundary of the zone. The WED includes a processor that generates binaural sound and a display that displays a virtual image of the boundary of the zone and a visual warning that notifies the user of the physical object.

Abstract: An apparatus, method and computer program is described comprising: receiving a near-field audio source signal from a near-field microphone (22); receiving a far-field audio signal from an array comprising one or more far-field microphones (23); determining a filter length of a first portion of a room impulse response filter for the near-field microphone, wherein said filter length of said first portion is the same at each of a plurality of frequency bands of the filter and wherein said filter length of said first portion includes a direct acoustic propagation delay; and determining a filter length of a second portion of the room impulse response filter at each of the plurality of frequency bands, wherein the filter length of said second portion is frequency-dependent.

Abstract: There is disclosed inter alia an apparatus for spatial audio signal encoding which determines at least one spatial audio parameter comprising a direction parameter with an elevation component and an azimuth component. The elevation component and azimuth component of the direction parameter are then converted to an index value.

Abstract: The present disclosure relates to an acoustic output apparatus. The acoustic output apparatus comprising: at least one low-frequency acoustic driver that outputs sound from at least two first sound guiding holes; at least one high-frequency acoustic driver that outputs sound from at least two second sound guiding holes; and a controller configured to cause the low-frequency acoustic driver to output sound in a first frequency range, and cause the high-frequency acoustic driver to output sound in a second frequency range, wherein the second frequency range includes frequencies higher than the first frequency range.

Abstract: This patent includes a method to generate a six dimensional audio dataset. When listening to the six dimensional audio dataset with an advanced headset disclosed in this patent, a user will enable a user to have precision sound localization and maximize the listening experience. For example, when the audio file is registered to a living room, a user would be able to sit in the “middle of the band” and if desired, reposition the instruments to different locations within the living room to change the audio experience.

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: 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 is an apparatus and method for processing a multichannel audio signal. A multichannel audio signal processing method may include: generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; and generating a stereo audio signal by performing binaural rendering of the N-channel audio signal.

Abstract: Methods (700, 800, 900), systems (200, 300, 400, 500, 600) and computer program products are provided. Location metadata (620) associated with an audio object is received (801). The location metadata defines a position of the audio object in an audio scene. It is estimated (630, 802), based on the location metadata, whether the audio object includes dialog. A value representative of a result of the estimation is assigned (803) to an object type parameter (231). In some example embodiments, audio objects are selected (661, 662, 804) based on values of their respective of object type parameters. In some example embodiments, at least one of the selected audio objects is submitted to dialog enhancement (690, 807).

Abstract: Techniques for estimating a direction of arrival of sound in an environment are discussed. First and second audio data are received from a first pair of audio sensors associated with a vehicle. A first region of ambiguity associated with the first pair of audio sensors is determined based on the first and second audio data. Third and fourth audio data are received from a second pair of audio sensors. A second region of ambiguity associated with the second pair of audio sensors is determined based on the third and fourth audio data. The regions of ambiguity can be further based on confidence levels associated with sensor or audio data. An area of intersection of the first region of ambiguity and the second region of ambiguity can be determined. A direction of arrival of an audio event can be determined based on the area of intersection.

Abstract: An apparatus comprising means for: obtaining an indication of a position of at least one user in real space; mapping the position of the user in real space to a position of the user in a sound space; and controlling an output audio signal, for rendering a sound scene via multiple audio channels, to provide rendering of a first sub-set of sound sources via at least a first sub-set of the multiple audio output channels and rendering a second sub-set of sound sources via at least a second sub-set of the multiple audio output channels, wherein an allocation of a plurality of sound sources to the first sub-set of sound sources or the second sub-set of sound sources is dependent upon at least a position of the user in the sound space; wherein an allocation of the multiple audio output channels to the first sub-set of the multiple audio output channels or the second sub-set of the multiple audio output channels is dependent upon available audio paths for the multiple audio output channels.

Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

Abstract: An example device includes a memory configured to store at least one spatial component and at least one audio source within a plurality of audio streams. The device also includes one or more processors coupled to the memory. The one or more processors are configured to receive, from motion sensors, rotation information. The one or more processors are configured to rotate the at least one spatial component based on the rotation information to form at least one rotated spatial component. The one or more processors are also configured to reconstruct ambisonic signals from the at least one rotated spatial component and the at least one audio source, wherein the at least one spatial component describes spatial characteristics associated with the at least one audio source in a spherical harmonic domain representation.

Abstract: A method for decoding a bitstream by an apparatus, includes obtaining a decoded audio signal and metadata from the bitstream, the metadata comprising scene orientation information; and rendering the decoded audio signal based on the scene orientation information, wherein the scene orientation information is information for a direction of a video scene related to the decoded audio signal.

Abstract: Provided is an automatic grading method including calculating a first strain ratio between a three-dimensional (3D) source avatar and a 3D target avatar, determining a mapping relationship between 3D source garment draped over the source avatar and a body portion of the source avatar, converting the source garment into 3D target garment draped over the target avatar, based on the first strain ratio and the mapping relationship, and outputting a two-dimensional (2D) target pattern constituting the target garment.