Abstract: An audio signal is enhanced using interaural coherence to control noise reduction and binaural cue preservation. Sounds from a local area are detected via an acoustic array. An interaural coherence is determined using the detected sounds. Sound filters for an audio signal are generated based on the interaural coherence. The sound filters implement a tradeoff between increasing signal-to-noise ratio (SNR) between a target source and an interfering source and preserving of binaural information of the interfering source. The tradeoff is controlled based on the interaural coherence. The sound filters are applied to the audio signal to generate audio content. The audio content is presented via a speaker array.

Abstract: Provided are a display apparatus and a computing apparatus including the same. A display apparatus includes: a display module including a display panel configured to display an image, a vibration plate on the display module, a first air gap between the vibration plate and the rear surface of the display module, a vibration module on the vibration plate, and a system rear cover on the vibration plate, a second air gap between the system rear cover and the vibration plate, wherein the vibration plate includes a communication part configured to allow air to flow between the first air gap and the second air gap.

Abstract: A method, computer program product, and computing system for receiving a signal from each microphone of a plurality of microphones, thus defining a plurality of signals. One or more inter-microphone gain-based augmentations may be performed on the plurality of signals, thus defining one or more inter-microphone gain-augmented signals.

Abstract: An electro-acoustical transducer such as a Piezoelectric Micromachined Ultrasonic Transducers is coupled with an adjustable load circuit having a set of adjustable load parameters including resistance and inductance parameters. Starting from at least one resonance frequency or at least one ring-down parameter of the electro-acoustical transducer a set of model parameters is calculated for a Butterworth-Van Dyke (BVD) model of the electro-acoustical transducer. The BVD model includes an equivalent circuit network having a constant capacitance coupled to a RLC branch and the adjustable load circuit is coupled with the electro-acoustical transducer at an input port of the equivalent circuit network of the model of the electro-acoustical transducer. The adjustable load parameters are adjusted as a function of the set of model parameters calculated for the BVD model of the electro-acoustic transducer to increase the bandwidth or the sensitivity of the electro-acoustic transducer.

Abstract: A microphone array device including microphone capsules and at least one processing unit configured to receive output signals of the microphone capsules, dynamically steer an audio beam based on the received output signal of the microphone capsules, and generate and provide an audio output signal based on the received output signal of the microphone capsules. The processing unit is configured to operate in a dynamic beam mode where at least one focused audio beam is formed that points towards a detected audio source and in a default beam mode where a broader audio beam is formed that covers substantially a default detection area. The microphone array may be incorporated into a conference system.

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: A method, computer program product, and computing system for receiving a signal from each microphone of a plurality of microphones, thus defining a plurality of signals. One or more microphone frequency responses associated with at least one microphone may be received. One or more microphone frequency response-based augmentations may be performed on the plurality of signals based upon, at least in part, the one or more microphone frequency responses, thus defining one or more microphone frequency response-based augmented signals.

Abstract: A device includes one or more processors configured to execute instructions to determine a first phase based on a first audio signal of first audio signals and to determine a second phase based on a second audio signal of second audio signals. The one or more processors are also configured to execute the instructions to apply spatial filtering to selected audio signals of the first audio signals and the second audio signals to generate an enhanced audio signal. The one or more processors are further configured to execute the instructions to generate a first output signal including combining a magnitude of the enhanced audio signal with the first phase and to generate a second output signal including combining the magnitude of the enhanced audio signal with the second phase. The first output signal and the second output signal correspond to an audio zoomed signal.

Abstract: A vibration device enabling a large area and a display apparatus including the vibration device are provided. The vibration device may be configured to include a vibration array including a plurality of vibration modules which are spaced apart from one another by a first interval of less than 3 cm in a first direction.

Abstract: A voice processing method, an electronic device and a readable storage medium, which relate to the field of voice processing technologies, are disclosed. The method includes: collecting a first audio signal; processing the first audio signal using a preset algorithm to obtain a second audio signal; and sending the second audio signal to a first device, such that the first device performs a voice processing operation on the second audio signal.

Abstract: A digital microphone includes at least one integrator; a state detection and parameter control component directly coupled to an output of the integrator; and a signal processing component coupled to an output of the state detection and parameter control component, wherein a parameter of the signal processing component includes a first value in a first operational mode and a second value in a second operational mode different from the first operational mode.

Abstract: Certain embodiments relate generally to modifying audio playing on a first device based on detection of that audio by a second device. Other embodiments relate to transferring audio between a first device and a second device. More particularly, audio playing from a first device may be muted, stopped, or adjusted in volume based on detection of that audio by, or interaction with, a second device. Likewise, audio may be transferred from a first device to a second device based on communications between the first and second devices, proximity of the first and second devices relative to one another, proximity of a user to either the first or second device, and so on.

Abstract: Disclosed are a robot which localizes a sound source based on sound spot information of at least one sound spot, and a sound source localization method. The robot can communicate with other electronic devices and a server in a 5G communication environment.

Abstract: A method of recording ambisonic sound fields with a spatially distributed plurality of ambisonic microphones comprising a step of recording sound signals from plurality of ambisonic microphones a step of converting recorded sound signals to ambisonic sound fields and a step of interpolation of the ambisonic sound fields according to the invention comprises a step of generating synchronizing signals for particular ambisonic microphones for synchronized recording of sound signals from plurality of ambisonic microphones and during the step of interpolation of the ambisonic sound fields it includes filtering sound signals from particular microphones with individual filters having a distance-dependent impulse response having a cut-off frequency fc(dm) depending on distance dm between point of interpolation and m-th microphone applying gradual distance dependent attenuation applying re-balancing with amplification of 0th ordered ambisonic component and attenuating remaining ambisonic components.

Abstract: A method and an apparatus for enabling adaptive audio signal alteration are described. When an input audio signal is received, a determination of whether the user of an audio device hears the input audio signal is performed based upon brain activity of the user. A determination of whether the user is distracted by the audio signal is performed based upon sensor measurements indicating a physical state of the user. In response to determining that the user hears the input audio signal and that the input audio signal causes the user to be distracted, a determination of configuration parameter(s) is performed. An alteration of audio signal(s) is caused based upon the configuration parameter(s) to obtain modified version(s) of the audio signal(s) that are intended to address the distraction caused by the input audio signal, and output audio signals are output, where the output audio signals include the modified versions.

Abstract: A method, computer program product, and computing system for receiving a signal from each microphone of a plurality of microphones, thus defining a plurality of signals. One or more inter-microphone gain-based augmentations may be performed on the plurality of signals, thus defining one or more inter-microphone gain-augmented signals.

Abstract: A system for dynamically updating a head-related transfer function (HRTF) model that is customized to a user. The system receives one or more images of the user captured by one or more imaging devices. The system determines a pose of the user using the one or more captured images. The pose of the user includes a head-torso orientation of the user. The system updates a HRTF model for the user based on the determined pose including the head-torso orientation. The system generates one or more sound filters using the updated HRTF model and applies the one or more sound filters to audio content to generate spatialized audio content. The system provides the spatialized audio content to the user.

Abstract: Adaptive operations of a first noise control system and a second noise control system may include a speaker that outputs noise cancellation sound, a microphone that detects an error signal, an auxiliary filter that generates, from a noise signal, a correction signal that corrects the error signal so that a difference in a position between the microphone and a noise cancellation position is compensated, and an adaptive filter that performs an adaptive operation using the corrected error signal to generate the noise cancellation sound from the noise signal are alternately performed. A transfer function learned in a state in which the second noise control system is stopped is set in the auxiliary filter of the first noise control system, and a transfer function learned in a state in which the adaptive operation of the first noise control system is stopped is set in the auxiliary filter of the second noise control system.

Abstract: Aspects of the disclosure include methods, apparatuses, and non-transitory computer-readable storage mediums for representing a space of interest of an audio scene. One apparatus includes processing circuitry that decodes audio scene data for the audio scene. The audio scene data includes (i) audio content for a plurality of items representing the audio scene and (ii) a first syntax element indicating a type of a subset of the plurality of items. The subset of the plurality of items represents the space of interest of the audio scene. The processing circuitry determines a part of the audio content for the subset of the plurality of items based on the type of the subset of the plurality of items indicated in the first syntax element. The processing circuitry renders the determined part of the audio content.

Abstract: A system and method for externalizing sound. The system includes a headphone assembly and a localizer configured to collect information related to a location of the user and of an acoustically reflective surface in the environment. A controller is configured to determine a location of at least one virtual sound source, and generate head related transfer functions that simulate characteristics of sound from the virtual sound source directly to the user and to the user via a reflection by the reflective surface. A signal processing assembly is configured to create one or more output signals by filtering the sound signal respectively with the HRTFs. Each speaker of the headphone assembly is configured to produce sound in accordance with the output signal.